XXHighEnd

I did not know whether to open a new thread with this or whether to respond under the "hunting for noise" thread anyway please move it Peter if you need to.

This post relates my experiences changing the standard cheapo 24Mhz clocks in the PC and NOS1 interfaces to high quality clocks. By some amazing co-incidence Mani dropped in for a quick visit at just the right time and was able to hear the results of the first part of the test and so I am sure he will add a short note about what he heard.

Regular readers will know that recently I went over to Nick's to listen to his system and in particular to listen to the impact of installing high quality clocks in the USB link from PC to NOS1. Nick's experiences have been posted elsewhere but since that visit I took the plunge and ordered a high quality Dexa clock of the 24Mhz variety. My thinking was that I had already invested in a Paul Pang USB3 pcie card (low jitter) for the PC end and so I would not need to go to the expense of a Dexa for the PC end would I?

So on to PART 1 of the test - Leaving the USB clock in NOS1 untouched I took a stock USB3 pcie card and removed the quartz clock and the 2 caps that are strapped across the clock leads and connected the Dexa clock powered by its dedicated high quality power supply. The Paul Pang low jitter pcie card had previously provided a welcome improvement and I really did not think that using a stock pcie card clocked with the Dexa would make much difference. Well I was wrong the Dexa powered pcie card sounded MUCH better.  When I first listened I thought there was less treble - but I soon realised that the treble was much smoother and more detailed. Going back to the PP card the treble sounded 'splashy' and ill defined less precise. Drums and symbols became grey and less precise in the mix. The mids with the PP card sounded 'grey' lacking tonal colour compared with the Dexa. Bass I would say sounded maybe slightly more controlled with the Dexa but no big difference really to my ears. But the biggest surprise of all was just how much better the Dexa clocked pcie sounded than the (already good) Paul Pang card. As mentioned above Mani heard that comparison so over to you Mani.


PART 2 this is where the fun really starts. I was already really enjoying much improved sound (I hope Mani agees!!) with the dexa clocked pcie card but I took a big plunge and ordered a second 24Mhz Dexa clock for the NOS1 end of the USB link. That is significant money for me I have to say. Especially as I was REALLY happy with the sound of the Dexa pcie card so I did wonder if it was possible to improve the sound quality further - but there again I had already heard what happened to Nicks system with this change so I took the plunge.

Making changes to SMD's on the NOS1 is not for the faint hearted a screw up there and it is big bucks to fix - there would also be the very considerable embarrassment of returning NOS1 to Peter for repair. I am sure that eventuality would have also cost me a bottle of whisky ha ha. Anyway having already honed my SMD skills on old boards lying around and then finally on the pcie card I dived into the NOS1. After a short nerve wracked time period I connected up NOS1 with Dexa clock running. So starting everything up in the right sequence I sat back to listen and all I can say is WOW what a difference. The already very good bass (and oh boy have I spent serious time getting the bass right in my system/room) was REALLY improved. Suddenly bass had real attack and speed and really excellent control (which are all qualities I thought it had before). But that was just for starters the already improved detail and subtly in the sound took another big step up. Also the tonal quality of the sound just superb. But what is really impressive is that sound just moves to a completely new level in sounding REAL and PALPABLE. Those are qualities I just do not get used to every day I switch on listen and smile. Put an album on, any album and I just do not want to take it off.

Vocals are just incredible - crystal clear - changing from Dexa pcie back to normal USB at the PC end and it sounds as though the singer has developed a lisp.

BUT a note of caution I found that the Dexa's need days to run in before they lose a slight hard edge. Even though the improvements were clear to start with there was a hard edge (also a tad hard and loud) but those problems start to disappear rapidly as the Dexa's burn in. Two nights brought about significant improvement but apparently 7-10 days are needed for best results.  I got to three days before a Dexa PS failed and the sound was getting to be truly magical.

Now as mentioned one of my Dexa power supply units failed after a few days so now I am listening to Dexa NOS1 and Paul Pang pcie (whilst a replacement unit wings its way over from Denmark). Now that should make me happy shouldn't it?  after all these are the best sounds I have heard in my system as of a few days ago. Well no I just cannot get used to the drop in sound quality NOT having Dexa's at both ends. It is really hard to go back. The sound is grey and flat now. The bass is just not in the same league. I cannot wait to get that replacement PS.

With 2 Dexa I can only describe the sound as sheer magic. I have not heard anything like it anywhere. IMHO now is the first time I have really heard what NOS1 and XX are capable of. And what I hear is something I have never heard from any source anywhere - nothing I have heard comes close it sounds that good.

So am I right in what I hear? well I cannot measure anything and why would improving the USB clocks make such a difference? That would take some thinking about but all I can say is something in a USB link with cheap clocks just cannot be working well - and maybe (as Nick pointed out) when we heard USB3 sounding better than USB2 there is a clue.

A note of caution. I can only relay my experiences with my system in this house my ears etc etc maybe some proper engineering and understanding needs to be applied. Dexa's are expensive maybe there is a better cheaper answer (Crystek PLEASE make a 24Mhz crystal). For sure though this is a revelation.

Cheers

Paul

Thanks Paul, great post. How much do those clocks cost?

Thanks for the review Paul I believe you and Nick are on to something...

This is unfortunately a rather expensive upgrade. From the newclassd site the neurronstar is 430 euro and the psu 215 euro. But then again it makes much more sense to spend you x-mas bonus on two clocks than on any other part (like an expensive usb cable) if you feel to spend it on audio.

Keep the good stuff coming!

Regards,  Coen

Greetings:
Partsconnexion has a sale on the neutron dexa 24 and PS @ $399 USD per set.  I hope I'm allowed to say that?!

Regards to all

Welcome brunok! Yes, you are right about the price in the Partsconnexion: http://www.partsconnexion.com/digital_dexans.html (http://www.partsconnexion.com/digital_dexans.html)

Best regards,
Juan

That's more like it!

I can't see an orderable 24MHz clock though....


Regards, Coen

No problem with 24Mhz they are not shown on the site but can be ordered as a "special" - actually it seems they are kept in stock.

I don't know what other options are available out there for 24Mhz clocks but for sure the Dexa's work well here.

As I said above the mods are not for the feint of heart and I learned a few lessons on the way so if anyone decides to go for it drop me a PM I can save you some potential problems with a couple of simple tricks. The first would be to get hold of a pcie card that has a surface mount crystal which is much easier to remove than the bigger metal can crystals which are not easy to remove without trashing the pcie card.

Also both Nick and I have made some other mods to the USB board which Nick detailed elsewhere on the blog mainly to with distributing capacitance. So at this stage neither of us can say for sure what the mods would do to a standard NOS1 although I would have to think that it could only be good!

Also I have been somewhat reticent to post about this because mods like this should first be looked at by Peter properly from an engineering point of view and there is a very real possibility of damaging NOS1 with the modification. So even now I wonder if I should have posted.

Incidentally I have another standard NOS1 here that I shall soon be applying the clock mod to so I will be able to report on that as soon as I get another pair of Dexa's.

Last night I sitting listening to my system with only one Dexa at NOS1 and wondering where the bass went. I cannot imagine what this mod will sound like with the Orelino - a mouthwatering prospect. But come January I shall be going to Mani's to hear would the modded NOS1 sounds like with Orelino's. Now there's a thought.

Cheers

Paul

Hi Paul,

Apparently there is great merit in upgrading the clock on both ends of the USB link. I feel that the XX software based compensation of clock imperfection will have its limits and that the last mile only can be addressed with hardware.

Anyway I wondered how you connected the PSU mains and where you placed the PSU. Also the Neutronstar board seems quite large. Does it come with mounting stuts?

regards, Coen

Coen, All,

I was in correspondence with dexa some time ago to ask if they could supply 24mhz. I mentioned that there may be a few orders to follow  :)

Having used other clocks previously currently the neutron star is my favourite clock type right now hence the selection despite its price. This is due to its transformer decoupling and subjective performance. The dexa nutrino looks good also. Either should be straight forward to apply.

Be aware that there are still some small factors that need further understanding regarding the two clock set up which might lead to slight sound quality issues. I am thinking about how to address these which may change the implementation and i'll post with an approach if I can solve them.

To obtain the balanced sound quality improvement from the clocks, my view is that the mods to the usb board that I posted about in the hunting for noise thread will be needed which Paul has applied.
I had not really wanted to encourage people to go down the clock and interface modification route due to the risk of damaging a NOS1 and the  issues that this could cause for Peter. For those who intend to go ahead anyway, I would be happy to provide the advice and guidance provided to Paul on how to implement.

Regards.

Nick.

Hey Paul,

Of course this deserves a new topic. I mean, once you are ahead of Nick don't burry it in any of his.
:grazy:

Crazy stuff which I think deserves real thorough investigation. This time with the notice that when this indeed works out for the (so much) better, we should not only be applying this to the NOS1 but also to the XXHighEnd PC.

Uhm, before I forget, Nick, you as the originator should be thanked once again; but next time spend less money on cars and more on clocks in it. I recall the price of the clock in our Maserati ...
So Paul spends his money where it should go. Clocks. And without clocks no 0-60 timing either.
Btw, it is of course a secret that I have a Dexa laying around for maybe 3 years by now. Yea, wrong frequency (22.xxx).

:wacko::wacko::wacko:


Ok, I don't see how this works but something must be going on. Of course I warn for the only-more-noise and your description Paul does not take that out. But I also don't have a reason not to believe you, nor do I have a reason to doubt your ears which always are bigger. So I think it is time ...

It is time that you drop yourself and your NOS1 plus PCie card on a plane so we can compare a bit here and hook up some measurement. The best what could happen is that all shows the same (but still sounds better), not believing in that it can show better. Worse it can though and when still sounding better we should reason out why and whether we could be fooling ourselves.

Assuming for now it works out, a next NOS1 upgrade is due. That apart (and as the usual message goes) we will be able to obtain the Dexa's at a far better price (which actually has already been arranged for these few years ago).
But more possibilities exist, if only this really works.

What I really want to say is : Since you, Paul, and Nick spend this crazy time (and money) of which I know is of course for yourselves but with in mind the "for all", I should now be following up on this seriously.

So what do you say ?
If it is easy to drag Nick along, please Nick, don't hesitate. It's only that I know Paul's Whiskey habits and I will be prepared for that.

I hope that at least the major part of this post looks serious enough ...
Peter

Hi Peter - I would be delighted to come over with modified NOS1 and modified pcie card oh and of course that bottle of scotch (ha ha I knew one way or another this was going to cost me a bottle of scotch).

And Peter I understand your point that more noise cannot be precluded just yet - but in the absence of further information my ears are telling me "much less jitter" but hey what do I know?

Anyway big thanks as you say to Nick because he started all this and actually much of my system is down to Nick (including bridged GC amp).

I will speak to Nick today to find out what days he can make it and I will PM you.

Cheers

Paul

Paul/Nick/Peter,

I would be most interested if the PPAstudio card could be measured as well...just to see if there is a difference to the mobo port or the dexa modified card..

Peter, I have my oscilloscope here but I am quite sure that 0.5mV vertical sensitivity is not a low enough noise floor for the measurements you propose.  Is that correct?

Cheers,

Anthony

Anthony,

A bit difficult to answer;
If this is measured "rawly" on the USB cable your 0.5mV will probably show something but you won't be able to recognize anything of it. And, maybe in relation to another USB connection in the PC you may find a difference, but you still won't be knowing what you are looking at and what's the reference. Additionally it won't tell much because it is all how the NOS1 "rejects" it and that now is related to so many things that I can't predict it. For example, your PPAStudio should not behave differently from any other USB connection (but still it can) while I expect the Dexa solution to behave differently in the base. This is all about how the PSU relates and how new groundloops (which can be for the better !) emerge and that kind of stuff.

If you'd measure at the end and how (outside of jitter) the analogue signal comes out (the NOS!) your 0.5mV (per vertical section that is) will never show a thing and if it does, things will be not the best.
And then still you can't do much because it really needs an FFT to observe what is needed (frequencies) and if *that* shows up at the NOS1 outputs you are dealing with something like 300uV (normal is 8uV RMS).

It will be a bit vague, but if things are all OK any USB noise (internally) will be under that 8uV, while still sound is able to change drastically already by software.
Summarized : Only when we (me too) are able to dig out some frequency which normally is not there, it will be audible for sure and for the worst. I can't see a thing changing with any type of USB connection, while for example USB2 and USB3 already make a very noticeable difference.

I hope that this isn't going offtopic, but if it were for me these SQ changes are ALL because of software behavior and are otherwise (or together with) incurred for by the general current draw (say power usage) which changes drastically.
With this new knowledge I was able to create yesterday a "sound" which can plainly be seen as an I-don't-know-what-kind-off equalizer/filter which again makes your albums unrecognizable, shows the most sweet highs with a fairly high degree of lacking snap - is close up to zooming into standing waves (bass blooms more) and which yesterday played at 90dBSPL for 3 hours before finally someone told me "isn't this loud ?".

What I tried to tell with the above is that at this moment I am rather explicitly working on the electrical influence on current spikes and spreading them which in my view is similar to anything which is injecting noise (more frequencies) which spreads the peaks just the same. One difference : with the latter the "RMS" (read : average) noise level will go up, while my software means brings it down. So what I am doing (trying to do) is with some sort of Ohms law (but better : "energy never gets lost") utilizing the current peaks for the gaps in between them. Ok, this should not bring down RMS noise, but at least it won't go up as well (and spikes vanish).

That something like this is working out, is audible by the Q5=1 setting and that this really can be overdone prooves my super equalizer from yesterday.
So to quote a bit what Coen was suggesting "we now need hardware solutions" ... no. I am far from finished. Actually just started in this new dimension although it was already sneaked in in 0.9z-9 (Q5).
I am also now regularly thinking how crazy it is that a version from half a year old can today bring so drastical SQ changes if only someone tries "that" combination.

Yes, this was offtopic. But I really must add that my first USB3 PCIe card was destroyed with a 24.xxx oscillator and I never listened to the merits of it (always used the MoBo USB3) and that the second (way more expensive) card never found its way to the PCIe bus because I personally and 100% think that *I* am not up to that at all. Thus, as long as software needs to be explored way more and can lead to so much better suddenly, why would I (anyone ?) try to improve with hardware. And I am serious : this hardware is more out of (my) control than the software plus the software can influence the hardware (mind you, also how your disk behaves, just saying).

Yes, this is quite another post than my first in this topic.
So I must be honest :
My previous post was merely and finally filfilling up to recognizing a two guys needs on very long and hard and $$ work on improvement. This post more honestly shows what I am really thinking and that -for example- there is no single way that I am able to judge in my current new means how all the *other* software parameters can influence this particular setting. Arjan pointed out the 5ms ClockRes instead of my 0.5ms. I tried his 5ms (with another DAC of course) and it didn't work out. Still it brought me back to 1ms which made me scratch my head at thinking "yea, start all over - takes a year".
With yesterday's new application I found the sound very similar to W8.1 and that again made me think that I may have found some ugly W8.1 base to possibly undo and proceed with that again.

Last notice and to be on-topic again :
If it were for me, not any low jitter USB clock can improve on jitter with asynchronous USB. But and as I said elsewhere, maybe it can after all. It is, however, a 100 times more likely that it is just that other noise pattern doing this (what about very low oscillation because of two now very closely running oscillators at both ends). So, it sure all is about jitter, but the means to it (better pattern up to lower jitter) should be very indirect and should not be about low jitter oscillators in the first place (but mind the low oscillation because then it is after all - hence it is more direct).
But if it works, it works. Sadly also : if it works, what happens with one of the drastical changing software settings.

All together (and offtopic again) I don't think I am up to hardware changes - if that was not clear already. And you know, for me it is really difficult to even stuff in a PCIe USB3 card when I don't see any need anywhere. Not as long as I know how much just putting in one additonal disk makes a difference and my perception of how the last one to remove should do jobs. Then I rather work for a year on the latter than spend 3 minutes on the former, in the end also knowing that all of it (whatever it is) in the end hammers on this interface.

Wow, I talk too long / too much. :sorry:
Peter

PS: Paul, Nick - still invited. But know what you're up to.

Thanks for the elaboration Peter.

The clock experiments hold a key to better understanding what makes good sound in a computer audio setup. Since it seems that many influences are somehow powersupply related it raises also the question what the major contributor is: the precision clock or the PSU (as far as they can be seperated influences).

I am pleased to read that there are healthy theories popping up that can be scrutinised on a software level. This also promises great progress still to come. We are not near the point that we Háve to resort to hardware upgrades at all. Still these enable you the enjoy today the benefits that are yet to come ;).
Anyway this process also relies on the bold experiments of our most commited forumfellows, so keep up the good work and enjoy your results!

Regards, Coen

Yes Peter, thanks for the elaboration.  Very interesting...I have read your post three times now and I think that I better understand the angle that you are coming from.

Yes, of course the whole dexa clock experiment is likely to be substituting one set of noise for another set of noise, and the consensus of the two proponents is that the new set of noise is more pleasing to the ear.  This makes me wonder even further just what impact the ATX linear psu experiment will have on the entire XXHE/NOS1 ecosystem.  Will a new balance be created or will we prefer the old set of noise?  As you explained in that thread the lowest noise does not necessarily make for the best sound if we are exposing more offensive noise from within the pc.

So many combinations and permutations...and I think that you are right Peter to explore the software fully before finally tweaking the hardware...but in the meantime Nick and Paul have to their tastes the next level of sound reproduction and I am barreling headlong into a different major hardware change with the ATX LPS.  Heady times indeed.

Cheers,

Anthony

Just a thought:

You can reduce the number of clock variables in the system by slaving one of the USB ends to another. Iow like slave the PC to the NOS clock. Wether this one clock approach works out for the better remains to be seen. This seems to work for recording studios, but then again that is a true synchronous signal. Anyway the low oscillation will not be there.

Regards, Coen

Wow.

Coen, you are a genius;

You just brought me THE idea which is so so SO smart that I can't believe it myself. And it has to work.
OMG, if this really works then it probably leads to something so devistatingly new that all SQ goes to another level.

:secret:
Peter

Coen hi,

When I mentioned the "issues" that still have a possible effect sound quality (above), relative clock speed is one of them. It might even go so far as one end of the link being better running slightly faster or slower than the other. This might point to PPL sync / recive buffering being a factor.

Slaving both end is one of the things I am trying amongst a couple of other approaches.

The only way to fine out is to give it a try  :)

Regards,

Nick.

Our posts crossed Peter. Im on it  :)

Peter,
I just re-read your post I think now you may mean something other than clock slaving ?

Regards,

Nick.

Literally yes I mean something else. Functionaly no.
:secret:

Wouldn't it be good if there was only one clock for the 2 USB (emitter and receiver)... It would even simplify things...

Alain 

Yes ...

:secret::secret::secret:

 :blush1:


Regards, Coen

First feedback,

Trimming the speed of pc end usb clock.

I was speaking to Paul earlier about the slave clock setup. This reminded me that the sound characteristics I have here now are similar in some respects to a modified m-audio transit card I had years ago (there is a post about the set up on one of the boards). The recognisable sound between the two system arose from trimming the clock speed on the m-audio transit interface (it had an audiocom superclock 4 fitted to the usb interface).

5 min and the trim pot on the dexa clock of my pc usb card clock was trimmed by ear and the same control over sound emerged. One adjustment extreme gives emphasised sharp edge and slightly splashy sound the other super super smoooooooth and clear detailed sound. In between comes a double sized sound stage pure pure detail ultra real just there in front of you sound. Exactly like the m-audio sound but sooooo much better !!!!

The trim pot is hence forth to be know and the "heavenly trimmer", haha.

I love this sound !

Slave (single) usb clock results to follow. Its going to be interesting to see if a set difference in clock speed is better or worse than the same clock speed at both ends of the usb link.

Nick,

Ps variations between systems / usb cards might be in part down to luck of what the relative speed of usb clocks at each end of the link happen to be........ there is a thought. ...

Great work Nick,

Now that was a superfast practical experiment!

If you hear such differences in sound whilst only offsetting the clock a littlebit, it makes sense that we might be lucky or lost either way with standard "large-but-sufficient" tolerance clocks. This also means that our milages will vary and we could have quite different 'objective' sonic experiences with the same hardware...or different hardware.... ;)

Please elaborate on your one clock solution when you get it done. I gather you have to arrange for some kind of clock distribution that is least detrimental to the clock signal synchronicity.

Regards, Coen

Ps you beat me on this one!

I'm sitting back with the metaphorical popcorn waiting to see how this plays out.  Keep it up guys!!

