XXHighEnd

I'm not sure if this is particular to 0.9z-1, but I now have an 'issue' with the sound of Arc Prediction. When it was first released, I thought it was a revelation, exposing detail that just wasn't there on 16/44.1 material.

However, I've just done a pretty thorough listening session and have come to the conclusion that AP is making everything sound too 'thin'. The body, weight and substance of instruments is definitely diminished. I have three versions of the same track from Linn Records - 16/44.1, 24/96 and 24/192. Without going into all the permutations that I've tried, I can say that 4xAA on the 16/44.1 sounds sounds closer to the 24/192 version than 4xAP does. Furthermore, the native 24/96 sounds closer to the 24/192 than 2xAP does.

Peter, what has changed with AP? Anything? I really hope we can come up with something, because right now, I can't listen to it...

I did do a few tests with RME's Digicheck, but couldn't see anything that obviously points to a thinning of the sound. (There is however an HF 'click' on pressing play with 4xAP - and perhaps intermittently throughout the track also.)

Mani.

Mani,

I did not gave the latest (0.9z-1) version a thorough listen.
And I am restricted to DAP (88,2 is my dac max), so i was hoping (looking forward) to your listening experience.
Especially on the hirez stuff ;)

I have one question: is your signature still up-to-date ? I see you use 32 samples with Adaptive mode
Try 1024 samples instead.

Did you missed the 1024 samples discussion ?

Roy

Hi Roy,

Yes, I saw the thread and I agree with you - 1024 seems to be the sweet spot.

I've been all over the place lately with buffers, as I've been trying to improve my AES connection. What totally amazes me is that everything seems to matter, down to the voltage of the AES connection. In some instances, 32 samples is better, in others 1024 is better. With 4xAP, taking it right up to 4096 helps, but doesn't solve the issue IMO.

FWIW, the comparisons that I mentioned were done with the buffer set to 1024. I'll change my sig...

Mani.

4xAP seems to work well with HDCD.exe. This doesn't sound thin to me. Maybe I'm just losing the plot...

Unfortunately, I can't make a direct comparison with the HDCD decoder in my DAC as I now use the XX vol control and have no preamp in the chain (one of the best improvements in the sound of my system BTW, and I was a die-hard anti digital vol guy, until I tried what Peter's been suggesting all this time).

Mani

Hi Mani. Thanks for your clear reporting. And of course everything what is beyond that to get there.


Can you please clarify this a bit ? I mean, you did listen to 0.9z(b) as well, or didn't you ? Thus, if not, the quote above makes much more sense to me than when so.

From 0.9y to 0.9z things sure changed, and this change implied a change of KS being the best at Secial Mode and low latency, to KS Adaptive Mode at high(er) latency. This by itself was not intended, but this is how it worked out to be.
From 0.9zb to 0.9z-1 nothing changed explicitly BUT I can have made a 1000 mistakes in the code. This is related to the 24 bit upsampling, and *everything* had to change. This is why I am the most happy you had a glance with DigiCheck, which actually I should have done myself, but it can't be done for all the combinations.


Yes, this could be key, and I am serious. This is exactly what bugs me anyway. So, at least overhere I am in the stage of things getting theoretically better, but the sound may get worse from it. A maybe too simple example - but which shows what I mean - is the now being even of all the frequencies I have always been seeking for, and then mainly thinking of the SPL of cymbals. So, I was just listening to Heard it through the Grapevine (CCR) of which I always thought it was a nice cosey long track. It is early in the morning here, so I thought to keep it smooth at testing something. But this now appears to be one big long row of hitting cymbals throughout, and whoever wrote the track he must have been on cymbal dope. As said, the example may be too simple, but if you hear these cymbals all over and for 11 minutes long, they better be 100% good. And, since this is the foremost difficult "part" of it all, maybe they are not. Sweep in a preamp, and you may be better off in this case.

The interpretation of things become complex once things are overdriven. Also, it may be impossible to overcome it (overdriven = overdriven), unless the part concerned is changed for the better.

Allright, the above is totally unrelated to anything about "but this can't be XX !" ... the contrary. I am only trying to explain how difficult it is for myself.

At this moment, I too, am in the stage of something can happen to the PC at a reboot. Or maybe after a hang and a reboot. It happened to me day before yesterday, and besides I can see that the PC is behaving a little different (such as a stupid little thing the tasks won't go to the taskbar by means of the "blue screen" button as fast as before), and I think by now we all understand how important the response of the PC is. If that changes for, say, 10%, the sound will just be totally different. So, am I suffering from that without knowing what to do to solve it ? or did I too loose the plot.

Peter

My comments apply more generally to 0.9z vs. 0.9y. With the latter, I simply never had an issue with AP.

I've (finally) finished making up some cables that will allow me to record from analogue. What I want to try is recording the same track at 16/44.1 (with and without Peak Extend) and at 24/176.4. This is what I'll then do:

1) Tune the 'XX chain' playing the native 24/176.4 file until it sounds as close as possible to the analogue source going through the 'A-D/D-A chain'. This will entail playing with all the variables in XX, my PC, the PCI/firewire interface, power and cables. The biggest issue I have here is that I am going to have to use a vol control for the A-D/D-A chain. To make it consistent, I will use the same vol control for the XX chain and set the XX vol to -0.0. (The vol control will be an Audio Synthesis ProPassion connected to the power amp via 0.5m cables.)

2) Find the configuration in XX that gets the two 16/44.1 files closest to the 24/176.4 file.

Then I think I'm done.

Mani.

Mani, underclock/undervolt your CPU as much as possible and run it on a single core! In my system there is a huge difference between one and two cores (voltages, clocks and other setting are the same). On a single core I can pump up the volume to the limits, while on 2 cores I can barely stand the sound at 11 o'clock.

A maybe stupid question Marcin : What would be your opinion on an Appointment Scheme which excludes the second (and 3, 4) core completely ?

And don't you have problems at track boundaries now ? (ticks)

I don't know Peter, what difference would it make if I only have one core available? :D And no, I don't have any problems with ticks and pops, I only have to choose Low/Realtime priority for hi-rez files. I prefer High thread priority, I think that the sound is fuller this way.  Split size at 75 MB. Of course, my PC is hardly usable while listening to hirez tracks with quad arc predicion. My cursor freezes every 30 seconds or so, but no ticks :) Any plans on improving memory management? You wrote that it's impossible to load the whole album into memory. Isn't that Engine3 "stability issue"?  Anyway, like you wrote in other topic, the bar is so high right now...  :)

The question was a little different (please reread) and about the simulation of one core, instead of tweaking the PC (bios) so. Do notice that you'll always be happier with more cores at the preprocessing stuff. So ... during playback all must think there is one core only, and I can do that with a new Appointment Scheme. BUT, it won't be a hardware solution, and for sure no Undervolt solution. That is how I asked the question ...


As in : 192 x 4, right ? Hahaha, show me.
I can show you though. :swoon::swoon:

:whistle:

Yes, I know what you mean, but the difference is in power consumption of CPU and noise/ripple related to number of cores active (hardware). But who knows, maybe a special scheme for one core would bring some further improvements.

I meant 24/48 files  ;)

Let's call that underrated hires. :)

Mani,

From your description, and from what I can hear myself, it could be a 16bit -> 24 bit issue.
Notice that HDCD is converted to 24 bit before (thus outside of) the