Peter has gone ninja on us with this post...


If Peter was Australian and our humour was in play I would not know how to take that post...it is either deep sarcasm or genuine excitement...I hope it is the latter.

Anthony

This description makes sense. The offset increases (hf?) distortions on one end and filters the sound on the other. It makes one wonder how complex this gets when serious jitter comes into play....

Regards, Coen

Hi Anthony,

Imho this is serious exitement on Peter's part.

I am the one that did 1+1=2, Peter is the genius who knows like no other what its implications are.

But I think we're catching up :grin:

Regards, Coen

P.s. Enjoy your "popcorn" !

I am sure you are right Coen.  These little cultural differences are funny at times...and I am enjoying my popcorn.  The cricket is starting so it's time to turn off my music and put on the radio (which sounds awesome through the NOS1...hehehehe).


I am watching this space.


Cheers,

Anthony

No Frank but those Kiwis are certainly showing the Windies how it is done.

Nice morning for Aussie cricket though...

As I said or implied or whatever, the two clocks running on an ALWAYS different speed is the importance here. Jitter gives the noise coming from that another pattern, or IOW oscillation (resonance) varies (at high speed).

When using one oscillator this just can not happen. And the thing can be as jittery as it likes. Well, this is what I think so far; can be proven to be diferent later.

Peter

Nick,

Yea, nice tuning facility you made yourself there. Of course I did not know this (trust me) but as often things seem to come together by again very different routes; you just prooved in advance that this is going to work (one oscillator). Still Coen has the credits for my part.
Well, assumed that is able to give the best sound which (of course and sadly) depends on the "pattern" created by "nothing" (no oscillation).

My part of the game here is the means HOW to realize this one oscillator. THAT will be the new stuff ...
So Anthony, no sarcasm at all here.

Regards,
Peter


PS: I think I am going to move this topic to the NOS1 board. That should make it serious, right ?

That does sound serious Peter.

Ha ha isn't that typical of my luck (blown Dexa power supply) with only one clock working I cannot even play with my "heavenly trimmer". I wonder what my wife will make of that.

This is a really interesting stuff - we could soon know whether noise or data transfer errors are the culprit. We all know what Peter believes and my initial thoughts were that it could not possibly be data transfer errors could it?

I just cannot get used to this step backward in sound quality - I only heard two Dexa's for a few days - where is the bass, where is the crystal easy clarity?

Can't wait to get that Dexa PS due Monday ggrrhhh!!!

Paul

Its a cruel drug not to have  once tried :)

The heavenly trimmer  :) could still work with the paul pang card. Just remember where it was set in case it needs to go back. It might be an interesting result.

Best Nick

So Peter - Just a thought  -  with a Dexa at the NOS1 and a standard pcie USB3 at the other end - if I tweak the Dexa "heavenly trimmer" and I am lucky enough to be able to match the Dexa clock frequency to the pcie clock frequency then in theory it should sound as good as two Dexa clocks if the noise theory is correct?

Paul

Paul,

The jitter won't be the same and they have both different tempraturesensitivity (and different temperatures during the year).

But it is closer to ideal than having an unoptimised clock sync.

Regards, Coen

Paul hi,

Just to clarify, that is what I'm suggesting above, use the dexa trim pot. This is the set up I had years ago with the m-audio card and audiocom clock. You will just be setting the best sounding frequency on the dexa relative to the Paul pang card.  I could make a big difference. Listen for the sound stage expanding whilst keeping good dynamics and no sharpness.

There are a number of ways of thinking about the effect of relative clock speed. I won't go into the details but equal speed clocks (or slaved) may not be the best setting. There could very well be an optimum difference in speed that suits the data transmision rate and the buffer size at the reciving end.

I think its imporant to understand the relationship so i have equipment on order to test explicitly these theories. Will post back with results, then we will know of the trouble of transmitting a clock 1.5m whilst maintaining clock performance is worth the effort.

Cheers,

Nick.

Hi Coen,

Appreciate about the jitter not sure about temperature as that is a longer term drift effect so for a quick check should not be a problem. Anyway I shall have a play later on.

Nick apologies missed your post - look forward to seeing the results of your tests.

P

Paul hi,

Do give it a go, it's the trim pot next to the xtal can on the dexa. It's a variable cap with one full turn of adjustment. If the speeds are out it could REALLY make you smile (the 5.1 type sound stage is back again here  :) )and it takes moments to adjust.

Cheers,

Nick.

Hi Nick,

Trying to read about the "master-slave" relation between two USB ports... Is it something that is established normally between the sender and the receiver in the PC USB port and the NOS1 USB port or is it something that has to be "programmed" first ? If so, I would have thought that the sender would be the master ? By that I mean the "A" type USB as the master and the "B" type USB as the receiver ?

And would this achieve what you, Paul, Cohen and Peter are talking about ?

Regards,

Alain

Hi Alain - NOS1 is "asynchronous" but the way the data is transferred is maybe not strictly that. I think (and I am sure Peter will correct me if I am wrong) that when NOS1 is ready (i.e. its receiving buffer is running low) it requests a block of data to fill it ("bulk data transfer"). So Nos1 is the Master and the PC is the slave. That is why on the face of it, it is hard to understand how jitter in the USB link can be a problem because (I think) the data is transferred with timing information and so should be immune to the effects of jitter. I am sure that is why Peter thinks this is all about noise.

Another possible way of transmitting over USB is Isochronous which means that the data is transmitted equally spaced - but I am pretty sure that is not the way NOS1 works and I am also sure Peter will very soon correct me if I am wrong.

Cheers Paul

Maybe a few scattered responses, with for Nick the message that I did not read your lengthy email yet. But it makes me more objective.



Maybe not all the way through on the subject, but the Dexa is a TCXO or IOW temperature controlled. Of course the PCIe card is not, so what you say is correct in the end. With a Dexa at each end this is different though;

I think Nick already said it, but we won't be able to exactly "match" anyway. With real slaving (VCXO - Voltage Controlled) this is different, but still at a low frequency (like a few times per second) the oscillator at the other end will be corrected (adjusted in speed to match the master) and for NOISE your miles will vary as much - but probably things will even get worse. This is about the low oscillation (low change of frequency) which now also will be completely repeatebale (each second the same pattern repeats). Thus, in my view haveing them both much off related to eachother would be the better thing.

Of course this is all theories and how the real work out will be is to be seen (heard).

This is how I think that NO oscillation will at least imply a steady situation which we can depend on, and then still it is to be seen whether it is for the better. This too might not be for the better at all, because now we don't have nothing to trim ...
Also (and once again) the noise patteren there will conists of less freqencies (nothing runs into eachother) so it could be worse to begin with.

But I am going to try and I know it can be setup.

Must go now, but will be back.
Peter

Nick,

As long as we realize that you are doing this with a 600 euro device on one side and a relative el cheapo on the other, while all it takes is adjusting one of the clocks. That either (actually the master) should be fairly stable is something else but that does not take 600 euro. No separate PSU either.

I am not stating this out of all context; I only say that when this works it needs a low ppm oscillator as master and a VCXO as slave. WHEN this works.

For others : A stable oscillator like in ppm terms (parts per million) is nothing like a low phase noise (low RMS jitter) oscillator or clock setup. To this regard the Dexa is only a low ppm device (long term stability) and isn't about "low jitter" as we know it for audio.
Of course this is a totally different subject and all I say with it is that the 600 euro is crazy if you ask me.
Nick, until someone comes up with phase noise plots (and for us both not anything new).

What I am really saying is that - assumed we are heading for the better solution with some sort of "trimming" is that any low cost oscillator can be used. And with that : That it is quite useless to have it all perfectly working after maybe several trials and much listening, and then all we know is that it costs a 600 up to two times of that. To me it (now) looks as starting at the wrong end as long as these (new) theories are assumed to be valid.

Peter

A litte more reasoning (only that guys !) :

Assumed your ears are still very OK Paul, this (see quote) should tell something;
IMO it very well can be that the very stable Dexa's (at both ends) - that's what they are - do not even create an out of sync often (this in itself leads to other theories but never mind that for now). So, they may not run at the same exact speed, but the time time period needed before two up (or down) going slopes miss eachother may be very long. Just think 24M pulses per second, and next look at the ppm rating of them (I don't know at this time) and it could take hours for things going wrong. Don't ask me what happens when it *is* wrong but now I need to look into how async USB really works and how things which needs to transmit are once per x milliseconds in the first place. It may not even play a role (because all covered for in the protocol). However :

What I wanted to say is that envisioning the two clocks just running behind eachother, their both pulses implying noise, that noise pattern will always be the same *because* they are so stable. But what it would need is
a. getting them on that same speed;
b. find the pattern which suits you (sound wise).

But do you see what happens when this is all working like this anyway ?
Once you found the speed matching, it needs slowing down one of them and bring it up to the same speed, only to find that other pattern.
You can't have done such a thing.

So next theory :
They both never run at the same speed, but both are still as stable. This is now about the low oscillation I talked about, and this is instead of the more random wild influencing thing (the one clock catching up a full cycle many times per second). Now, the pattern will still be important but is out of our control, but the pattern has I think less wild excursions (very high peaks of coincidentally same pulses at exact same time). Still it should be so that once in a while this same excursion happens, and possibly it is audible through a kind of flanger. It could widen (or deepen) the sound stage if often enough (like maybe 10-20 times per second) and now it even can be so that the RMS jitter we do talk about with audio also plays a role;
Now suddenly the *high* (short term) jitter of the Dexa (hey, that assumed) plays a role because it will spread those peaks coming together for a longer time (because they are so long term stable).

Anything else ? :innocent:
If this were all true, it needs a high long term stable device (thus TCXO) with a cheap high jitter oscillator in there.

or ...

A high jitter oscillator at one end, connected to both.
Yeah, made this one up because it follows from my own logic.
Ultimately stable, spreading ... hmm ... its own noise pattern ?
Not sure about this latter.

Peter

Peter hi,


The Paul Pang card has a TXCO and the Dexa is temp controlled (not oven but 37 deg constant). The Spec that Dexa quote for stability is 0.2ppm which is OCXO territory so I think there is a good chance that Paul can get close enough. I'm just suggesting that Paul gives it a go just to see what happens.

My context for this is that the last time I was doing this with clocks was back in about 2008/9 with a £52 pound M-Audio transit interface with an Audiocom Superclock 4 fitted (not in the same class IMHO as the dexa and the NOS) and the results were very good and repeatable when tuning the M-Audio clock against a stock PCI USB 2 card with a standard Xtal fitted. I don't think we need to be too hung up on precise cock speed matching, just tune my ear, believe me you know when you reach the sweet spot  :)

Some nostalgia

http://www.phasure.com/index.php?topic=979.msg12931#msg12931

I agree the Dexs the costs is a bummer but the based on subjective listening its just the best clock that I'v listened to to date, and MUCH easier to apply consistently without ground loop problems than direct coupled clocks (these have been a problem for me in the past) which is convenient for DIY. I don't know what the Phase Noise Spec is but they do sound very good. I'm not suggesting that anyone else uses them, they work well but you might have to be a little mad to have the three that I have in my system :grazy:

I am with you about the intermodulation of the two clocks being a possible problem. I lost a post earlier (dammed ipad battery) about the clock speed and thoughts on possible effects of the clocks on data and SQ. In essence i'm not really seeing this as an electrical noise issue. My money is on data transmission error rates being vastly improved and if USB "bulk transfer" mode is being used to transfer data then the reduced error rate will reduce the number CRC errors and so the number of "Stalls" and retransmissions on the link. Of course this assumes that the stalls disrupt data processing further down stream but this seems possible. I think the USB receive buffers on the NOS USB chip may also have a role in the effect of clock speed mismatch in that tuning the relative speed of the clocks might mean that there is less data transmission speed management needed between the ends of the USB link. The event above only have to happen at audio frequencies and its not a big jump to think that this is going to be audible.

All theory, I guess the only way to prove this would be with a 480Mhz USB analyser to watch the USB Protocol error rate and frequency of dataflow management packets to see if they change with better clocks and lock speed matching. But such analysers are big bucks and really whilst I would love to know what is happening the resulting sound may have to be enough.

My thoughts at the moment on clock speed are that if clocks running at the same speed (slaved) turns out to be the best set up, then this will have a very nice simple elegance to it. I think this might well be the best solution but part of me is still thinking that, taking the USB receive buffering into account, it might be that a slightly faster NOS clock could be the sweet point. As mentioned above I'v ordered some test equipment to try to pin what the best solution is. Meanwhile the slaved clock build continues just to see what happens  :)

Regards,

Nick.

Yes Nick. I'm sure you didn't read my attempt to start this subject while posting your last post (see my previous), but this is another area which could be more happening than we like. But I don't know really.

Btw, I plan to have my "link" running next Monday.
If I manage to read your email before that. Haha.

Best regards and of course thanks,
Peter

Your right Peter you posted whilst I was drafting. I think we have similar ideas. Wouldn't it be nice to have access to USB analyser just to know one way or the other.

Great that your link could be up by Monday. I think ill back off on my build and see what you make of it first. Fingers crossed for a good result ! Do let us know  :)

Regards,

Nick.

So with Dexa in NOS1 and Pang pcie at the PC I adjusted the "heavenly trimmer" as far as I could but I can't honestly say I can hear any difference - just now it is hard to tell if it is having any effect at all.

Maybe the Pang card clock is not close enough to the Dexa clock in frequency.

Paul

Interesting in itself Paul;
Your guess is probably the best bet.

But I must say that by now I'm a bit lost on your and Nick's situation; Wasn't Nick doing the same ? I mean, he is not the one with two Dexa's (24.000), right ?
So what is the difference ? Or maybe Nick doesn't have a Paul Pang card ?

Regards,
Peter (sorry that I didn't read through all again to find out)

Hi Peter,

Here I have 3 dexas, two 24mhz for usb and one 24.5mhz for audio stream. No Paul Pang card used here, prior to the dexas two vcxo 24mhz diy built clocks were used to test the approach using transformer coupling before fitting the two 24mhz dexas when I was sure that a reference grade clocks were warranted.

Here trimming of the usb clocks is carried out on one of the 24mhz  dexas which happens to be at the pcie card end of the link. It can just as well be at the nos end dexa as the result in the same.

I'm not sure why Paul is not experiencing change but my initial guess would be, as you both mention, that the trim available on Pauls dexa is not sufficient to match pauls one daxa with his Paul's Paul Pang card.

Regards,
Nick.

Paul,

Good to give it a try anyway but a distinctly "average trimmer". I hope it might go better when dexa no 2 is back in place.   My guess would also be that the gap in speed is too big to "match" well. Thanks for coming back with the results, it all adds to the picture of what might be going on  :)

Nick.

Thank you for clarifying Nick.


I think I saw you writing this more often, but (and not important) I think you mean the 22.xxx MHz here ?

Peter

I recently read about a genius idea for clock distribution that I like to share.
Credit is due to the guy from ecdesigns who has developed a 44.1/16 SD card reference player.

The principle is that splitting a clock signal and distributing over a distance it will degrade the clock quality and introduce noise. The strikingly simple answer is to filter the incoming distribited signal with an chrystal filter, which has a very narrow passband letting only the original signal trough. In his wording this filter acts as a highly specific "tuning fork" requiring little energy to resonate and as such providing for a light load tot the master clock generator. One can use this principle for any clock distribution ánd/ór transport.

How about that!

Regards, Coen

A passive repeater eh ?

Thank you Coen !
Peter

Guys, I can use a little help - just making some short cuts because of a lack of time to sort it out decently at this moment;

So, I have my super sneaky connection running - partly. Still in all over secrecy think like I have such a NEC based PCIe card tweaked with the direct clock connection (card's side is the clock). That card does not run; it does per any normal means of connection (and without direct clock connection).
I have another of such a card (different brand). That *does* run per the special connection, but that is not the tweaked one.

With "run" I mean that the PC sees it. So "not run" means the PC does not see it.

Many variables are there to change, but too many (and all too fragile) to start without more knowledge. One of the options is that the card which does not run now needs to have the Molex connected (hence needs that separate supply). With the normal connection it does not need that.
The card which runs does not have the separate supply connected, but also no clock connection; this can matter.

Now finally the question :
We always seem to speak about the card needing 12V, or I misread or just dreamt that. Anyway, I did supply the 12V with a sufficient amount of Amperes;
Does it need 12V or does it need 5V ?? ... or both ?
Watch out : I say "need" but I already know it needs neither (from the Molex). So it is merely what it utilizes once the Molex is connected (both 12V and 5V being in there). The "utilizes" comes from the special connection if the card thinks so, so to speak.

My short cut is about that it is not so easy to for fun (and no avail) give it a 5V as well with sufficient juice. And so I like to know it in advance, if anyone knows the answer anyway.

And FYI : I can not understand why the card is not seen by the PC just because of a possible failing clock connection, IOW that should not be it.
Of course, my fastest option would be to hop over the clock connection from that one card to the other, but as said, this is all fragile and I like to stay away from it when possible.
And again in other words, if someone tells me that 12V is what's needed, then I'll hop over the clock connection (because 12V it has now). Or when 5V is needed I change the PSU. When both are needed, I'll add another (also not easy for reasons and by then I'll have Anthony's LPSU mimiced - haha).

Ok, maybe this was a combined question with status report; all looking good, but a step which should work fails at this moment.

Great thanks for any insight,
Peter

PS: Apologies that I do talk in secrets indeed, because it is too much fun what I'm doing, and when it fails I had rather not tell it. But I will when it works !

Hey Peter - I am not sure if this helps but it might. The Paul Pang pcie USB3 card is a standard NEC based pcie card with a piggy back board (added by Paul Pang) that has the low jitter clock on it. On both the main board and the piggy back board there are regulators that get warm each with a lump of Al glued to them - certainly added by Paul Pang. The card has to have a Molex ps  or it will not work and it only uses the 5V supply on the MOLEX - mine only has the 5V connected so I know it does not need 12V.

That might indicate that the standard pcie card does not need 12V - maybe!!.

Paul

Thanks a bunch Paul !
Peter

Peter hi,

Just to add to Pauls points.

Most cards will use the 3.2v pcie bus voltage by default. In this situation this bus voltage is used to power the nec chip and via a dc to dc converter the the 3.2v is stepped up to 5 volts to power the aux usb supply (5volts carried via the usb cable)

With the molex I think only the 5 volt supply is used. I know it will be used to supply the 5v aux supply (the enable pin on the dc to dc converter above is used to switch off the 5v generated from the 3.2v pcie bus). I am not sure if the molex 5v supply is used to power the nec chip, I suspect not.

One thought about the card that will not work. Is the clock live signal connected to the right xtal pcb pad. There will be two traces from where the old xtal is mounted to the nec chip. The correct connection usually has a series resistor on its way to the nec chip. If in doubt it should be ok to just try to solder to each pad in turn (this is what I did on the nos pcb to work out which xtal pad to inject the signal. If the nec chip has no clock signal the pc will not see the device.

Apologies if you already know some of the above. Hope it might help and best of luck.

Regards,

Nick.

Ps I lost another post yesterday. :(
Im reasonably sure now that sound quality is dependant on both absolute clock speed AND relative clock  speed. There are some other effects in play as well which ill come back on later but tuning the usb link clocks appears to have more variables that need to be understood. The good news is that this is about getting the last 20 to 30% of performance on top of the really significant improvement that just putting the clocks in provides. 
 

Maybe Paul Pang is using regulators to derive 3.3v from the 5v rail hence the chunky heat sinks - dc-dc converters could be expected to significantly more noisy.

P

Guys, those bits of aluminium on the Pang card are not for heat, they are for rfi.  He glues those pieces on to components that may be impacted by or radiate the nasties.

Anthony

Ha ha the replacement power supply for my Dexa arrived today. What a relief to get two Dexa's back into service. Oh my the sound quality is just stunning with 2 Dexa's. The fantastic bass is back and just as fantastic everything else.

Anyway now that I have two Dexa's again I have been playing with the "heavenly trimmer". And this time the effect on sound quality is unmistakable. From the as delivered setting a quarter turn one way makes the sound softer. A quarter turn in the other direction again seems to make the sound softer but also a refined sound. After just a short time it is not possible to say which sounds correct or even the best - it does not sound clear cut.

Paul

Anthony - thanks for the clarification the one that I did get the back of my finger on was quiet warm hence the erroneous assumption.

P

Paul hi,

That seems like a better way to do it I guess on Paul Pang's card.

Sorry I realised that I said that I was adding to your point but actually my comment above were about the molex on "vanilla" USB cards, sorry if I did not make that clear. As you know I have only seen your Paul Pang card for a limited amount of time so I'm not really able to say anything about the supply of the card.

Best,

Nick.

From what I can tell the modded Pang card uses the external 5v supply to run only the upgraded TXCO.  The remainder of the card runs as normal from the pcie bus.

Paul,

Fantastic !  :) normal service resumed.

Interesting to here that the trimmers do have an effect now. It took a while to work out what they were doing because most settings seem to have qualities that are appealing in some way, so I agree that it is not so clear cut.

The sound I'm homing in on has a nice balance of dynamic, tone sound stage and emotion. It seems to be just moving from the slight sharp setting to towards the smooth setting on the trimmer, just so that the edge disappears.

It might be worth having a chat about this, as mentioned above I think absolute speed and relative speed of the clocks seems to have an effect so it would be good to bounce a few ideas about and check and compare results.

Speak soon,

Nick.

I can confirm my PC does not recognise the pcie card when there is no clock in it (when the crystal is removed) or when the Dexa signal is not switched on.

I have another problem and that is when I reboot my PC with the pcie card clocked by the Dexa (and the Dexa switched on) the Bios does not fire up correctly - I always have to reset the Bios just as the reboot starts and then select default Bios values - then and only then will the PC boot. The output of the Dexa clock to the pcie card is completely isolated as it comes out of an isolating pulse transformer. I have replaced the BIOS battery on the mobo earthed the Dexa card to the PC but none of this makes any difference. Also I powered the pcie card from a molex supply from the ATX but again made no difference.

If I remove the card I can get the PC to boot properly.

Any ideas Gents would be appreciated.

Cheers

Paul

Hey Nick - I did exactly the same and just moved slightly away from the point where the sound had a slight edge.

I was expecting that maybe the ps would need some running in (and maybe it does) but the sound is really smooth and just effortlessly glides through difficult pieces. String quartets are sounding wonderful.

P

The latest nec chip runs on a very low voltage (like 1.3V or something). Like a pc motherboard this voltage is made from probably the 3.3v by a switching regulator but I did not measure its source. The switching regultor is on the pp card on the right of the nec chip.

Pci-e has no native 5V connection, so that has to be made on the board or by an external supply. In case of the pp card this is the only function of the molex. The voltage is filtered by an rf pi-filter with a "big" rf choke on the top part of the board.

The card designer has two voltages available on the pci bus to make the 5V voltage from: 3.3 and 12V. I guess the 12V will be preferred for the 5V USB power if no external supply is available, at least that makes the most sense to me fom an efficiency and power availability perspective.

Anyway you for shure don't need the extra molex for the 12v and in case of a molex, likely the 12v is not used at all. Too bad you can't prevent it from entering the card since its also on the bus ;).

Regards, Coen

Paul,

You are probably violating the startup requirements with a running clock on the card before power-up. This could be either a bios or a pcie card startup problem. My guess is the latter.

I would suggest you use a card that works for others. Nick?

Regards, Coen

Hi Coen - This is my second pcie card both have exhibited the same problem. What I will do is try Nick's he was expressing an interest in coming down here soonish. Nick does not have this problem.

P

Just a point about the "heavenly trimmer" the point where the sound has a slightly hard edge to it is where the Dexa cards trimmers were factory set. I would have to assume that the Dexa's are set very close to 24Mhz in testing simply because they can be!! So maybe the slightly hard edge comes when the two Dexa's are very close in frequency.

Nick is that what you found?

Wish I could measure it - Nick presumably you could see the two clocks on your funky scope?

Paul

Paul,

Bingo my thought too ! :)

 This is one of the reasons I have been saying that I think clock speed faster than 24mz may be needed. My scope will not measure to 8 digits but something that will is on the way. Im really wanting to understand the variables,  I think the understanding could really help tune the usb link .

Cheers,

Nick.

Super stuff again lads !

Paul, great that you have the PSU back and can use your heavenly trimmer again. How does your wife feel about it ? ha ha

But special thanks on ideas about when the card is not seen. This will be the most helpful. Will proceed on it today.

And hey, no heavenly trimmer here. It's on viagra all the time you know. Fixed Value. 8)

Regards,
Peter

I just modded my paul pang card a little and ran into an odd situation.

What i at first did was connecting the secondaries of the hf trafo to the PCB x-tal and PCB gnd at the removed capacitors respectively.
The result: no card detected.

This got me thinking: the original capacitor on the secondary could be necessary in the end to block a dc component at the NEC chip side.

Reconnecting the capacitor made the card work again. So that makes me wonder how the DEXA clocks work. They must have some capacitive decoupling on the pcb to avoid this situation. Or maybe this only holds true for the way the pp usb3 card is working.
Nick, how did you connect you previous clock experiments clock signal to the usb chip?

Anyway, the sound has more bass and naturalness than I can rember. It is also a bit too laid back and soft for my taste. This could just be the clock difference thing here.
Unfortunately i have no heavenly trimmer available.

Now lets see what the one clock thing brings. Suggestions how to get this working with a minimum of effort are welcome. (Does the NOS1 usb oscillator also have a dc component?)

Regards,  Coen

Coen hi


My DIY clock used capacitor decoupling on both output of the clock module in serise with the transformer primary coil and also on the secondary of the transformer on its output in serise before the signal connects to either the NOS PCB or the PCIE USB card.

I tried direct coupling of the clock module to the transformer primary and the transformer secondary to the PCB clock inputs but this resulted in a loss of signal amplitude and would not drive the USB PCIe card and NOS PCB IIRC.

Hope this helps,

Regards,

Nick.

Thanks Nick, this is helpfull!

The signal loss would be due to the dc component saturating the transformer core.

Next to providing galvanic isolation, which the capacitors allready can do if you would add one on the other leg, I think the benefit of the transformer is that it acts as a band pass filter. by its nature it removes the steepness of the signal. This leaves a signal with less unneccesary hf energy that would be detrimental to the sound.

Regards, Coen

LOL

Coen, aren't we getting crazy (by) now ?

I must say that I have applied similar elsewhere by means of lowering the (official) voltage. So if you are crazy I am the same. But careful now, because while my reaons were exactly the same (hey, we do get along, don't we ?) in your case there's theoretical jitter in order, while in my case it was unrelated to that. So Coen, keep in mind - we are on the track of "no jitter can be in order here" but from the other angles (stability - spreading peaks) we might be after all and actually it is my best bet at this moment.
Btw, spreading peaks sounds nice as a phenomenon, but that can be in control, while your idea here will spread too, but now out of control. So there is a didference.

So I am sure you see that flattening signal tops is the worst for jitter and it should be the other way around, and I'm just saying it because it may have skipped your (net) consideration.

But once again : great stuff in itself.
Peter

I am glad I am not the only one with the most easy appliances not working at all. So I did not tell it yet, but the most basic connection in my special situation already does not work (clocks normally running on all sides). So I am as far as no NOS1 to be seen anywhere.

You guys should see what's all done to deteriorate further and further to come to the conclusion in the end that the base "configuration" even doesn't work.
The only thing what keeps on working is that my PC keeps on booting. Ok, once out of two times, so that is progressing also.

:swoon:


PS: We just moved the lot out of the listening room, Waiting for you now Coen. That's more easy. Too many variables here at this moment, while I apperantly seem to think that all will work but maybe one, while it is the other way around and I will not know which one.
LOL

Coen hi,

I am not aware of the spec. of the tranformer used by the Paul pang card so difficult to comment there but concentrating for a moment on my DIY clocks build and the Dexa clocks. The main application of the output transformer used in both of these is for ethernet network transceivers signal isolation, bandwidth is rated >100Mhz. I guess to preserve the signal waveform the transformers effective maximum pass band will be much higher than this.

What has struck me with the experiments on USB clocks so far is how much more that jitter reduction seems to impact on sound quality than clock RF impacts sound as HF components of square clock waveforms increase.

Given this my thinking is that bandwidth limiting the clock waveform might improve general electrical noise but is likely to also impact phase performance which in turn could have a big impact on sound.

Im still very keen on clock transformer decoupling as it is appears to take ground loops on the clock connection out of the picture which can otherwise be a big problem for sound quality. But generally it think high bandwidth passive components on the out put of the clock may be the best way to go.

Kind regards,

Nick.

Quick question :

Can anyone (or everyone) describe what happened to the sound when Paul Pang's card was used instead of a normal NEC based USB3 card ?

I'm into the 5th track of "some" setup and ready to post about something strange and that maybe I found something ...

Peter

Mmmm  as it happens I have one of Mani's NOS1's here (totally standard no mods) so I can compare it with my modified NOS1 with Dexa's. I can try any combination (1 Dexa, 2 Dexa's, std NOS1, modified NOS1 etc) except I do not have a standard Pcie USB3 card - both mine have been hacked so they no longer have crystals in them one of them is being clocked by a Dexa.

So for various reasons the only tests I can do are against mobo USB2/3 which I can turn on.

Is that any use to you Peter?

The only other thing is I have to re tune my ears because I have been listening to a number of combinations and I am starting to show the first signs of madness (OK maybe not for the first time.......)

Oh and of course I have a Paul Pang card in and working...

Cheers

Paul

I have not done the comparison with XXHE/NOS1, but when I did it with my previous dac and computer setup the Pang card brought better resolution across the highs and low but most importantly it brought a more balanced presentation i.e. there was no bloated bass or shrieking treble..just balance, and smoothness.  But as I said earlier, that was with my previous dac and computer.


Anthony

OK using Mani's standard NOS1 I have been listening with it connected to a mobo USB2 port. Both USB2 & 3 on the mobo switched on. I guess there is no need to describe what I hear because most NOS1 users out there should be familiar with the SQ.

Next I switched over to the Paul Pang card - there is a clear difference in sound quality. The sound becomes "fuller" less thin definitely a more "analogue" sound. Perhaps "Lush" would be a way to describe it and at first listening I would have said that it was tonally better - but now having heard the Dexa USB I am not so sure maybe it just sounds "fuller".

However, with the benefit of now having listened to the Dexa clocked USB link it is clear that the Paul Pang card does not improve on the clarity of the sound. But it certainly changes it.

Hope that helps

Paul

With the standard pp card I posted before that I felt it was delicate and natural sounding, but with less drive, less highs and especially less bass.

That is compared to my standard oem usb3 pcie card (earlier version of the NEC chip with onboard switching regulators for the chip and 5V). Reversely this sounds more loose and melodious, together with more realistic highs and dynamics.

Now with the little modification (secondary grounded near NEC chip instead of clockboard), I concur with Paul. i have to add that the sound is still a bit less dynamic (smaller?) and it seems more tilted to the right channel. The latter could be or is a frequency balance left-right thing.

Is this the kind of report that is addressing your question?

Next I plan to do some psu connection experiments, before taking the plunge to a one-clock setup.

Regards, Coen

A lttle update:

There is a significant difference in sound between the two USB ports. One (furthest from the mobo that I use normally) sounds round and a little thick and the other sharp and a little restless.

Furthermore I used a MOLEX to SATA power adapter cable to harvest the 5V and Ground (red and black wire) and soldered them directly to the clockboard supply, leaving the PP wires in place for the USB 5V.

Now this little change immediately sounded like an improvement: more looseness, natural detail and melody. Though this is a little hasty conclusion, I will give it some time and come back later wheter I still think it is an improvement.

Regards, Coen

Hey, thank you all (three). So now it is my task to make something out of that. Haha.

My own reference : Standard MoBo USB3 plus one night of listening through some NEC card the other day where I didn't really notice anything for differences. Maybe there was, but I wasn't looking for something and at least nothing occurred. No 5V.

But now last night ...

I used the other card I have (22 euros or so and with 4 USB3 connections) and thought some hell had broken loose. Well, at least in my mind. To start at the end :
At some stage I grabbed the SPL meter to see what I actually had been setting for volume, because I saw I started out for some album at -24dB in XXHighEnd which already looked fairly high to me, while I had been Alt-u-ing so many times that something had to be going on. I had been playing at 94-96dB for hours.
At passing the partner in crime here when walking with the SPL meter towards the speakers I heard her say "isn't that a bit on the harsh side maybe ?". -> No no, but I think it is loud (at that time only recally my Alt-u finger excersizes). "No, that is not loud, I think it is too harsh".

After this she told me to be totally tired because of the music and I - by that time - could explain it as "too much drive". And well, "drive" has never ever been a parameter for me. It exists, but it never occurred to me, and this is the first post I spoke about it in this forum.

---

Pumping. Not being able to catch up with your own breathing.

---

So how surprised was I to see Coen writing about "drive". He may have written about it more often and I just read it without notice, but d*mn, what important can this be. Well, once you are able to translate it to this "function" after hours of thinking what the heck I was actually listening to for changes. So, now back to the beginning of last night :

The very first (work of seconds) thing I notice (a.), is this really crazy emhpasis on a frequency which indiretly I dedicate to cymbals showing their underlaying main vibration. Say the lowest frequency they can express (all sh*t on top of that).
With the hope I am able to repeat all in the same chronological sequence I noticed things :

b. where is my fine bass (next 10 seconds);
c. finally that stupid delicacy is out (!) (1 minute);
d. What the h*ll is happing with all that rattle I hear and never was there by any degree ?;
e. where the heck is that frequency that is so profound (almost throughout the first 2 hours);
f. This is not about frequency ? it is on/off (see d.).
g. How can such a most profound change not cause a flavor in all ? I'm quite sure it does NOT;
h. This can't be right; I must be playing crazy loud by now. Nobody complains either ?
i. Sh*t mon, I am moving body parts which otherwise won't.
j. Why is all swinging so much. C'mon, this is no swinging music !
k. 96dBSPL ?? what the f*ck ?! and now someone says "maybe harsh" ??
l. So where is my bass; put that to the test. Put on a most profound.

It was actually l. when I finally dedicated the whole thing happening as DRIVE. How ?

Well, that super bass was still there. But it hadn't grown with teh SPL of the remainder - or something. It now occurred to me that there was super (can't find a "higher" level word for it at this moment) BALANCE.
Can balance imply drive ? never mind for now.

Seconds before I found "drive" I was sure it was about speed. Sheer racing speed, but at frequencies I again could not find. For sure not in the highest frequencies, which seemed to have completely overwhelmed by what was now happening under there. My bass test - same story actually; it is just overwhelmed. Still there and as good (felt the woofers many times - still feels the same), but all overwhelmed with where the real music plays. And I don't know where that is because it is not about frequencies. It is about on/off.

It seems to be the reverse effect of better resolution which I now tend to call "separation". Mind you, at the technical level. So, any On/Off shows as on/of while before it was a grey On. Not filled by injected higher frequencies ? (this one I make up right on the spot now).
How Leonard Cohen starts to more sound like him instead of Freddy Mercury. That kind of thing, but now in everything.

Yesterday I had been dying to try so many other albums, rock as a first, but I couldn't let go of whatever was playing and played because of again other "tests" first.
It is the biggest change I ever heard (which made me sort of post about it after being into a fifth track only, which is NOT like me at all).

But is this (topic) about SQ, or is it actually about something else ? Let's put that in a next post.

So, finally such a shouting post from my hand, on a subject normally a few others shout as loud as they can. I tried to stay away from "all for the better !!", but it is obvious that I can't. And maybe it is not. At this moment I don't care much. But I do care about is :

How the f*ck can sound change so much (and in my view at this moment even for the better) just because of touching a few USB interfaces. Hey, I like to be recognized as the one now with the largest change ever. But *did* I achieve a largest change ever, or has Nick's or Paul's change been in the same leage ? It can't. It can't because I see no description which even closely resembles mine. In fact they go the complete different direction. And I STILL don't agree with what's described; did you read my "stupid delicacy? That.

Sure, we like a delicate sound opposed to "digital". We like it to be palpable. We are pussies.
So PeterSt says we must like roughness or something ?

No. But what PeterSt likes to achieve is that somehow - without listening to eachother's systems - we can find the direction to head for. And when someone is glad that he now "can play" 6dB more soft while the other is the most glad that he plays at 10dB more loud without notice, something HAS to be quite wrong were it about consensus.
Agreed ?

But this is how I found it important to let the few of you express about your changes compared to standard MoBo connections and how I possibly could make something of *that*. But all so much depends on how we express about it. Like my stupid technical lists may tell nothing to you and "better resolution" tells nothing to me. That is, until a few combined expressions start to match.

Can I use Coen's ?

And shall we now finally tell what actually changed in my system to let it sound like that ? then at least you understand why I asked you the question in the first place :

Because of my other little project (one-clock) I via-via-via found that this other (untweaked) USB3 board did not want to work without 5V connected. Stupid little thing.
So for startes, that's the change. Or at least it seems a major one; I have one card perfectly working without the 5V where I noticed no difference (and didn't pay attention) and I have this other needing the 5V and I dropped on all floors from all chairs (and didn't like to pay attention BUT tried it explicitly because ...);

By cheer accident - and because of this other one-clock little project - I found that ONLY after the install of the formal NEC drivers because it didn't want to work - ... that right after the driver install there was a "power surge" balloon in the right corner. This happened with and without NOS1 connected, and I couln't understand. Still not BTW. Without the attempt to install this driver I would never have seen it, but while the PC recognized it happily without complaints, no NOS1 wanted to connect to it. And no way I thought that this card suddenly would need the 5V because I think I saw nobody write about any such cards needing it. And my own other of course does not. Only Mr. Pang's and that is dedicated to drive the oscillator. Maybe.

So this is how I tried to pose the question - actually to find out about similar changes to the sound, hoping to proove that all is about this 5V and nothing about any Paul Pang.

At least - and in this stage - we must seriously think about (and of course I am repeating myself for the xth time) WHAT actually implies the differences. A better clock in general, a more stable one etc. which needs so much reasoning, many more options, or the sheer randomness of injected noise, or removed noise if you want.

So, what I get from Paul's (and in the end Nick's) description is at least similat to what I perceive after inserting such a 5V fed card. So allow me, I for now just dedicate it to the 5V thing (don't know what it will lead to). Thus, by replacing *that* with something else like Dexa, the roughness disappears and all becomes a perceived silky etc. Too bad that this silky can be too much of it in my view and that more bass never has been good in my view, but what it is about is the similar description which I can easily call roughness (note : all for the better because *here* it seems to express as better separation on whatever micro level).

So believe it or not, I agree with Paul's description; happens the same here, though not with a Pang card. 5V yes.

This is how we need to look at Coen's description(s) because they far more resemble what I am saying. This is not all that easy to see, but here it is :


Aha ! So Coen too manged to get hold of a card which needs 5V. Well, that is what I like to take from this quote (Coen ?).
Might it help, my card is a Silverstone - can look at the Chip type when needed.

Further down the line I see Coen talk about Drive which was never in my mind, but which I made up yesterday and I never looked in the forum again. Coincidence ?


Actually this is to be it for now. I know, my way long story may look like not much being to the point, but only when we start to recognize eachother's descriptions and hunches and environment (changes !) we might get to something final. For me Coen achieves the similar sound and liking of the various aspects as I do. This tells nothing, but of course I regard it important when it resembles my thang. Paul's and Nick's go the exact other direction. Can be as good, but won't sound like I like it. Important ? h*ll yes, because we like to achieve reality. And like I was happy two days ago, and seem to be more happy todat with a 180 degree different sound, what IS the sound to be ?
And we try this through written words ? Quite a task.


May it help, this is now my sound by more normal wording :

Any top layer - no matter silky and refined, totally gone. But I don't know and knew what to do with it anyway, because it could only disturb because too profound (masks reality of cymbals).

Power power power. Yes, drive. But also into the rythm (ok, must try Grace Jones now). Brought forward through detail in all areas - hard to define. I can imagine to find new washboards in upcoming albums to play, while no washboard has been in there previously. That kind of thing.

What puzzles and intrigues is why (read how) it is possible that the detail/separation (estimated at maybe 50ms of On/Off separation) and which is the most sharply boundaried, does not show any single harshness at all. This, as the lead in to rattling cymbals where no rattling cymbal has gone before (blame Spock). This, along with the higher pitched more bell like coming from it, generally dedicated as "more music". So that area (think 1000 - 2000Hz) has become super profound, is super dangerous in the mean time, but doesn't disturb at all.

Yes, no bass either. But still still there. It did not bother me by even one second; only the puzzling about how they could have gond did.
This is a dangerous one for anyone judging "wrong ! because no bass !". I can very well be wrong, but focus on the better coherence and how the overwhelming of the real music could be for the better after all;
I speak often about "so normal". In this case it suddenly was "so normal" that this profound bass was not there (but mind my speakers). That bass has been so mighty interesting, plays main melodies etc., but I only now wonder how normal that is.

With the latter in mind : all is the opposite of "thin". Strangely though, this emerges by the removing of the silkyness as I think a first, to next being filled out again by the roughness which is round (and no single way I can perceive it as distortion).

The whole top layer (but think above 1000Hz) has become a live concert instead of pussy audio homo desires. Yes, strange wording, maybe not meant to say what it looks like, but the opposite of live music that would be. Also (but I am trying to dig that out afterwards) : smashing cymbals like never before, without the nasty top end of them being able to hurt.


Ok, too long again. But thinking about electrical merit (just hunches) :

Earlier, noise implies false HF to be there and it expresses (or can do that) as a HF top layer which though silky does not match the underlaying implied (!) foundation. In the mean time this same noise is actually everywhere, and fills "separation gaps" unauthorized.

Thinking about more high (volume) peaks at this now not anynore flattened "mid" (??) frequencies, this is THUS about flatting not happening anymore just the same. So, not only gaps re-emerging, but also flattening less happening. I *think* that both should lead to LESS noise (not more, which as you know is so often my idea about changes).

How ?
Maybe others can get wild here. The NOS1 does not require "power" as such. Still, maybe that 5V supply provies more of it. Oh hey, but what about the SNR in there ? Say more signal same noise ? So, better signal ? If so, where is that utilized ? Fewer USB retransmissions (with following goodies) ?

Guys ... IF it is correct that this is for the better ! So I am posing that, but I know I can't do that in a few hours. But what I try to do is find the WHY of the changes and it would be so much comfortable when we'd knew for a fact that the changes are for the better to begin with.

Blahblahblahblah. But trying.
Peter (sorry for undoubtedly countless typos)

Peter hi,

Great to see such excitement and focus on improving USB clocks.

I am really keen to respond to some of the comments because at various points over the last few months and with different clock configurations I think I recognise so much of what your describing. What has become apparent is that the details of the clock implementation totally drive the magnitude of the changes in sound and just as importantly the residual "problems" that remain. Because the magnitude of "improvements" any remaining problem really stands with upgraded clock in place (a nice problem to have  :-) )

Anyway this is a long way of saying I would really like to offer comments but its not really possible without understanding the config you have at the moment :- number of clocks, clock types, PSUs, clock coupling (transformer / capacitive / direct), clock speed. All make such big differences to sound.

Can you let us know what config is generating the thoughts above ?


I think we very likely to be listening to similar changes. Remember this from recent PMs.


"To be honest, I have really underplayed the effect of the clock upgrades on line, off line i'v never heard such a change in sound. The NOS is simply absolutely stella set up in this way. I think you might even want the guys at 6 moons to re-review, I just don't know what they would make of it though, or what comes after a statement level recommendation for that matter."

I still feel a common reference would be extreamly useful and now we are back to the North Sea again haha.

Anyway, fantastic to hear about so much progress.

Regards,

Nick.

Yes Peter, please tell us actually what changes you have made...all that I can take from those two splendid posts are that you are using a standard Silverstone card...but this is surely not true is it?

Anthony

PS:  I am quite excited by your excitement

Peter - Wow not sure I follow all of that will have to re read a few times. For sure I do not recognise all you mention in terms of SQ but just a couple of quick notes about the sound quality I get with Dexa's

1  With Dexa's I do NOT get more bass as such. I DO get more speed and "snap" even at low volumes it is noticeable. In some situations with Dexa's it even sounds as though there is less bass but it is because (I think) it is cleaner and more controlled.

2 With stock NOS1 if I stand up and walk around the room I can hear bloated (arrghh horrible) bass in certain areas (my thin odd shaped room is probably more like an organ pipe ha ha) but with Dexa's if I stand up and walk around sure the bass emphasis is still there but it sounds soooo much better quality.

3  I definitely do NOT listen at lower volume with the Dexa's I do not perceive any difference in volume. In this respect I do NOT concur with Nick. There was a hardness to the sound to begin with but that disappeared with a week running in of the Dexa's.

4  Compared with Dexa's the standard NOS1 has a slight digital "glare" or "haze" - the Paul Pang card in place of the pcie sounds warm analogue but does not seem to do achieve anything like the Dexa'd pcie card achieves which simply seems to remove nasties.

What I hear now is that the Dexa's easily set the bench mark qualitatively - absolutely no question to my ears (even just considering removal of "nasties"). Then there is Paul Pang and stock NOS1 - now have heard the bench mark the difference between Pang and stock pcie card is more a question of flavour.

There is a big caveat though - my NOS1 has some other mods to do with distributing capacitance on the USB board and changes in types of capacitors elsewhere (I know very naughty !!! - and I know it needs its noise profile re-calibrating Peter) - I will soon be installing Dexa's in the stock NOS1 so watch this space.

Cheers

Paul

This is exciting stuff, but not at our control at the moment. Like said before the mechanism that causes the very big audible changes is not known. For shure it has to do with powersupplies, clocks and data transport.

Lets not forget about the latter. I remember posts that some usb cable made the biggest difference yet (this was before the Nick/Paul clock experiments). I wonder if USB cables still make a big difference after the dexa clocks have been installed....

Anyway it must have to do with the same underlying causes, like noise profiles, signal degradation and (a)synchronicity. We must have an extraordinary sensitivity to the resulting effects because I think getting these gremlins in focus would need extreme measurements.

Hopefully our future experiments will reveal more about the direction to look for. For now we know that we have an awfull lot of parameters to tune and if we believe Paul and Nick (and I do) an expermimentally derived working solution is available. Now I am going to puzzle to get the one clock thing going.

Regards, Coen

I know this is not in line with what you wrote, but are you (Peter) saying that the 5V coming from the molex is not needed at all ? Will the PP card, it does not work, nor does it with the Sotm card...

Alain

Quick response for now about my configuration :

(I didn't tell much because there is nothing)

Just that Silverstone USB3 card in the normal audio PC ("XXHighEnd PC") with the difference most probably coming from the 5V which this card NEEDS (or otherwise no NOS1 is seen).

Of course the card itself is not exactly the same as the other one (but both are the normal NEC based), but the only difference is that 5V connection which the Silverstone requires.

Otherwise all is normal. No secrect connection in order at all (yet).

Regards and thanks once again for your great responses.
Peter

Peter thanks,

I will read your posts again with this in mind.

My initial toughts are that three ingreedients could be on the Silverstone card by happy coincidence.

- Improved PSU to card and clock via the molex.
- A better quality XTal / clock circuit performance
- The USB Clock speed happens to be at the sweet point frequency relative to the NOS clock speed (this seems to be slightly faster and makes quite a difference)

A while ago, I tried the same USB card type that Pauls has had excelant results from in the past when I have been to listen to his system in my system and overall it did not work so well for me.

I think the clock journey is showing that "matching" of the USB cards to the rest of the system could help. So far this has been a happy coincidence if it happens but now clocks are being looked at I think it is becoming possible understand how to tune the USB card to the NOS.

Best,

Nick.

Edit,

Actually I should also have mentioned the two versions of the NEC chip that Coen pointed out in an earler post. Could one sound better that the other ?

Paul,

Yes. But this is how sometimes it is difficult to grasp real merit when two people work on the same solution, but one of them is just that small tad differerent (which often is not revealed right away because regarded not important (perhaps)). Anyway, as you recall I invited you and in slipstream Nick with that, but in this sequence because I heard something from your setup which I thought was correct (thinking about the better). So (or also) without many words spent so far :


you got that; I can't imagine this the other way around than you said, although that too is and remains dangerous for a definite standpoint (but it is mine for quite a long time).
Please notice the (perceived) difficulty when you two guys always concur, which is and remains so until in very bold it is written you do not (on subjects). You really should (put in bold when needed), because it is important. Well, not crucial, but when we put ourselves to the ridiculous task to find out "physics reality" by means of a bunch of words, I like to think it is.

What about making quite another "message" about this :
We should all praise ourselves not to be subjective in the first place and be the most glad we can make anything out of eachother's message, announcement or shouting to begin with and that we are all HONEST. I mean, I don't see this working even the slightest on any other audio forum I know. And this is only for a very small part about me you know ...

Peter

I like to point out one more thing, because it is crucial when we talk about noise as a "SQ control" and from my previous posts from today it is so easy to get the wrong message from that :

I know for a fact that it is very very easy to play even at 96dB without notice, when highs are blanketed and which actually feels your ears are stuffed. So, we "tune" our volume in a sensible area (make that 2Khz for now) and the lower that area expresses the higher we "can" set the volume. We do this until that area is at level (to some extend you can call it the threshold of pain when things go really loud), but in the mean time the bass goes up and up and creates the SPL.

In this special case (whatever the case really is) I must have done the same, BUT I know how this usually is related to some more top layer as talked about. For me this is about those actually refined nice highs (usually the highest frequency audible when not about super transients and with the notice that cymbals play maybe up to 3KHz for their most audible frequencies) because at some point that too (thus even when refined) becomes "too much". That this is related to speaker (or in-tweeter) output is something else so already that makes it not the same for eveybody (assumed we do have the same ears). So :
In this case this could not be about that highs output because it even was overwhelmed with actually the most snapping tingles and stuff at the lower frequency level, with the same feel (super resolution) and most certainly the opposite from blanketed.

This combination was a totally new experience for me. Thus, the most snappy highs became much louder (generally the mid-high horn showed a way more high SPL than the woofers which is not intended to be so (should be around equal) BUT it does not hurt (ears)).

So message : Before one thinks that high SPL is created through bass output only and which is the most easy : here it is the other way around.

Peter

May it help anywhere in future : The main chip on the one requiering the 5V is D720201. The one which does not has the D720202.

Tried to find the difference of the both, but too difficult to do that in even 30 minutes (knowing Chinese language could be helpful but it still would be problematic).

I have Dexa clocked 2 pcie cards with NEC D720201 (only because I had them lying around) and D720202 chips respectively and very different board implementations - I cannot hear any differences - certainly nothing significant compared with the other things that are going on in my system at the moment - not sure if that means anything at all (I have not checked data sheets at all) but I share it for what its worth!!

Oh and both cards were powered from the backplane - ie the molex was not plugged in.

Paul

Thank you Paul. So both chips can work without the Molex supply.
Peter

Hey Peter - First of all this is a totally brilliant Forum we are all honest and objective (as far as mere humans can be) and there is none of the stupidity of other Fora. For three people (and more!!) in three different locations with different systems and room acoustics (and different ears) to come up with stuff that is even close to objective does not happen anywhere else on the WonderWeb.

Nick and I (the "choir boys") are holding back on that visit to the "Altar" only because you put us off (as I perceived it) pending outcome of your "ubiquitous" clock experiment. No problems let me/us know - I wait here like a "coiled spring" ready to come over (even knowing that it will cost me that Scotch ha ha).

Meanwhile - my wife has insisted that I demonstrate the difference between Dexa and none Dexa - now that really will be interesting!! also she has shown particular interest in my "heavenly trimmer". MMmmm.

Cheers

Paul

Thanks for listing the chips Peter. I had not been aware that there are two versions running at different voltages until Coen mentioned it. Iv been trying to get my hands on the NEC USB3 chipset datasheet for a long time, they don't make it easy to get a copy. If anyone has it would be really great to have a copy.

Cheers,

Nick.

I should perhaps have updated on this point. With the Dexas fitted I did get that the same perceived loudness was produced with greater XX Vol attenuation settings. I know this in the past has generally been a sign that all is not well. But in this case is sounds like a richer fuller presentation whilst maintaining detail.

By way of update, it seems that the effect is to do with clock speed at the ends of the USB link. The extra loud perceived sound can be tuned in and out of the presentation by trimming clock speed.

I guess Paul might not have had his clocks trimmed in that way.

I'm waiting for my frequency counter to arrive so that I can start relating relative and absolute clock speed to perceived sound. What most interests me is understanding how to tune for the sweet spot reliably.

Cheers Nick.

OK so my better half has just been listening to Dexa versus not Dexa AND she insisted from the start that I must not tell her which NOS1 she was listening to (Dexa'd or standard). Sensible girl so "single blind" I tried to keep a poker face to not give any hints as to which was which.

So we started with "NOS A" and played the first track (Elvis Costello Wise Up Ghost - hard processed recording) and then I changed to "NOS B" and played the same again. OK mmm where is that "embarrassed" emoticon :blush1: she said "have you turned the volume up?" to which I replied "no". She said "well it sounds louder, fuller, more out in the room, clearer, more detail, I can hear the emotion in the voice better I prefer this one" - the first one [NOS A] sounded flatter more recessed less clear not so engaging".

We tried the same comparison on a number of different tracks different types of music and every time she easily spotted which NOS was which and every time she preferred the Dexa'd version with similar comments to above.

But then we listened to Eldar - good quality jazz recording on NOS A - with this much better quality recording immediately she said "this is your NOS the blue one" so I said "well you ought to hear the other NOS before you make up your mind". So I changed to NOS B and she said straight away "no I got it wrong this is your NOS" - well maybe not very scientific that last "blind" test but anyway.

That is pretty much verbatim - anyway she 100% knew whether she was listening to Dexa or none Dexa - and preferred Dexa.

But that does not mean Dexa is more accurate of course but that is what happened.

By the way maybe when Ros said "louder" maybe I would say "dynamic" well anyway now you know maybe I was wrong!!

Oh by the way for quick comparison / switch over I left the PC end connected the DEXA'd pcie card - which does sound better than when connected to a none Dexa'd pcie card. So a bigger difference could have been heard with standard USB3 none Dexa'd card.

Cheers

Paul



 

Paul,

I think Ros deserves a medal for being so patient and I have to say I think she has "first class hearing" as she spotted the extra "good" loudness of the dexa setup   :)

Really though, very interesting and I hope you are both enjoying the sound.

Kind regards,

Nick. 

Thanks for sharing Paul and very interesting stuff.  I am still kind of glad that my wife regards this audio stuff as a folly.

One thing to possibly do is to measure the SPL between the two dacs which would get out of the way if one was in fact louder than the other.  I am with you that more dynamic can fool us into thinking it is louder, so it would be nice to know what you measure.

Cheers,

Anthony

Also remember that louder music often seems better...level matching is very important when doing critical listening, so if one is actually louder it may explain at least some of Ros' impressions.

Hey,

The below is worth nothing, because it is just about that "random" setup of mine and though I express about it heavily I feel it is off topic. Anyway, this is what was going on here yesterday :

With our both reference of the day before and the eventual expression by Ciska "isn't that a tad on the hasrh side" with my response "no, it is way loud", yesterday it took seconds only to make her say "too loud !". But it wasn't at all (80 maybe) and I didn't change (this is dangerous as we know). A next album was harsh according to her, but I did not hear that. But changed the album (hey, these days we can use Madonna only as test object).
Again I was thrown at with a "tirering" (like the day before).

It puzzled me; I could see where the remarks came from, but to me to doesn't work out like that.
I put on one of her favorites (a Jimmy Smith) and thus well known, and AGAIN it was harsh.
Btw, we use the term "sharp" when literally translated, but it can't be literally translated; my mentioned harsh would be in between English harsh and sharp.

So what in the world could be harsh while I only hear detail in a new way. Is it this "new way" ?
Then in the first track of Jimmy I heard tambourines. Ah, shoot, I wanted to tell you about that yesterday as THE example of on/off separation. Ok, in between the lines :

What comes so much forward is how the individual bells of tambourines are so auudible. And, I think we all know about this in the first place; can be a grey-ish sounding instruments which shows a flat one-graduate hit on the hand, or when the system improves the color comes more in and you start to hear separate bells (okay, tiny little cymbals).
So, this occurred to me the first nightm and that from synths (yea, sorry).

Back to Jimmy. So, I noticed tambourines I never ever even heard. Don't ask me what I heard instead (the sound itself can't be missed), but the guys use tambourines suddenly. Ehm, all over actually.
And so I pointed that out like "hey, I don't know what's up with you or possibly me, but did you notice these tambourines now ?" ... and this with a tone of "you better think this is better because it has to be". LOL

But crucial side note :
It never happened before that we had so much disagreement. Actually we never have.

One of the things coming from this was her "so boy, then let me hear S'es". Yup, this is always crucial because when S'es don't work out (like not at all or hurting) something is wrong in the base for dead sure. Anyway here too, it went with a tone like "for sure they don't work, because otherwise I wouldn't have problems somewhere".

But they worked and they worked perfectly.

I noticed another thing yesterday;
One of the measures is always applause. This so easily fails. However, we (or at least I) tend to judge this two ways only : Or it is to sharp and therefore something is wrong (hey, now I use the literal sharp !) or it is not and no matter it could be blanketed no real more measure points exist; It is too difficult (perceivedbly because no recording engineer tried to capture the audience clapping for the best anyway BUT when too sharp it should be wrong anyway).
Yesterday for the very first time I heard "sharp" clapping as my own hand clapping would be and it was NOT too sharp. So, in all the by now 1000s of test hours for the first time I found a next (third) gradation in how applause can sound.
Should tell something.

All I can tell is that the perceived harshness/sharpness is at a totally different "frequency" level. I *was* able to point out how much natural a hit on a snare drum suddenly sounds and how it does exactly NOT comes along with a too transient hit (think again about the one transient now being "unfilled" with small gaps of on/off). I was able to point out that S'es even perceive a tiny "vibration" (not the zzzoom type) which CREATES the S, instead of hearing an S which is just there and can easily go wrong.
Hmm, I see myself writing this CREATES and may have found a key there.

Ok, as said in the beginning, never mind much. But here too we have our little fights sometimes about who is right and who must be wrong, but this one is strange and not sorted out yet. The sound is totally unsusual, that is right ...

Peter

Hey Anthony - I just ordered an SPL meter well I needed one anyway!

Paul

LOL

Yeah, and what do we use that for ??

"Hey, see, this IS not loud. Don't nag now !"

:swoon:

You could use your phone or iPad as an interim measure Paul.

Peter, what are your next steps?  Are you tempted by a clock change?

Anthony

Anthony, no. Not at all.
Or yes when it is actually about the "trimming" hence my situation is purely coincidental and I have it trimmed right.
Notice the possibility for "trimming" needs other oscillators anyway.

But honestly someone should get the "one clock" to run first; this will be a so much different situation that it will be too important to just lay aside. I wanted to do this through special means, but I can't get the basics of that running (while it just should so that makes me wild). Also : I really need to do other things at this moment - like finishing some XXHighEnd version ...

Peter

With thanks to Nik.d AKA George, see attached.

Peter

PS: How this is related to "NEC" as such was beyond me right from the start. But not important I think.

PPS: Two versions running at different voltages ?

George,

Thanks for responding to the ask for help with these data sheets further up in this post and for your pm.  I'm been trying to get my hands on a copy for ages !

Also Peter thanks for linking them.

Now for some reading  :)

Cheers,

Nick.

This is great ! I'm just scanning though and already some some ideas that are simple modes and could help performance are coming to mind.

This made me laugh  :)

"Note 1:
An external modular oscillator cannot be used instead of a crystal, due to aggressive clock management in reduced power states."

Guess we will have to avoid reduced power states then haha.

Nick.

Hey Nick, nice. Btw, I didn't look into the PDF's myself yet.

Well, at least that tells that any "one clock" situation should be on the card's side.

Keep up the good work.
Peter

Hi,

This is a post that i’m really pleased to be making, there is a very nice gain in sound quality but I'v had a feeling that there should be a more consistent approach available to tuning the PC and NOS than I could find before.

What is here is the coming together of work on clocks, messings with BIOS settings and some help from the NEC USB chip datasheet the George posted a few days ago.

Lets start off with some sound quality characteristics that come about when tuning USB link clocks using the dexa "heavenly trimmer" haha (first touched on here  http://www.phasure.com/index.php?topic=2784.msg29143#msg29143)




So as the dexa usb clock speed at the PCIe USB card is tuned there are three sound quality conditions that occur:

1)   Sound characteristic 1 is super smooth, it sounds right and very appealing but when listened to you realise that dynamics / transients / weight / scale / and presence have suffered BUT the hardness in 2) has gone which is good.

2)   Sound characteristic 2 has a harsh element in the upper frequencies too much sibilance and on very complex passages particularly with lot of high frequency energy it feels loud in a poor sense.

3)   The “sweet spot”, sound characteristic 3 has smoothness and detail with good portrayal of dynamics / transients / weight / scale / and presence in addition there is detail but without the sibilance and harshness of case 2.
As you tune the dexa clock trimmer the tipping point to move from sound 1) to 2) via 3) (the sweet spot) is very rapid and takes a little work to find where sound 3 is to be found.




So far so good, some folks might recognise the sound types described. I have posted before on the CPU BIOS clock ratio setting and memory speed needing to be carefully set, on thinking these experiences and the dexa trimming about I was starting to see similarities in the sound types produced.

Then whist reading the NEC Data sheet that kindly George provided it was clear that data is being clocked in to the NEC chip from the PCIe bus which nominally runs at 100Mhz, hummmmm.....

So thoughts moved on, what if the main music data transition points can be tuned going backwards up the data stream from the NOS1 all the way back to the HDD, AND what if they behave in a similar way ? Now THAT would be something to work with ! -  a common principle that might be applicable to much of the PC and the USB link into the NOS.





So possible tuning points working back from the NOS upstream to the PC:

a)   PCIe USB clocl speed => NOS USB usb clock speed (using dexa clock trimmer on the PCIe USB card)

b)   PCIe bus speed => PCIe USB card clock speed (set using BIOS BCLK setting)

c)   CPU speed => PCIe bus speed (set using BIOS CPU ratio, (need an Extream mode lntel CPU for this) )

d)   RAM => CPU Speed (using BIOS Ram Speed setting)

So when tried the interesting thing is that each of the above four tuning points have exactly the same effect on sound. When the upstream side of the data transition is fast I get sound chrateristic no 1 when it is slow I get sound characteristic no 2 and then there is a knife edge transition setting which produces the sweet spot 3 in each case !

As an example a change in BIOS BCLK setting of less than 1% hops over the sweet spot setting from sound no 1 to sound no 2. In reality BCLK of 100mhz in BIOS = sound characteristic no 2 and Bios setting of 99.5mhz = sound characteristic no 3, the sweet spot, BCLK of 99mhz gives sound no2.

So I have spent  the afternoon messing about (being suppressed at the similarity of the effect of tuning each parameter) working from tuning points a) to d) above. At the end of this process sound is really excellent, better than I have heard before, silky smooth with ultra detail, coherent and with presence and scale tone and authority.

By a happy coincidence, last night I went to a carol concert at our town's cathedral to hear my daughter sing in a choir. It was a really nice evening but it also meant that I was listening to live church organ last night. Today with the tuning above the sound of reproduced church organ is very very close indeed to what I was hearing last night.





Settings on an ASRock z79 Extreme 4m mobo ended up as follows.

CPU BCLK = 99.5mhz
CPU ratio = 29
RAM 2100 Mhz
Dexa clock at the PCIe USB card 1/16th turn (about) clockwise from standard position (note running against a dexa set to standard trim position at the NOS USB interface.

Final thoughts.

I am strongly suspecting that the above may have something to do with any “wait state” that data encounters as it goes from one PC sub system to the next introducing jitter somehow. I'm also very excited about this, its the first time for me that a consistent approach applied to multiple points where speed can be tuned has resulted in totally consistent effects on sound quality.

Now I’m considering accelerating a project I have started to put a tuneable clock on a HDD, I know what I think it may do when tuned - but will it !!  :)

Best,

Nick.

Ha ! Yes. I read that elsewhere already and wondered.

Now I will try to get my head around all of this. Not so easy.
Great thanks Nick !
Peter

Something else : I find it quite confusing that this is now in this "noise" topic while the "trimming" tells it should be in that other one about *that*. So I think I will move this ...

[Edit : Which I just did, see the  "A consistent approach to PC / DAC tuning ?" posts.]

Regards,
Peter

Peter hi,

For me trimming is not really the point of the post.

The underlying points are:

 Highlighting that jitter may be produced as data is sent across pc subsystem interfaces.

That the effect of conditions at these interfaces have consistent effects on sound and,

That the understanding of these sources of noise might help develop ideas on the underlying reasons behind the jitter. Could this be wait state related ?




The effect of tuning on sound quality is not so much of interest as it is what the observations might highlight about digital noise. Im sort of excited as the pc hardware has always appeared like a black box and here may be something that links cause and effect with  consistency, something that shows logic at  last.

The effects need to be further tested but the interesting conversation for me is what is happening to impact noise.

Cheers,
Nick

Nick,

When you talked about jitter in your previous post, some neck hairs got upright here. So the answer to that is No, I don't think that this can be involved.
But I had another kind of response in mind which does not suite this topic anyway. ;) The other one yes.

Peter

Peter,

What are the options if not jitter induced changes in sound ?

So far back into the pc im not thinking this could be about analogue domain noise. Also a consistent effect from a similar situation in 4 different locations in the data chain and the sound characteristics sound like jitter spectrum.

The only other thing that springs to mind is that could have an effect like this would be data loss. But could this ? Seems unlikley.

So why the hairs rising at the mention of jitter ?

Nick

Edit actually when I say jitter im lumping in the idea of "lateness" of data moving through the sub systems. This could be different to jitter in a continuous stream I guss, a time related effect though. Actually maybe this is more likely.

Then there could be something that happens at an interface that effects other cpu processing. I guess this could be other stuff rather than jitter.

Anyway by all means move the post but im really not seeing how this fits with "usb clock experiences" if that is the post you had in mind. That post considers the effect of a single link in the chain whereas the discussion here should try to address a fat broader scope. In time I hope that with more experiment here the transfer from hdds can also be included or excluded in the scope of this discussion.


 

Digital jitter within the computer is a well known and understood phenomenon that is overcome with buffers and asynchronous protocols, but I am sure both you guys understand it.  I read an article a while back regarding design of sata cables to minimise jitter which was very interesting but long story short I don't think this jitter can directly impact the NOS1 which is the thought that may have raised Peters hackles.  Indirectly though, maybe there is some influence of the extra noise generated by the asynchronous digital resends with the computer that we are able to hear on a dac such as the nos1.  Maybe.

Anthony

Very interesting post Nick.  I will digest it I'm the coming days.  In the meantime you could probably consider playing with a low noise CF-sata adapter and a clock upgrade for it for the os drive in your XXHE pc rather than a HDD.  Cheaper and most likely lower noise than a HDD or SSD.

Cheers,

Anthony

Nick, consider me a little disappointed here. I thought at least I was on some track regarding how jitter can or will be noise induced and which is not the way you propose it. Am disappointed because it seems (or comes across as) you never read it. And yes, this is all in that other one topic (if my memory serves me well).
Odd.

Peter

PS: Maybe dive into asynchronous USB and how it is theoretically impossible to have that carry jitter. If you know about this, please support your claims by the reasoning how it in your view happens (no penalties when you are not correct, today or in infinite future). But just claiming it ? sorry.
What I don't get is that I spent maybe 10 posts about it and it seems you have missed them all ?

PPS: Apologies in advance for this not coming across as the most nice. ;) I guess I am too floored to let it go.

Peter,

No problems, I'm trying hard to make sense of this myself. Perhaps you could point me at the points in the other post which will help me understanding your thoughts.

Having thought about this overnight regardless of theory there is something happening at the boundaries that changes perceived sound characteristics  and the effect sounds at least similar to jitter but again thinking overnight about this I think the I should not jump to any conclusions here.

As mentioned above could this be data loss ? Seems unlikely.  Or perhaps a transmission pause at an asynchronous boundary causes the cpu to go and do something else and this effects other processes I dont know, or something else....

What there is to work from is that there are consistent timing related sound changes where asynchronous transfers are taking place as Anthony pointed out. Beyond that I can't really go far with experiment other than to practically  test to see if the principle holds for other transfers like from the hdd.  I wonder if testing software to test the effect might be possible / interesting, i dont know.

I was reflecting last night that i had come across  the cpu ratio and ram speed effects before but its the usb clock work and putting this together with the pcie 100mhz transfer speed that allows the effect to be heard consistently and hints at a general principle at play at asynchronous boundaries.   

If you can point me in the direction of your thoughts I would like to take a look.

Regards,

Nick.

Ok, I merged the two topics. Reason :

The one we are looking at right now is about "trimming" or at least it rapidly received that subject; two topics explicitly about that is too confusing while it also it too important to have it all right and together (my view but alas).

If it is not clear what this should be about at this moment :

With the perception of asynchronous USB (which can be considered normal (and error checked) computer data) not being subject to jitter, it has been long time proven that at the DAC's end "noise induced jitter" is there for a fact (because we know of no other reasons which can influence in-DAC SQ), unless it is about data loss to begin with *before* it travels the USB interface. This is not likely but to be taken into consideration just because it is an option.

That the latter is suggested to become more likely by myself through a number of posts in this topic can be considered "conspiracy thinking", but until proven otherwise - and with the notice that I don't know how to do *that* - it is and remains an option.

In the mean time and now NOT considering data loss, it will be noise travelling the USB interface that implies jitter at the DAC side, and when this is assumed to be the case, it is easily proven that when influencing the noise, the jitter will be influenced along with it.

It is also easy to see that influencing the noise can be done by means of at least two sources generating that noise, the both working into eachother creating a far bigger resonance (oscillation) with slower or faster repeating patterns than any single source could do in the first place (because no resonance possible).

All with the notice that for each and every source thinkable, all comes togetger in that USB interface (possibly any interface for that matter), with the confusing factor of the interface itself contributing to the effect because of its two oscillators at each end of the interface.

I can continue, but will not do this in this "recap" post because it will be speculation. What I said above is merely coming from logic though with the notice it is still mine which is no guarantee it is true.

Open to any kind of response - or let's just continue the subject ...
Peter

Peter,

I think with the merging with this thread the "consistent tuning of the pc" post I made is now out of context  even 

:offtopic:


This thread which deals with about 20% of the scope of the post I'made at best.

My intention with this post is not to further the discussions on usb turning, potentially there is much broader and more profound points to be debated stretching across the PC and dac system.

If the post must be moved (although I fail to see why as it was the ,"hunting for noise" thread that started the in depth reporting on usb clock experimentation after all), it should be placed in a new thread please.

Nick.

So now what I had in mind here;

What goes quite unnoticed for most if not all, is that I have been working on "stability" as such in the PC for a longer time; many of my posts in this topic should be seen through those glasses - but no need to read back on it.

Disclaimer on the idea : It will be impossible to reach total stability. That would mean a CPU response to all what is needed with the 100% exact same time response, like the 24000000Hz of the Dexa's in this topic should remain just that for ages. So, wobbling from, 24000001 to 23999999 many times per second already would not be that while the CPU response would be miles away from such a (implied) stability.
Still we can try to get as close as possible ...

Do notice that this now suddenly is about software (well, for me it always was, and if not explicitly said you can read it in between all of my lines) because when software does not imply steady load, how can the hardware - that implying the noise (changes) do that. So it has to start with software.

For what it is worth (and I myself can hardly believe (in) it :
Day before yesterday I achieved something which was in my mind for a long time : total stability as total can be (I have my means to check for that). Too bad that :

I was not able to stop the music which played flawlessly;
I was was not even able anymore to switch on the Monitor (and with that no clue whether the mouse still moved).
Without the 100% proof : a total lock up of the system, just with the goal of having it as stable as possible. Did I manage ?

Not important. What is important - and now read Nick's findings - is that it doesn't matter much which source we pull, as long as we "heavily" (up to heavenly) tune the noise for our best pattern;
It is not said (remember, all in my view) that it can be done by pulling one source only, but each source will have a kind of heavy influence once it oscillates heavily with the other.
Btw, tuning the memory frequency against the processor's also is a "known working" phenomenon; The PC is full of it.

I have some more to say at this moment, but maybe not too much in one post.
Peter

Nick,

Because to me all feels more like data loss than jitter influence ...

I don't know whether you have the equipment to do this :

Hook up that equipment to anywhere in the DAC hence at the end of the USB interface;
Capture a played track which probably needs to be very short so all can be captured in a one file to examine later (but if your capture can be infinitely large, a larger track is also good of course - even better).

- Start that playback and let the track die out;
- Trim it to the first visible music data at both ends (so no silence data is in there anymore);
- Make two runs of this, one with your sound "1" and one with your sound "3" (or anything, if only well audibly different).
- Compare the two captures for their length.

The thing you could be dealing with for false alarms is retransmissions, so I think the best would be to capture the i2s Data line (just one of the two).

In the mean time you could also try capture USB data itself, in front of the receiver chip, so you could proove retransmissions. Notice that this can be tricky for optimal results because you might tend to just observe raw USB data which comes in packets of fixed format and don't ask me at this time whether such a packed can be half full with real data, a header denoting that (and if so you won't be able to deal with it, unless a real USB logic analyser - and still).
So at first try to avoid this path (I think).

All obviously meaning :
When you can proove that especially first mentioned setup (capture i2s data) leads to the exact same length of data while the sound is totally different, then at least we can put that idea to the garbage can. So you know, this at least *is* something which can be physically checked for; how different patterns of noise influence jitter is quite something else for reliability testing.

No obligations of course, but the thing is damn important. So just look at this, when samples are discarded :

If so, ANYTHING done in the PC will matter; we can go as wild as we like. That the real solution is to get ourselves another means of playback is something else. :yes:

All 'n all this should have the priority because to me it is totally obvious that when data loss is in order all the remainder is completely moot.

Regards,
Peter

Emphasis is mine.

Yo Nick,

I think you didn't get what I tried to put across :

Firstly, the USB tuning is just one of the means to "get there". Well, you said or implied that yourself.
Next, there is one topic about that - with by now (your major post) the kind of proof of it.
Lastly, I do not want things to be scattered already for myself.

Maybe you (or all of you if you want) can deal with what is where and keep track of it, but I can not.
I sure thought of making it a new topic named "General trimming" or whatever, but then there would be a THIRD topic about it, if you only slightly thought it could fit in your "noise" topic (so by the grace of your own thinking it would be a third and I already don't like two).

Remember, *only* because it is not clear at this moment what causes which - but with the direction of "pick one", it is all the same subject, no matter this topic started out with USB-USB.

I can also tell you this - and I hope telling it one time is enough :
I plainly hate it that people start new topics while it actually is about a same or most similar subject as another one; Paul just did that with this topic (not you).
I plainly hate it when in all topics imaginable an ever the same subject arises about noise findings (you are prone to do that).
I regard it bad forum behavior when old posts are edited, no matter in another colour. Maybe the fact that you "can" do this makes us disagree on this little subject.

You may not be able to imagine, but when I see such a thing I say "oh no, not again !".

Hope it is clear now and/but also that it is clear that it is actually you (both) incurring for some moderation now and then. I do this for myself (hey, because *what* you guys spit out is considered important until proven otherwise) and in the end it should benefit yourselves (and not only yourselves).
I am not complaining (would have done so through PM when deemed necessary) - just responding. Defending if you want.

Best regards,
Peter

Anthony, maybe I am a bit mean here (not intended so anyway) by putting this more out of context (leaving out the last part) then this text in itself already is. Read : even with your last part this can not be understood by any general layman.

So, no. Digital jitter in a computer is not at all understood, at least not by me. So go ahead with quoting from somewhere or putting links, which may help at least me.
Maybe the definition of "digital jitter" is about data arriving more late than originally intended, but it is no subject at all because all what happens inside a computer is one big pile of asynchronous (yes you used that in the by me left out part) operation. Call it buffers needed, call it synchronization timers or events - it is UNrelated.

I liked to put the emphasis on this, because you could be the first one proceeding on general ideas about in-PC jitter which does NOT exist. And then I mean : not for any relevance anywhere. Not for general processing of the whatever data and most certainly not for our audio.

But ...

When this can be brought down to such "synchronization" failing, it becomes a complete other matter, because then samples can be skipped or just as easy arrive (sent out !) in the wrong sequence. And mind you, for normal data this would not be a problem because each word in your Word document will find its place undoubtedly (and by guarantee). This is NOT the case for audio though because there's nothing at the other end reorganizing it.
Keep in mind : the audio data we talk about is sent in normal "data fashion" (not as audio) but it becomes time related (which is audio) after all when there's no method of getting it in the right sequence at the other end. This is not jitter, but plainly errorneous. How ? because it wasn't sent as should.

Btw, please don't mind the "wrong sequence" because this makes it unnecessary difficult. Just think skipped samples.

The real jitter (audio-thinking) can only emerge at the clocking out of the audio samples of the D/A chips. So if that clock is not stable it implies jitter (each subsequent sample is put out with slightly different time intervals).
That our precious oscillator is fed by unintened noise per unexplained noise paths, is 100% clear in itself;
That even a hardisks's movent of uneven draw from an SSD implies "uneven" noise is also totally clear (to me).
But see the bold text ...

Peter


PS: At using the Internet for telephone, one of your spoken words can follow many path all in small pieces (maybe even normal telephone lines do that). At my end all will be formed into the right sequence at the grace of the low sample rate giving it the time to do that (not that this really works, listening to Skype means etc.). Now, to put this to the extreme and how "digital jitter" for time UNrelated data can work, look at how a Torrent works. Know it ? All the millions of bits and pieces are SENT as one big random mess, even from dozens of PCs (seeders). But oh wonder, after your an hour of download all is in the proper sequence in the file. Yes, even your audio file won't have one bit wrong. Point is : the random mess was allowed to exist for that one hour long, because the sample rate can be considered "1 sample per hour". Thus, only after that one hour it needed to be correct (of sequence). Call it a buffer of one hour, and see the synchronization point exist at the point something tells the last bit (byte) is in.
Nothing of this can exist in audio, unless all can be done within the 1/786000 of a second for those samples which got mixed up etc.; It is not assumed to be working like this anyway, with one solution only : there must be a guarantee that each of those 768000 samples (which are 49152000 bytes) arrive at the DAC's end, with all needed to read them, put them to the interface, push it to the other end, and preferrably before our oscillator tells to clock them out which happens at 768000 times per second.

Peter,

Wrt the dataloss, is it relevant for the USB interface? I mean that contrary to SPDIF, the USB interface will arrange for a resend if a fault is detected. How on earth can we else transfer programs over the interface that have to be "bit perfect' to work? There is no such thing as a lossy streaming over the USB async mode or is there?

Some psu experiments with the pp board suggest to me that is is all about noise patterns and that we neither want strong noise components nor large semi random spreads. What still puzzles me is the profoundness of this noise on the sound. Anecdotal reports on other forums suggest this is the effect a small amount of jitter entering the DACs can have.

regards, Coen

When I'm on my taking stand anyway ...

You guys may not know it, but the native 384Khz sample rate for USB exists because personal me created that. So it doesn't matter which literal interface exists (the one in the NOS1 is just one of them), but I did that. No matter Windows or Mac (and Linux I don't know about). Important ?

Well, only because I know how much things are at their limits and merely :

I forgot the exact situation, but either it is the too low buffer size (think Q1 x xQ1 x Device Buffer Size) or the KS Special Mode does not report any errors anywhere, while you can the most clearly hear the samples being skipped. So :

The Kernel Streaming core of Windows is a huge pile of code which is beyond the developer (it is even beyond Driver developers) and although with Asnychronous USB this is all error checked (also see the NOS1 Driver Control Panel), not all errors are reported by the OS which means all plays happily along but not with all the audio data. :swoon:
This is not only at the more (driver) hardware level, but also on the side of the audio playback software like XXHighEnd (XXEngine3.exe to be literal). Thus, errors can be reported also in there (my software) but if they just are not, what do do (but play and hear gaps).

The above is about the extreme that so many samples are skipped that you can easily hear them. Think 1000s. However, at just dialling your Q1 higher and higher you will see that at some stage you can't be sure anymore; the last ticks disappeared but with the (my) knowlegde that this will be a couple of dozens; what when it were 3 or 1 ?
Notice : just testing it like this will I think show errors in the Driver Control Panel, so not so easy to comprehend).

I am certainly not claiming this happens actually all the time; what I do say though is that it can exist without the notice of errors being reported.

I can tell you one other nice thing :

Before this audio life I was into a video life; Well, if you dive into how the frame rate of video can plainly never match the sample rate of audio (the both just can not be divided into eachother) ... you know what the official means is to let the video not stutter ?
Just leave out audio samples.
(n.b.: In later stage smart asses like I think Ogy or something created ReClock and that was ahead of things by means of first adjusting the frame rate of video so audio would match that; movie may run a tad too fast or slow, but who notices that).

This is a formal thing, with that only saying that "something" is in there which can do this officially, and which just as well can be used to re-synchronize audio samples which can't keep up with something else. But the other way around. So if only our processor is not in time with serving those samples to be pushed through in time, they can officially be discarded. I mean, the code to do this is in there somewhere and don't you think that the particular Windows OS core code can be (or has been) reviewed by anyone still alive (so to speak) because it is as old as the way to Rome (we say). In the end this is about modular software which comprises of so many small pieces that nobody knows anymore what is performed when in which situation. But who cares, because it just works.

Yup.

Peter

Coen, accidentally I just gave you the answers I think. But notice :

The skipping has to happen before all is formally put out. So, whether on purpose like with the video example, or whether because of a bug - it is crucial that this happens before it goes out to USB because after that error checking will be in order and report.

Think software (you must be used to it like I am) :
Once it is known in advance that things are not 100% right because they just can not be (video example), it would be quite inconvenient to pop up a message that some samples were lost, right ? So first those samples are officially discarded and next what remains is as officially error checked.
Result ? no errors.
But also not quite right.

Peter

Someone must stop me ... :)

Remember - from about day one and Windows 8 I suggested skipped samples to clarify how the strange left/right behaviour could exist.
Notice that I mentioned this in this topic again, because it seems that phenomenon has disppeared and that I exactly know what I did to imply that (or hope it would). So this is Q5=1. However :

Whithout saying it much, I also know that it really would need something beyond me to keep it "stereo" to begin with. So, skipping bytes would break windows, and keeping together left and right samples would need something special. I mean, when at random (lack of time without it being under control) ONE sample is lost, meaning a sample of one of the channels ... that can't be and sure does not happen because it would imply a L/R change until it happens again.

So what is new for myself is thinking about the video example, which the most obviously would very decently skip TWO samples or an even amount (but also think multi channel).

Now, with the notice that there is a formal sructure for audio data that tells how many bytes (or how many samples) will form one logical audio sample (say for all of the channels in order and with the noitice that a sample rate of e.g. 768KHz is actually two samples of that for stereo so the rate is actually double that 768) ...
... it can only be a programmed thing to skip full samples (one logical for all the channels).

Don't ask me how decent core (kernel) drivers are to do this in a self-cointained fashion, but if so it is still part of the OS.
Thus, not a few bytes or anything are lost; no, always full logical samples are gotten rid of once only time is too short for even one byte. That's the idea now.
This means to myself that it is explained how things don't end up in a mess. It has to be a formal thing.

Hey, if in order at all of course !
Peter

Small brainstorm - may lead to nothing :

My PC runs at 750MHz. This is Hyperthreaded, thus 375MHz per core.
Audio playback runs in one core.

Thinking Redbook, 705600(sampling rate) x 2(ch) x 4(bytes) = 5,644,800 bytes need to be transferred per second.

375,000,000 / 5,644,800 = 66 cycles of the processor are available to do it all for each byte.
It is my fault that only 375,000,000 are available for it and thus also that only 66 cpu cycles are available per byte.

Uhm, that does not seem very much to me !
Let's say that I only need one line of C++ code to hand that one byte to the audio buffer (never mind what that is) and that it will not be wildly off to think that one of such a line takes 6 lines of machine code, then only 11 cycles (66/6) are left to process one line of machine code. Uh-oh.
These 11 cycles should include wait cycles for the various transfers from memory. Next, the remainder of the work is about switching buffers, activate them and some more overhead. I already don't see it working ...

On the other hand, when the "load" of the playback is related to the cpu usage (which shows maybe 2% for all the cores in total, but the one crucial core also showing "nothing"), then what to think of that ? That the percentage showed it not correct ? can be. Or that I make so huge mistakes in above math ? can also be but of course I don't see that.

Point is a bit : I know why and how I created Q5=1 and how that hugely avoids unnecessary processor cycles needed by the OS itself, and how exactly that makes the L/R thing and more strange sound go away (btw still working on that).
*If* this avoids missing samples (all of them ?) then I think that my 750MHz general processor frequency isn't all that much too low because normal speed would only be 4 times higher that implying 4 times less sample loss; it would be a total coincidence when that 3GHz suddenly could process all samples, plus that 5 years back it couldn't have existed.

Let's stick to my math being wrong for now; seems more safe. :scratching:

Hi Peter,

I've been trying to figure out how to make a good post about digital jitter and attenuation but I am having trouble getting things done on this iPad (am travelling).  Here is a link (http://www.eetimes.com/document.asp?doc_id=1272147&page_number=3&piddl_msgorder=thrd#msgs) to an article instead.  

My key take home point is that the higher frequencies attenuate more easily and are more prone to jitter, which means that using a faster bus (eg. Sata 3 compared to sata 2) must run a higher frequency and is therefore more prone to jitter (but not the kind of jitter that is not totally eliminated before the dac).  How much extra noise is created by digital jitter through resends and the the like along with the size and frequencies of this noise are the pertinent questions.  Can this noise actually become audible?

Sorry for the lazy reply, but I have been trying to get this reply done for nearly an hour...damn iPad.

Anthony

Peter,

Have you done the same tests, this time with the CPU at its full speed ? And with the hyperthread off ?

Regards,

Alain

Nick,

If you'd examine the Wait States you will see that when set ("configured") the most badly, there's a cheer over 100,000 per second CHANGES of that. Here are your hints :

Look at what each C1, C2, C3 indiviually does. This is important. Incorporate the cache sizes (for each).
Only BIOS changes (weird combinations of settings) may bring mentioned 100,000 of the worst kind back to 10,000 of the best kind, one of hem being totally dead now.

I don't require you or anyone to dive into these matters, already because you can't (or it can't be expected that you can). But do notice (or trust me) that mentioned 100,000 is about copying 256K or 2MB of memory that number of times per second. This is not bad in itself because the processor cycles are just used to cool the processor (that's what the Wait States are about). Hey, CRAZY ! The thing must be quite busy just cooling itself. Got it ?
In the mean time it creates noise (that assumed and that especially assumed to be harmful).

So Nick, while you bring this up as a mere side note subject which could be a larger subject, "we" are now (and we = XXHighEnd) right down at that level manipulating). Let's keep that a secret, okay ?
haha

But ... when something like a PCI bus, or memory state or USB or whatever so much more, was not ready for the next processor cycle, there we go again with wait states and stuff which will be forever beyond me. But we might be able to reverse-engineer it. At least that's what I'm trying to do.

Before we get confused - this now again is about the sheer noise issue and not about missing samples. But I am as far as thinking that it can be related. Look :

What I have running today is the above being worked out to some extreme. So, what I intend with it, works. However ...
While in W7 I can see that "something" totally stalls for which I can't find the reason but audio plays, on W8 that same thing does not stall, but it makes the USB interface being unable to recover from ... (re ?)transmission errors. Aha. Here too, I see no reason for it, but it happens. So say that all I do is driving things over the limit and now USB can not recover. Tell me what happens right under that limit ? I should be implying retransmissions without knowing it. But, just a guess. And you know with what parameter I am playing with and trying all of the others which all do not influence this ?

Well ?
The SFS.
(and oh, although I spent a full day debuging it by now, it will turn out that it is my own bug somewhere; but what puzzles me is that this doesn't happen with W7 while no code is different for the both, and that simply an SFS of 18 and higher will just crash W8 - in my sneaky situation as of now).

And what has been that foremost major key sound influencing parameter forever long as it exists ?
No answer needed.

So I do not see pieces of the puzzle coming together as of yet, but I do see more pieces of the puzzle, while all the pieces are needed to form the puzzle anyway.

To now briefly emphasize an important (or crucial) piece of that puzzle : Your both guy's heavenly trimmer on USB. It just totally prooves that one way or the other (noise which implies jitter or skipping samples) is in order. A 10 fold number of pieces are added for me, when similar can be done with a PCI frequency and such. I myself already had a bunch of pieces because again software itself can do it (didn't we agree ove that, no matter no mutual listening has taken place).

Can we be enthusiastic ?
I think we can.

Peter

Alain, if I only knew which "tests" you are referring to. What must I test ? Nothing goes wrong you know (disgregard my last post of course). All I know is that the lower processor frequency brings me the best sound.

Regards,
Peter

Hey Anthony, thank you for that effort;

It is a bit tough for me to decently reply to it, because it seems that you are on the right track with your perception of necessary recends (because of a higher transfer frequency ? <- I don't think so) while using an in my view unrelated article. All I can do is repeat my post about this, which doesn't seem much useful.

That article discusses jitter in a different fashion, because it compares the integrity of a signal (with itself) through an analyser which can be connected the wrong way. So, envsion the original signal (virtually on that scope) while it is compared with a degraded signal because of cable reflections and all mentioned there; It would show a false result only because of how the analyser is hooked up.
"Jitter" there is just used as delay, which in the end is a form of jitter.

Only when jitter gets that bad that not only the sample is captured too late (like for audio this will matter audibly) but that it is completely missed (crucial for your data send over the internet etc,), you can call jitter important up to devistating (but resends can recover the good signal assumed the errorneous receipt can be trapped); So this "digital jitter" (I'd like to call our audio phenomonen analog jitter) is quite crucial when very long distances are in order and it becomes easy miss complete "samples" because the jitter gets too bad.
Missed sample in THIS case : "Sampler" just reads signal, but signal is low again because too late while when it would have been in time it would have read a high (this is not 100% correct but gives the idea, especially emphasizing that such a thing NEVER will happen with audio <- haha  So, people often think that plain errors slip through and therefore wrong valued samples and such, but for audio it is not about that; only timing (when is this always good value put out up to the femto second). This not to be confused with my little subject of skipped samples because the OS thought it better cook an egg first.)

Peter

I know you are excited about the data loss part of this, but please allow me to dwell for a minute on the other part, induced jitter.

What I have read from a number of people (going all the way back to Hawksford and Dunn, perhaps earlier) suggests jitter can be induced at the point where the DAC chip evaluates the "bits" and converts them to analog in at least two ways, by noise or by micro power supply fluctuations.  Would you consider any such power supply fluctuations to be lumped under the general heading of "noise"?

Hi Jud,

Yes, why not. And with the notice that both your (supply ?) noise and "fluctuations noise" are in the end the same (and notice my small twist of those words).
But with this given as a (well known) fact, it now is about how the various "outside" noise sources can travel that route, what those sources exactly are and how they can be to our benefit (assumed they can not be avoided anyway).

Regards,
Peter

Sorry Peter, I was referring to the potential missing samples in relation with a slower CPU, but at least your answer tells me that you consider this unrelated...

I thought of this because I have read recently that some people do not sense FLAC files to sound much different from WAV since they changed their PC for more powerful ones...

Alain

Not to forget that noise also can get in a dac chip circuit by EMI or RFI or can be correlated to the incoming signals (like on chip switching noise for the data line).

You can address a lot of this noise by design (also on the chipdesign itself), but vhf will remain hard to beat.

This noise either translates to imperfect timing (even if the input is perfect) or to physical noise on the outputs (which can be very HF) disturbing the analog stage.

Anyway it will be both the noise level and spectrum that matter. Both can be manipulated with software and hardware measures like grouding schemes etc. While Peter may focus on the software part lets not forget what allready has been archieved on the hardware level. Imho there is further potential in the two to explore.

Wrt to dropped data: it is allways dangerous to not know what you assume! For one thing results of the past are no guarantee for the future especially with an operting system that is not your own (hint  ;)).
Ultimately it would be interesting to know what data is offered to the driver and what data arrives at the dac chip. It should not be so hard to design (or buy?) a card that is able to capture the dac interface signals and store them as  music data in a file. Then software can be applied to analyse the correlation. If we find perfect correlation we can rule out the data drops and it will be a noise only problem.

Regards, Coen

I just dropped back into the thread whilst commuting on the train. I turn my back for a few hours and it's going to take a week to catch up  ;), really fascinating stuff !

Just one thought is anyone able to test the turning measures I mentioned in the post last night. Do other people get similar hardening smoothing and sweet spot sound from ram speed, bclk and cpu ratio settings. I know not many at the moment can play with usb clock speed but it would be good to know that the observations are repeatable.

Nick.

As mentioned last night Nick I cannot at this time because my PC will not boot unless I reset BIOS and load default BIOS values every time because my PC does not like the external clock running on the pcie card before it boots up or so it seems.

Very frustrating.

And wow there is some really interesting stuff here this seems to be a pandora's box with more in it than I could have imagined.

Paul

Hi Peter,

I have a little more time now.  For our purposes digital jitter is the jitter in any one of a number of locations _within the pc_ and does not have anything _directly_ to do with jitter in the dac.  As summarised in that paper that I linked to, jitter and attenuation in digital signals is influenced by the length and characteristics of the conduit that the signal must pass, the shield properties of the conduit, and the frequency at which the bus operates.  In the example that I gave a sata cable is able to alter the length of time that a computer takes to do a task such as transfer a file or boot.  The better the cable is shielded the less rfi influences jitter, the shorter the cable the less attenuation influences jitter, the more suitable the construction of the wires the less impact on jitter. 

Now the sata cable is just one part of one chain of asynchronous communication in the computer...there are other things in each of those chains that may be able to influence digital jitter such as the clocks that have been Nick and Paul's endeavour.  Nick and Paul's clocks are in the final stage (USB card) of the computer as well as in the dac USB input which is probably the link that would have the most potential for improvement in sq IF there is the room for an audible improvement.  Now, we have digital jitter occurring in a number of places within the computer and this jitter is solved in each of those chains before the data is passed to the next chain, so the jitter itself is never going to be a problem for audio BUT the processing overhead of the rate of resends may possibly be a problem and may become audible through rfi (airborne) and noise on the grounds IF the rate of resends can be substantially improved.  Like I said in the earlier post...maybe...if the digital jitter and attenuation is fine tuned to a level that is significantly lower than the default situation (which may not even be problematic in the first place), then perhaps there is some scope for an improved sq.  Maybe.

Anthony

Ok Anthony, I can't see any "duality" anymore in your last post. :)


...


Hey, I had a longer post here, but I feel out of shape today. So all I left in was above sentence, meaning that when you look at it from that consistent angle (your post), you most probably will be correct that it will be helpful.
But also out of control and one of my scratched lines said that I personally will be able to find a needle in a haystack if only it is guaranteed that the needle is in there or otherwise I won't even start looking.

I'll leave it to this, waiting for a better day. :)
Peter

Do I sense it correctly that nothing will be happening with this ? I mean, silence is not a good sign in this case.

Peter

Hi Peter,

Im in a push to sign agreements before Christmas at work so iv not had chance to read through this weeks posts in detail.

It would be great to compare i2s data to wav and I can proberbly capture something. The issue may be that my data analyser can only reliably capture to 4x upsampling rates. For the usb connection this may be fine as that will still run at 480mhz and errors may still occur. To understand transfer problems in the pc it would proberbly need ability to capture at 16 times. This is mased on your post above where you mention that data could be dropped at points in the os due to resources of the pc not being ready.

Ill think about how to set the test up for x4 upsampling rates.

Regards.
Nick     

Ah, that makes me happy Nick (also see next post which I almost finished by now).

If what I describe in there is audible at 4x just the same, or if your own means of trimming is audible through 4x only, 4x should be enough, right ?

Super ...
Peter

Nick, there is another crucial thing which I suddenly think of :

You will now about my so many times being the most doubtful how in the world these vast changes in SQ (it really goes mad by now) can be bit perfect, right ? (and that digital loopback to me would not prooce that because it takes (or at least could take) another more direct path);

Well, assumed that at some stage you will be able to trim off the leads-in and leads-out of the capture, save that back into a file ... why not compare the both once you find them of equal length. They should be totally equal, not only for length ...

Which btw also makes me think about the USB data, which will easily be able to be different of length because you won't be in control really of which USB data goes in which packets and which relates to the timings of both DAC clock and OS. So I shouldn't even attempt that I think. i2s yes.

Regards,
Peter

In the mean time, day before yesterday I was as far as being sure to have the "fastest system" ever. How to explain it ...

In Dutch I would have said " so enourmously 'rete snel' " and it will point out exactly what I mean to say. In English I don't know how to put that. But something like "beyond warp speed". But what would that be, hence how does it express ?

For starters, I tried to apply what was merely Nick's finding about "trimming" with Paul's trimming of the USB clocks as the base (for this thinking). Notice that in my case this is all about software only, BUT I used the Silverstone for the "SQ base" because for me (here) this is crucial.
The actual trimming is in there, but I can use it to the extreme only. A bit difficult to explain, but think like a theoretical best setting which indeed is theory only, and with that setting I may receive the best SQ. May, because in between that and a next too rough step should be more settings, which need further development (enabling those steps to be smaller) with the knowledgde that this is theoretically not possible. But maybe I can find something after all.

So, how does music sound which is subject to warp speed ? I'll give it a try :

It is explicitly NOT about super transients and this is the first to occur. What it is about is a "glide up" towards any transient which occurs within say less than 1 ms, while previously that "glide" comprised of maybe 5 steps. Now try to compare these 5 steps with a perceived unlimited with mentioned "glide".
The 5 steps sound rough and hard (I did not say harsh per se) while the infinite number of steps (hence glide) just is not able to show and hardness. This, while the transient is still the same transient, were it about the time needed for its development (from zero to max). Mind you, I talk about a development (or envelope) of 1ms, indicating how super short this is anyway; maybe it is even shorter, maybe it is way longer (like 10ms). But anyway very short.

The sort of sadness which is in my mind, is that this highly depends on the Silverstone card to begin with. So, that firstly allows for the "separation" I talked about (on/off working faster hence better), while by that this seems to imply that in the gaps again "music" is missing. Remember by best example of the "gaps" : a tambourine like we perceive it in live fashion. So first the Silverstone (by accident ?) allows for this separation, while next we need more resolution which I could enable by software.

If I now listen to Fireball or Who do we think we are (Deep Purple) I just can not imagine/envision how any real live sound can improve that (son performing is drumming practice upstairs right through the music).

Another description which keeps floating throug my mind during listening : a total freshness while "fresh" now is a totally wrong phenomenon. Fresh (in my English) is equal to snappy and a overal glare (positively meant) of highs. One problem : today the glare has gone, which lets excel the snappyness which now also can sing. Just take that tambourine again, and try to envision that the hit on a snare is now able to make "bell sound" (bell - sing). And I don't mean false harmonics or other distortion, but if now so many more over tones are shown that wherever the bell sound is, it will now let hear that (and NONE of that before).

Maybe we can envision that a rougher sounding sound is undistorted fresh, while the same sound with added resolution will make a tone instead of sounding fresh. So, freshness gone, reality instead.

Another means of explaining it comes to my mind :
I think we all know about the ride cymbal, and that at least that can be nicely audible because it's the only cymbal which definitely sounds for real and singing when used as "ride" (which means hitting it in the middle and not at the edges). It becomes profound hence tunable (for the right sound) in tracks like No No No (Fireball). So, we know what I mean. But what changed here ? That Ian Pace appearently likes to hit that ride cymbal right in its middle all over the place throughout any more wild drumming. So what happens ? the freshness of the cymbals which never was any real good (because not real enough) has vanished, and the singing of the ride cymbal now gets profound.
This is how I mean that fresh actually has become more fresh without the hardness of it. Things start to sing all over and in the mean time there an additional dimension because you suddenly "see" things (like Ian Pace moving around as a wild man to each time find that ride cymbal).

The major contribution of latter description will be the Silverstone card because without the separation nothing will work. The injected resolution becomes profound in the real fast transients like synthesizers can do that; it is easily audible how "hard" the music gets when my little trimpot is in another position. This is undistorted by itself (blame the NOS1) but it is hard. Or cold if you want. Now put my trimpot in that best found position and actually all highs disappear. But what comes back is that warp speed now transient with colour and which has become meaningful. Not a tick, but a sound. I could also say : nothing is squared anymore at the somewhat lower frequency level; it has all become rounded and square for square you hear that and what you get from it is the warp speed.

I know, this will come across as contradictionary, but when the squares are there I'd say "hey, what a super system that it can follow that so well !", while the injected resolution tells me "warp speed - wow, how is it done !?".
So, squares are not squares (they really are not because they can't exist in Redbook at all), but will sound like that when not enough samples are present to render the (thus) somewhat more sine. Skipped samples theory again ?

Bass sounds completely different too. More rounded and with more varying colour. No standing waves, no zooming, so should not be wrong. Must listen more for the real merits of this.


Right. I think I have prooven that this trimming on the inside of the hardware indeed does a couple of things, but the more I explicitly work on things like this with the goal of what Nick has done litteraly with hardware, the more I should be focusing on what the real thing is we influence because it will make it better again. This time my focus has been "skipped samples" and all I do leads to indeed that being the culprit.
*If* it could be so that it is "just" about this, then creating the best SQ will be a piece of cake; just turn some dials and look for the most samples to arrive at the other end.
But I'm afraid it is noise we're dealing with after all and which very well can be because my approach about the skipped samples is coincidentally or a super high noise creator or a super low one (but it should be the latter).

Peter

Peter hi.

Your post on transient response above is really interesting, I get the sense that we are hearing very similar probably the same effects.

In my post on consistently tuning the PC  (moved to this thread above) in order to kept  things simple i gave only basic descriptions of the "sharp", "smooth", and "sweet spot" sound characteristics. This was to place emphasis in the post on the consistency of results obtained from all of the tuned points in the pc.

Your post brilliantly describes some of the qualities of the "sweet spot " sound and the "sharp" sound I hear here. Particularly you descriptions of transients seeming to be less "square" in nature but actually much more true to life. The fresh / sharp sound appears to have speed and attack but its a "hifi" sound, not a real and true to life. At the sweet spot you can hear, almost see, the "nature" material the instrument from and way for instance cymbals are being played. Also your  descriptions of bass reproduction is ringing bells (haha). The heavenly trimmer post over in the "hunting for noise" thread was also trying (less articulately  :) ) to describe similar sound changes.

For me it took my resent chance to listen to live church organ music in a cathedral for me to really be confident that the presentation of the PC and usb trimmed sound, with its different portrayals of transients etc, is truer to life.

This is great that similar changes are giving what appear to be consistent  results and I was smiling reading your post because you  nailed the analysis and description of the sound so well (and much better than I could put it across).  There is a question of the size of the overall change to sound though now. Are you planning to put high quality tunable clocks in at both end of the usb link soon ? I think the major improvements on offer from the usb link might come from this.

Regards.

Nick.

Peter,

I just reread you post again. You absolutely nailed describing what is happening here  :)

Are you doing this in software   :)! ! ?

N

Hi Nick, super stuff again.

Yes, software only, but mind the Silverstone card which is needed for the good base.

I guess I have a very strange disorder in being able to mimic sound from description (or hearing) in software and then by this (still asumed) bit perfect means.
I think I recently explained somewhere how a Jan Garbarek sax sounded nasty when a first version of XXHighEnd was out and really not a single SQ control was in there yet. So in an evening I thought of "how to" and Q1 was born. Worked next day and Garbarek became listenable (which is a miracle to begin with). Similar happened dozens of times with Bert, him applying something to a speaker filter, telling about it, me thinking of how to mimic that by means of software. An the other way around btw (so credit to Bert there) and which is how we both now ended up with the ultimate playback chain (the whole lot created by our co-operation).
I tried the same in the context we are talking about now and possibily that suceeded. Of course it takes some knowlegde of the OS + PC and what actually can happen, and that is what you (and Paul) described.
That's really all. :swoon:
:)

All I now need further (I'm sure) is something like fine tuning at maximum the frequency of the processor. Quite undoable. At least the last 3 hours brought me nothing yet.
Anyway there's a new Q4 here so you can focus on that one later ...

Peter

Hi Peter,
What Silverstone card are you using? I have one laying around.
In the meantime I did some changes on the bios BLCK setting! Tested different settings, 93, 100, 99.8, 101, 101.5 etc Still more to try. And yes they make a difference. Currently my best setting is 94.6.
My daughter is helping me, she is 13 and hears the highs better than me, I am 50.
She complains about aer hurting highs with most of the setting, but the 94.6 gives the best results so far. Maybe there are better settings.
Feel free to delete this post if off topic......
Regards

No way OffTopic Arjan. My Silverstone is a SST-EC04-P. 6 or so of them exist, and I never searched for the difference.

Hey, the most nice that you (can) join this dicussion; the more the better.

And don't underestimate the value of someone around who just complains without you understanding why; maybe over here we must perform some test about who hears to up what frequency (me 55 here by now).

Regards,
Peter

Nick, here's a nice other one for you;

Remember sound stalling under W8 after 10:50 after a reboot - completely repeatable ? Sound continuing by moving the mouse ? Nothing to detect anywhere by whatever log files ?
Gone.

That issue not being there in W8.1 that sounding so different ?

Sit back and think what this tells (not really referring to W8.1).
Peter

Hummmm there is a riddle   :wacko:

Hi Peter,
My Silverstone usb is the same! I replaced it by the PP. What is the trimpot? The round thing underneath the clock next to the molex connection? And how does it work? I can do some tests with it.
Regards

Arjan, sorry for the confusement; the "trimpot" I referred to is just a software dial I have created in XXHighEnd here. So, not available to you all yet. Hopefully soon though.

Regards,
Peter

Hi Peter,

I've just shelled out for same USB card that you have there so that I can compare it to the PPA card before and after you release your software update.  It will be interesting.

Cheers,

Anthony

Ok, in the spirit of trying to understand a little of just what Nick and Pauls usb clock upgrades or Peters software 'dial' in the forthcoming XXHE release are achieving I looked into the asynchronous USB protocol today:  I did me some book learning.  In point form the following is what I have learned (please chime in and let me know if I am incorrect):

• Asych is one form of the isochronous protocol
• All forms of the isochronous protocol function by the host (the XXHHE PC in this case) spewing out packets of data at regular fixed intervals (1KHz or 8KHz) and varying the number of bytes in the packet to suit the transmission rate
• There is NEVER any re-transmission.  If data is lost for any reason it is not coming back.
• The receiver (the NOS1 in this case) welcomes the data into the FIFO (first in first out) buffer and clocks it out of there using a FIXED local clock.  It is the fixed local clock on the receiving side that means that the transfer is asynchronous.
• The clock rates at the host and the receiver do not match...they just don't, even though they may be rated the same speed.  When the clocks don't match the FIFO buffer will eventually empty or overflow both of which are bad.
• To keep the FIFO buffer within its population limits the receiver (the dac) will monitor the buffer and when needed send a packet back to the host (the PC) to say "whoa back" or "give it some spurs".  The host then alters the number of bytes in each packet.

 

So if I have this right, the NOS1 usb board does a little bit of work monitoring the FIFO buffer and then sending back packets to the XXHE PC to say speed-up or slow-down.  Then the USB card in the XXHE PC responds by doing a few calculations and changing how much data it throws in each packet.

Now, the NOS1 is not your average dac.  When XXHE upsamples to 16/705 that is 16 times more data than a simple Redbook transmission.  On the face of things this means 16x then number of 'correction calls' sent from the dac to the pc to vary the transmission rate but this may be alleviated in full or in part by how much data can be made to fit into a packet.  Peter will know this.

So how could the upgraded clocks or software dials result in an improvement in sound quality?  My guess is that by making the clock rates match each other more accurately that fewer 'correction calls' are made and therefore less noise produced in both the host and receiver usb interfaces which in turn has less of an impact within the dac and its production of jitter.

The next question therefore becomes is there an advantage by having super stable and accurate clocks in the usb interface?  My guess here would be "probably not" (nothing like sitting on the fence) because all that we should be trying to achieve is to absolutely minimise the number of 'correction calls'.  If the two clocks cycle at constant speeds in relation to each other over time then the number of 'correction calls' will be relatively low.  However the ideal thing here is to use only one clock for transfer, which is the slave idea that Coen, Peter and Nick had earlier.  I don't know how to do this or if it can even be done in this situation.

There may also be an advantage to using clocks that are super low noise in one way or another (I don't know which way really...just putting it out there).

Anyway, this has been my attempt to put this stuff in more laymans terms for some others to try and follow.  Please pick it to pieces and let me know where I am wrong and have not thought it through properly.

Cheers,

Anthony

Anthony, thank you for this mate.

Of course I sould have sorted this out myself but for some things I don't know where to get the additional time from. But what I planned for tomorrow was just call someone who really whould know. So even better when I'm now loaded with data. :)

That no resends take place is a bit of s surprise for me because it would be the property of "sending data" which is subject to error checking (contrary to sending "audio"). Well, the error checking is obviously there (see NOS1 Control Panel), but then no resends ? I'll ask tomorrow.

That the underlaying means should be isochronous (that guaranteeing the data to arrive in time always, meaning that the OS needs to take care of sending it in time always) is no surprise.

That the timing issue is solved by more or lesser filling the packet is something I could expect but actually did not know. It is fully logic to me now though.

All your subjective (?) observations of the real merits according to noise hence jitter in the end - I can only agree. But I guess (or hope) that is obvious by now.
That it would lead to one clock controlling it all, is also obvious, or logical. What is new of course, is that with one clock, only one "correction call" would be in order (for a playback session) to never have one again - unless the correction call doesn't lead to the proper expectatations hence the "setting" of the amount of data immediately leads to a "is wrong" and such a call is send out again to the other direction, and is observed to be wrong again - et cetera. But if so, the margin of what's right would be too small which then *IS* related to the underlaying speed indeed, as you pointed out so well.

Great work Anthony.
Peter

My understanding is that a checksum is performed, but there is nothing to really be done with the result other that some sort of alarm that information loss has occurred, which is what the NOS1 Driver front-end on the XXHE PC tells us.  That piece of information I got from post #52 here (http://www.computeraudiophile.com/f8-general-forum/distortion-time-domain-digital-sound-signal-human-hearing-very-sensitive-jitter-digital-devil-18676/index3.html), so not strictly from a textbook.

To be honest I was not game to post this information until I read that post today (John Swenson seems to have sensible things to say about computer audio in general) because I feared that I was not seeing the 'big picture' on the subject, but that post  resolved my doubts and I went ahead.  Sort of like all the pieces falling into place.

Anthony

Ok, careful then;
We have been "discussing" other articles from John where at least I could easily point out that he knows more than any other on this subject, but also that he clearly doesn't have the whole picture of it. (maybe this was 4-5 months ago).

Who, I think, can always be trusted is Ar-t, but he is always more talking in secrecy, a bit like I do. Just doesn't want to unveil everything because of his commercial good reasons. So needs a lot of reading in between the lines.

In any event it would be correct that errors of any kind showing up in the NOS1 Control Panel are audible, so let's say that resending is not in order. Or at least not in those cases, but with reference to disk read errors only showing up as errors to you, knowing that it doesn't make much sense to show you an error which wasn't an error really (because a retry solved it for net result).
Also, it would be my idea that it may depend on the chips used, and empirical finding (a bit like John is I'd say) not necessarily tells all. So, I'd still like to find the textbook story after all, meaning the USB spec and how it can be used or utilized.

Also look at the danger :
For years "we" copy the "statements" of others that Asynchronous USB is error checked and "thus" should end up with error free transmission. I am a first to do to while the "thus" is just my expectation because logical. And now someone tells it is not because of ?
Wikipedia would show a banner with "needs references" etc.
Still it can be correct, even without those references.

My reading in between the lines detects the perceived merits about "Tent Clocks". Did you notice ? maybe not, but I notice more. And so I also notice the sneaked-in underlining of Ar-t without futher words. Like I don't use further words on this little subject.
But to me it tells a lot about both good guys.

Or what about your post regarding controlling the jitter. It won't tell anything to either guy, while to me it tells everything (well, obviously - but all is about references).

Regards,
Peter

One more thing, which only testifies how difficult this all is (or should testify that) :

I am pretty sure that I am the only one as an "end user" who has the USB driver code, including all the Windows Kernel Code and including the FPGA image code (bought a license for that ever back). I recently told that already. Now :

I am also the only one (as it looks like) who pointed out bugs in there, not known to the original driver code programmers. This *is* highly related to our subject here. Still, I have not been able to have these bugs solved, no matter I could provide all the test data and means to reproduce it. Can't solve those bugs myself either, because a 10,000 programs are involved and it is just too much for a simple soul. One of these bugs is about not reporting errors while clearly audible they are there (yes, I am repeating myself from a previous post).

Message : I should know a lot, but I know nothing.
Next message : Anyone in this field with "statements" really doesn't get way with it so easily when I'm around.

I will try to find out more tomorrow, with most probably the same outcome (no resends - just saying).

Peter

I am thinking aloud now Peter, so please don't hold what I say against me. :grin:

So, we have a clock on our pc USB card that is spewing out information in frames (that contain the packets) at a set rate...sometimes it sends out data, sometimes it does not, but what the rest of the computer is doing (read latency) does not have a tremendous effect on the oscillator (the clock just turns over on its power supply and does its job) because if there is no data to be sent out it just sends out an empty frame with no packets just the timing information.  If the USB bus operates at 8MHz there are 8000 frames allocated per second.  Unfortunately there is no possibility of re-sending packets that contain errors because that will mean that the next frame would need to be delayed (the reference is here (http://www.tech-pro.net/intro_usb.html)).  That is basic isochronous USB operation.  Will a very stable, low noise clock/USB card make a difference here?  Perhaps...at least that seems like what Nick and Paul are reporting to some degree.  The very stable clock on the USB card is probably important from the perspective of it oscillating at the same very predictable rate relative to the clock on the NOS1 USB board in that it will not trigger a varying rate of 'correction calls'...I think that the only thing more important here than a stable rate of 'correction calls' is zero 'correction calls'.  So a good low noise power supply for the pc USB card and a stable low noise clock are things to investigate in the pc.

Controversial and ill-considered thought #1 - Does latency in the computer (i.e. not getting the data to the USB card buffer in time) affect noise because it causes empty packets to be sent to the dac when instead they should have been populated, thus causing 'corrective calls' to say 'speed up the transfer' and then once the latency issue resolves further 'corrective calls' are required to slow down the transfer rate....then there is another latency issue and the cycle starts again.

Hey Anthony, nobody who is thinking outloud can be held anything against. It can only be helpful.

I must digest your last post somewhat, but it triggers me to say this :

When you fire up the NOS1 Driver Control Panel, the USB streaming is active. This should be with empty packets, although I can't tell whether the contents is important and things are just "silenced". Also good to know : on the DAC board side (of the NOS1) i2s is always running, so no matter USB streaming is going on or not). So it's actually always making sound, though silent sound. Now :

What I can see easily in any situation the USB noise is not surpressed sufficiently on the outputs, is that 8KHz. Will be 130-140dB down, but it can be seen. Now what's important for our story here is that this noise is lower when only streaming is going on, opposed to when music data is in those packets, because the noise gets somewhat higher (say a few dB).

At this moment I don't know how much this is related to the "adjustment calls" or ... that this is just about the analog signal which now carries 1's and 0's and the transition to the other of that. For this to understand you must read Nick's postings about i2s noise, which in the end will come down to the same. Here : Hunting for noise (http://www.phasure.com/index.php?topic=2640.0) (scroll down until you see pictures).

Careful now, because I can be confusing;
What I say is that already those transitions from 1 to 0 and back will cause noise as per Nick's shown proof of that (such a thing). BUT (!!!) :

The noise I see rising is at the 8KHz frequency hence something about more steep current used. What I can't tell is

whether this is about all that pile of 0's and 1's and their transitions in there (keep in mind, it is all an analog signal, never mind we call it digital) which collect within the "one send" of a packet.
or
whether this is about packets getting larger once real data is to be in there (I just don't know because never investigated it, but I can imagine it (USB packets contain a header)).
With the notice that this already lurks for a combination with the above reason.
or
Whether once real data is transmistted, right away the "adjustment calls" start to happen and that this implies higher current = noise to be seen. But if so, I'd say that each sent packet implies soch a call (back), because it is still 8KHz I see, and nothing else (also not "white noise" getting higher).

AND

That it already can happen that *acoustical* 8KHz can be heard from the MoBo once MoBo drivers are not the best or whatever, and where this acoustical noise seems to spring from the normal cpu (ASRock 3930K situation) with no idea how such acoustical noise will imply electrical noise.


I just thought to mention this, because it seems related to your last post, Anthony.
Peter

PS: Is that 8MHz you mentioned correct, or was it meant to be 8KHz ? -> 8KHz is the rate of the packets sent.

Yes, KHz is the right one.

Looks like I have some reading to do...

Anthony

Anthony hi,

We are both a little sad to be trawling USB protocol information  ;). I have posted these points elsewhere but they may help again here.

lets assume for the USB link only that there might be four main forms of error / management interruption to data that could impact sound here (there are properly more, but for now).

1) Processing of resend requests (depends on usb transfer mode used)
2) Bit transmission errors
3) Audio word errors (out of sequence or missing words) 
4) Transfer speed management requests


USB Async transfer mode is intended for media stream transfers of video / data and as you say there is no USB protocol error checking, just signalling from the receiving end to speed up / slow down the link. So here the only point 1 of the points 1 to 4 above would not apply. The rest will apply and there will be no USB protocol handling / correction of errors so once the error has happened it will make it to the DACs as corrupt data and we hear it. Since the data lines are simplex (one set of wires carrying data for both directions) for the receiving end to issue commands to the transmitting end to speed up or slow down it has to interrupt the transfer. This is unlikely to be a "set and forget" tuning by the receiving end because the clocks at both end of the link will drift (I know even the temp stabilised Dexa clocks do this) which means that the speed up slow down requests will happen throughout replay.

Another USB transfer mode is bulk transfer mode. If you read back and look at the hunting for noise thread you will see that this is mentioned. I asked in one of these thread posts IIRC if Peter knows from the source code of the NOS USB driver if bulk transfer mode is used by the NOS interface. This could be very relevant to what we hear. With bulk transfer mode there are CRC checks performed on packets and transfer resend requests are made where there are errors. So in this mode it is possible that all four error types listed above could be happening.

The errors only have to happen at some thing like audio frequency intervals and we are going to hear the effects.

Finally wrt clock quality and the accuracy of transmission over the USB lin. Again as mentioned else where both the transmitting and receiving end of the USB link synthesize their 480mhz transfer clock speed by multiplying the 24mhz oscillator wave form by 20X. This makes phase noise in the 24mhz oscillator VERY important if there is to be a well formed 480Mhz link clock. Consider if one beat of the transmitting 24mhz xtal is say 25% faster this could mean that for 20 beats of the transmitting 480Mhz clock it runs 25% faster than the receiving clock so 20 beats at the transmitting end to a relative 15 beats so 5 beats missed.

This is an extreme (I hope  :) ) example, but its not hard to see that not having the 480mhz USB carrier running at consistent speed at both ends of the USB link could / will cause all sorts of data transfer problems.

Taking all this into account my view was when I started with DIY clocks and then selected the Dexas was "how would better quality clocks not improve the USB link transfer quality ?" (actually I did have some experience of improving USB clock quality from years ago  :) which helped. Then it was just a case of waiting for the free PSU offer that NewClassD run on Neutron stars from time to time to come round again)

Its important not to go down the "data is data" route of thinking when tuning the implementation of digital audio components. Practical experience is that if you buy into the data is data approach then there are some really big improvements that can be missed out on.

I think the best thing to do would be to use the information I sent over by email and take to plunge. Build the DIY clocks or put Dexas in. Don't get too hung up on transmission line lengths and noise etc. The effect of improved clocks is much much greater then the electrical effects. Three folks are up and running with Dexas with two more in the pipeline that I know of. Listening to the effects puts a very different perspective on the discussion  :).

Kind regards,

Nick.

Re: USB Clock Upgrade My Experiences (so far)  (http://www.phasure.com/index.php?topic=2784.msg29170#msg29170)

Nick, really this topic. And a you see you're even confused yourself now (yes, "and old cow" as we say over here, but maybe you can try avoid confusing possible useful posts from you by each line in it "referring to" ? let me add Please).

Anyway, might you have seen this as a question towards me : No, of course no bulk transfer is in order. But I must honestly say that once you start to relate things as you do now (previous post, but feel free to combine it with put quote) I have another Dutch saying : Can't make much chocolate of it.

Didn't I express the other day that I was a kind of disappointed ? It happened again. Not because of your serious quest to the whole lot, but because of a too much charismatic expressing which comes across as "I know !". Let me help you with quoting what bothers me (not all as much of course); please notice that this is all without proove that I can see and that it just takes things for granted from one angle or base that's worked towards; I can do that too but I think or at least hope that there's so much more context given that or I express explicitly to be possibly wrong, or that the context put forward has to be worked out first. So let's see :











Nick, this is almost all of your post. And no, these points don't need to be worked out because I think most of them have passed already with doubts, context for review, things against them and what not. So, you go your own way ? Yes, it seems you are. Here the foremost example, because dealt with a couple of times by now :

I put it in bold now. So, I claim these Dexa's are sh*t for Phase Noise. And as long as you don't come up with the plots of it, I will remain right on this. And no, it is not important that I am right on this, but to me it is important that you press your "statements" (on others) rather than finding out first and that you advise people to spend a lot of money which is worth nothin,g or can be - or should be solved by quite different means (and I hope it is clear enough what these means can be). And just saying : STILL you can be right, but not by just pressing.

Yes, VERY reluctant to post, especially because I am sure it is well meant on your side. Well, same here. If we really want to get somewhere that is.
Best regards,
Peter

Hey you dual Dexa clock guys, what about the USB cable? Does another cable still make a difference?

Some (audio) USB cable manufacturers claim an influence on dataloss....

regards, Coen

Wow Peter, you got out of the wrong side of bed this morning. I'm not going to dignify this with any form detailed response, it can just hang  ;)

Coen hi,

I have not tried, but can do. I'll give alternate cables a go and report back.

Cheers,
Nick.

Hi Nick,

That would be great!

regards, Coen

Anthony's Controversial Thought #2

Ok, I have been flat out at work today trying to get finished for Christmas when suddenly I have this eureka moment:  could the noise in the FIFO buffer of the NOS1 work in more than one way?  This is the scenario...data is being written into the buffer and data is being read out of the the buffer at the same time.  If we have 'correction calls' causing the FIFO to do more work than it needs to then in theory the FIFO develops more noise of its own...while it is reading data out to the dac!  In addition to the noise on the ground plane, could this noise from the extra FIFO activity be hitching a ride out of the FIFO buffer with the music file? 

A simple thought, but one that had not struck me earlier, and I apologise if someone else had considered this or if this is ill-considered in the first place.

Cheers,

Anthony

Thanks for the reply Nick.  I missed the bit in the protocol about the x20 multiplication of the 24MHz clock to 480MHz...interesting...I will have to look that up when/if I get a chance. 

I guess my main beef with the DEXA's at this stage is the hideous cost (nearly AU$2000 for a pair buy the time they get here even with the *free* PSU - that is about one-third of the cost of the NOS1) and the less-than-optimal implementation (the wires).  I don't know anything about their noise characteristics and can't seem to find anything on their website or elsewhere.  If you have some data can you please post it?

So I guess what I am getting at is I think that a comparable result can be achieved with more effort and much less expense by firstly figuring out the exact things that drive the improved sq and then developing a reasonably priced solution.  After all this should be available to everyone and who knows, maybe we will figure out something else along the way that will improve things further.

Personally, I think that yourself and Paul need to get together with Peter and measure some things and listen to some music.  If these clocks are as great as they appear to be and if Peter has not already equalled it with his software 'trimpot' then we really will have a much better idea of where things sit.

I cannot over-emphasize just how much I appreciate all that you are doing and on a lighter note how easily England have rolled over in the cricket (phew - we needed that).

Regards,

Anthony

Hi Anthony,

Maybe you make it too complicated by putting the emphasis one element (FIFO) too much. I mean, make it more general and anything can happen in your direction.

Side note, but related to some extend : Do we now finally see why any Adnaco (glass) doesn't help and can theoretically only make it worse once the, say, inherent noise is sufficiently under control ?

So, not so new to me. However, we (you) start to come up with other reasons for it to happen.

Let's recall what I have always been saying : XXHighEnd is capable of influencing the DAC, and I know of no other means than that it influences the jitter in there (at the clocking out of the D/A chips).
I have also repeatedly been saying that I can not imagine that this is thermal or other noise inside of the PC doing that, hence that we're influencing the noise that travels over the USB cable. Careful, because the sort of fact that I can not imagine this, doesn't make it truth of course.
But wouldn't it be far more logical by now that the kind of processes we talk about cause this ? And that these processes can be influenced ? Just think of the general term "oscillation" and how the digital data (read : pulses) coming over from the USB cable, hit the pulses in the receiver etc., and that causing the not yet understood nastyness. Actually all is sheerly "created" in-DAC, but the way (and whether) it is created is under our unfluence (now think very generally the number of bytes being in a USB packet (or frame if you like)).

So, *everything* influences to begin with (in-DAC I now mean) and the best influencers will be those with a pattern, preferrably at the ms level or maybe somewhat higher. Well, USB transfer is per 1.25ms, so that's a good start. :swoon:

Almost offtopic (because I can't do enough with it yet) :
With the software I am as far that I can be too close to limits, across limits and before limits. But I still can't apply enough granularity to really control it. However, it allows me to see things. So, I can repeatedly start playback, wait for a second or two and see the lot go out of sync which I can't explain otherwise than hardware running into eachother (which was my aim, so I'm placebo'd). Sadly I can not really proove this is hardware, because it can also be the cpu itself, where processor cycles just don't match for two processes needing eachother. Hard to explain. But all being OK for a second or two and then collapsing makes me wonder.

Anyway Anthony, when the noise sneaks into the data as you propose, you're suggesting jitter up to the degree of molested samples (but read missed and repeated) which in itself would imply more jitter than ever can be. This can theoretically be in order, but if so I wouldn't expect a better oscillator to sound better (for the D/A I now mean, so not USB). Btw, this can also be seen the other way around : how can a 200fs oscillator sound so much better than a 1ps one while all "scientific tests" indicate that we won't be able to perceive the difference. But alas.
So, I can not imagine that such a thing is the case (high noise in the digital data) *unless* this is again at the 1ms level which makes it happen once in the 1000 or so (768K) samples). Thus, jitter like 1ps is way more low than 1/768000 which allows for the 1/1000 not to be perceived as jitter but something quite else (higher level distortion);

We then could say that once per 1ms or whatever it is, we have high jitter for a very small fraction of time (something like the slew rate of the electronics) that possibly being so bad that a sample is missed or repeated.
And something like this happening once per 1ms gives a nice flair to the sound. Or practice : Once the 1.25ms USB transfer is visible at the outputs, this is perceived as more bass and blanketed highs. Easy to hear and thus easy to measure while this noise is say 135dB down (but rides on the signal).

hmm ...
Peter

Peter, sorry to arrive late in the party, but as I understand, you are able to have the Silverstone USB card to work without having the molex connected ?

Alain

Alain - No ?
Maybe I wrote something wrongly somewhere, but no. And it seems a virtue to SQ that it does need the Molex (but this is just in comparison which normal MoBo USB and that other card I have which does not need it, plus the report of others with cards that need it ...
(sorry to be complicated)

Peter

No problem Peter. I ordered the Silverstone card - the first PCIe standard USB card I have since I got my music pc (the 2 I have are the Sotm USB PCIe card and the PP v2 USB PCIe card... Of course, I do not leave each of them in the pc - just one at a time...

Alain

With the Dexa's (USB clocks) firmly established in my system they (or at least the PC Dexa) is less established in terms of its final location. So today I moved the clock and its PS from its temporary location (lying at the side of and outside the PC case) to the inside of the PC case. First I located the clock on a "dummy" PCI card with the PS on the inside back of the PC case. Mmmm something did not sound quite right. So I moved it back outside - OK sounded OK again. So then I tried another location inside the PC this time well out of the way inside. But still the excellent SQ was reduced - slight untidyness in the higher registers but the bass was impacted. This maybe a small change in SQ compared with where my system used to be but it highlighted one of the really big changes the Dexa's made to the SQ and that is in the bass.

Just moving the Dexa's a short distance to the inside of the PC case impacts the bass. It gets a bit "fat" and overblown. OK this maybe a small change overall but now that I am used to hearing excellent bass quality with the Dexa's when it is diminished even slightly you really notice a difference.

OK I will find a way to install the Dexa and its PS in the PC without impacting SQ (parts on the way!!). But this brings me on to the main reason for this post and it is that Less Bass really is much More Bass now in this system - tight controlled accurate sounds on the face of it less than fat and bloated (even slightly). And it also goes right through the mid range impacting the accuracy and tonal quality - at least to these tired old ears it does.

Another always surprising thing is that with each of these improvements is just how much better poor quality recordings sound.

Onwards and upwards to 2014!!

Cheers

Paul

Airborne radiation affecting the clocks Paul?

Hi Anthony - yes possibly also could be earthing I had to connect the ps and clock earth's to that of the pc. Whereas outside the pc they are floating.

But the strange thing is that the Dexa website states that the mounting screws are earth points, however, I checked them out and they appear not to be connected to anything.

Anyway ps and clock isolated from "earth" inside a project box connected to pe hopefully will sort it.

More tests today.

P

Today I found a home inside the pc for the dexa clock and its power supply. They are both tucked well away in the bottom of the hard disc partition of the pc case well shielded from the rest of the pc behind metal partitions and fully isolated from PE.

The surprise (or maybe not that much of a surprise) is that the sound is even better than when they were outside the pc case. A very welcome result. The sound is incredibly smooth but remains dynamic and and attention grabbing when required.

Nick is coming down on Friday so it will be interesting to see what he thinks - things have progressed muchly since his last visit here!!

Mani - you are welcome to join us let me know.

Cheers

Paul

Hi Paul,

This seems to confirm that a PC is a bad place to be for a sensitive device and that it would make sense to have some "modular" approach in terms of physical isolation... Not really the way a mobo nor a pc case is really made for...

But it sure is interesting :)

Alain

Hi Alain,

I am listening now to an Album I know really well Wise Up Ghost - Elvis Costello (a stonking good album by the way IMHO). It just sounds superb no hint of harshness a very definite improvement. Elvis's albums have never been at the top of my list for sound quality over produced and hard sounding generally but now I am beginning to wonder. It just does not sound hard anymore.

The dexa modules were isolated from PE outside the case and are also now isolated inside the case. So that just leaves RF. As you say very interesting.

But also interesting is that these (what must be) small changes to the clock on the USB link have such a clearly audible effect on the sound. Whatever the reason USB clocks sure are important to sound quality.

It would be great if in 2014 we can find out why!!

Cheers

Paul

Hi Paul,

Well I discover how clocks are everywhere and they interact with each other... I never really read this anywhere else than here, at least for the first time. It is important indeed...

Couldn't we have our "Big Ben" super clock for all USB ports :) Let's wish for the best !

Yesterday, someone came to change the analog electrical counter for a new, wireless one. I am less convinced about the outcomes of the wireless things, but (it could be expectation bias) I sense there is something in SQ that is better... ????

"Everything Matters"...

Alain :)

Hey Paul,

Position of the clocks certainly has audible effects, my nos is still looking like a project because of this. No top on the box still, maybe some time in 2014 it will finally go back on haha. Right now a long coax for the clock signal and have options regards placement. Seems the best route for SQ.

I found some extruded aluminium boxes for clock and psus at last after a long search so will give these a go within the pc and nos.  The hard drive bay sounds like a nice solution.

I'm looking forwards to a nice session listening on Friday without too much A/Bing to do.... for once  :).

Best,

Nick.

Hi Peter,

I have been struggling to both comprehend and formulate a reply to your last big post for a while now.  My biggest issue has been getting my head around the many issues that you raise and my beginners perspective on them as well as trying to see the 'whole picture' so to speak.  I have a few questions that might put you on the spot but here goes.


Could the pc noise also be influencing the analogue output stage of the NOS1 via ground loops (along with the dac clock)?  If my understanding is correct this would most likely show as a distortion rather than as jitter.



I can follow this reasoning.  It makes sense.


This is where I am lost (I just don't know the theory) but I will take your word for it Peter.  Please don't feel the need to explain it to me.

In trying to make sense of this I have begun to wonder if there is improvement to be made by improving the power supply for the NOS1 USB board.  Is this something that you have trialled Peter?  I ask this because for the ATX LPS project we have some very low noise supplies that would be suitable for the task, and once they are built and tested perhaps I could try one of those.

Regards,

Anthony

I now see that I made a mistake here. This must be 0.125ms.

(more later)

I don't really have the time to write in as much detail as I would like, but I'd like to share my experience of visiting Paul's (Scroobius) place earlier this week.

A few weeks ago, I asked him to take one of my two NOS1s and change the clock to a Dexa (and power supply). He completed this last week, and on Tuesday this week, I took my other unmodified NOS1 to his place to hear what the Dexas are really doing.

The standard, unmodified NOS1 sounded great, of course (using the mobo's USB3 port). [As an aside, I have to say that Paul has really hit gold with the rest of his setup - it's very, very impressive, especially the LF performance.] The 'clean', 'clear' sound of the NOS1 that we've all come to love is there in spades.

We then switched to the Dexa-modified NOS1 (using a Dexa-modded PCIe USB3 card in the PC). It became clear to me immediately that some sharp edges had totally disappeared from the sound. I wasn't really aware that they were there until we switched. The sound took on a 'wholeness' that is hard to explain. Much easier on the ear. Sweeter. But still totally detailed. It's almost as if the sound was 'pixelated' before and became a lot less pixelated with the Dexas. I had a strong preference for this sound over the non-modded NOS1 sound.

There is no question in my mind that Nick and Paul are onto something here. The Dexa clocks clearly change the sound of the NOS1 for the better IMHO.

But I feel there is one caveat though: Paul cannot use the Q5=1 setting in XX1.186 without his PC locking-up. I haven't played around with the Q-settings enough to comment, but it could be that Peter is able to replicate the effect of the Dexa clocks through software. This would obviously be brilliant... and a lot more convenient for us all.

As a final comparison, we switched to Paul's own NOS1. This has the Dexa clock and power supply mod (exactly like mine) but also has a bunch of Black Gate capacitors replacing many of the standard caps in the NOS1. OMG! The bottom end just 'explodes'. There's oodles of LF energy that is simply not there with the standard caps. A grand piano now has the presence that it does in real life, and no longer sounds like a small upright piano. I don't want to go into any more detail describing Paul's NOS1 because there may be some controversy here about changing the caps in the NOS1, but I will say that I preferred the sound of Paul's NOS1 to both my Dexa-modded NOS1 and to my standard NOS1.

Just wanted to share...

Mani.

Mani, this feels like a joke. Maybe you missed a couple of 100 posts, but that Q5 (and 3 and 4) *IS* about mimicing this and it is 100% explicit.
So are you kidding me ?
Too bad that it is exactly Paul who can not use it. Maybe it isn't even coincidence.

*IF* Q3,4,5 can mimic what is happening, then I can wonder what will be happening to those Dexa implementations. At least it won't be the same, but I also can't predict what will come from it. That things just stall has slipped my mind but is a too long shot at this moment. I guess you did not try Q5=1 with your old NOS1 at Paul's ?
Otherwise let's take care that this topic won't be about Paul's problem; there is a topic for that.

Thank you for sharing this Mani !
Peter

:offtopic: well, kind of.


Mani, I'd like to advise you the most strongly to let your other NOS1 untouched. You don't know what bass is. With your new speaker on the horizon you will though.
Now think of it; it is me who tunes the speaker and this is obviously done through what I have here and this is (also obviously) unmodded NOS1. If you guys "claim" that you can squeeze out so much more bass it will 100% sure (did you read ?) for the worse regarding your new speaker. OBVIOUSLY.
That it should be wrong in the absolute sense is something quite else, but there each his own (because each has another speaker with so many failures). Remember, I measure as well.

Hope this makes some sense ...
Peter

Yes, I know you're trying to mimic the effect with Q3, 4 and 5. But the Dexas might still be doing something else too, no?

Anyway, once I get my modded NOS1 back from Paul, I'll be able to play around with the various Q-settings and see if I can replicate the Dexa-clock effect totally using the non-modded NOS1, simply through software. Again, this would be superb if it were possible.

Quick question Peter. Is it actually possible to take some measurements of the affect of Q3, 4 and 5 on the USB data transmission? Just wondered.

Mani.

LOL. You really travel too much. I think this is all in this topic - 3-4 weeks back. But I can't do it - Nick maybe. But obviously not yet or otherwise he undoubtedly would have reported.

Regards,
Peter

Haha... I'll believe it when I see it!

No. I will always keep one NOS1 totally standard and totally untouched. Of course, I will compare the modded to the standard with the new speakers and decide which I prefer.

But thanks for the warning.

Mani.

Anthony,


That is beyond my imagination because of the too much influence "we" can imply. I mean, by now the difference (seen over XXHighEnd versions and/or PC Settings) has become larger than a destroying oversampling DAC vs the NOS1. And *that* can clearly be seen my measurement.
If it all is about jitter it would make much more sense to me although I'd still like to see this in similar measurements as just referred to. But *that* does not show at all ...


Not so much the power supply, but the whole "system" around the USB interface (including the USB interface itself). I have that (and running) but only partly and it can't be trusted for SQ merits at this moment. Btw, not working on that either.

This is a bit stupid maybe, but replacing the PSU itself could do wonders while I never tried. So, just put in such a main PSU (the one in the right leg) and done.
(not entirely true that I did not try because what I refer to above uses such a PSU).

I can only keep on saying that most probably we are persuing very difficult targets - at least difficult to try out because we don't even know exactly what we are doing :) while I can clearly see that a bit mangling with the software takes us further. In my view this does not count for things like replacing a PCI USB card like that Silverstone but I regard this as pure coincidence just as it is coincidence that for once *I* was the one coming up with something like that. It wasn't on purpose you know ...
Just as that it never has been on purpose that we all (without exception) started to use USB3 because it sounds better and one guy just tried it (two in parallel actually). Why it sounds better ? I always doubted it because I did not see the reason. I still don't but at least today we are way (WAY) further in the acknowledgement that this is an explicit field to explore.

The latter is the real answer to why I never tried a top notch PSU on the USB side. And how could I know. All *I* know is that I was the first (and maybe still the only one) who actually put async USB to hell because it implies the most jitter. Well, tackling that part was difficult enough. Might you have followed Mr Rankin throughout time, you would see him tell first that async USB can not "carry" jitter, next that his own DAC doesn't measure the best without clear reason, next that "hey, jitter seems to come through anyway without me understanding" (which btw is not true at all) to end with about today that he still doesn't understand. So, allow me.
Well, the latter maybe because Rankin wasn't the first with async USB no matter it is claimed all over the globe. Guess who ? (hint : the answer is not *that* obvious).

Anyway by now this is all from quite some years ago, and today I approach this all very differently; Just don't give anything the smallest chance to deteriorate at whatever level or part, no matter you think it won't influence anyway. And you don't want to know what it all takes to cover for this all. But some day ...

Regards,
Peter

Mani you gave a very good summary and yes there is more bass energy but that is not the most important part of the bass story. I have spent literally months getting the bass of my system right. My big transmission lines generate a lot of bass energy and there are some big interactions with my room which is an absolute dog for acoustics. They needed a lot of love and care in room position, changes to the "stuffing" and crossovers etc.  Also obviously I have done a lot of listening whilst modifying NOS1.

IMHO the single most striking aspect of the bass now is its quality. Really this is bass of the highest quality it is tight fast & accurate. I did warn Mani that for many recordings he may even find them reticent in the bass - because if there ain't any bass they don't play it.

For sure I get fewer room interaction problems now with the BG's and that is counter intuitive given the "increased energy".

When Nick came down we walked around the room and the previous big room interactions are now much reduced. This absolutely is not just about bass energy.

Peter has made a very good point in that he has voiced the Orelo's with a stock NOS1 and we have to be careful.

Mani  -  obviously it is entirely up to you whether you want the BG caps installed but as I will be coming up to yours with the modified DAC we can spend time to find out which you prefer. We can try your other "Stock NOS1" (the joys of owning 2 NOS1's ha ha) compared with the modified NOS1. But also it would be pretty easy to move the Dexa's to your unmodified NOS1. So all combinations are possible.

It's good that you will be able to spend some time with the Orelo's and your stock NOS1 before I come along.

As far as Q5=1 is concerned I can actually listen to Q5=1 but it is also possible that I won't be able to stop it either and a reboot could be required.

The latest settings  6x10 1 1 1 &  120  Peter advised work well here and I can clearly hear the difference. But I have to say I will be very surprised if Q5=1 can do what the Dexa's do. But there again I have been wrong before (many times ha ha). But hey we can try it when I come up.

Actually it has to be interesting that I CAN hear the Q5=1 difference with the Dexa's installed.

Could be a long day.

Cheers

Paul

Peter,

I would really grateful if you would clarify exactly what you mean by


I am sure I misunderstand what I think you mean by it.

Regards

Paul

Hey Paul,

I hope you got me right, although it would be an implied message : Totally nothing wrong with whatever and who wants to do. It's only that I will not be able to apply your Dexa('s) or caps or anything while making a speaker. At least not at this time (and as you'll understand can take years until all is sorted out definitely). So no implied "strange acts" needed from your side either. If Mani has that one unmodded NOS1 all will be fine always and indeed he can choose.
So, clear ? Yes.

Otherwise it is exactly like I tried to tell it; there is so (really !) much wrong in ALL of our chains, that everybody needs special attention or modding if you like. So it's only that I cannot anticipate on that all for a general product (that coincidentally being a complete system, outside the mains supply (sadly).

I hope all is fine.
Peter

PS: Paul, I'll hope over to your topic about your last post. Give me a minute.

Who in the US can do all these clock/BG mods.

Any of you Clock/BG modders want to holiday in Destin FL.? If so I can set you up in a beach house of one of my clients in exchange for work done on my NOS1 :)

We just compared agendas here, and from of June 23 up to August 24 would be fine.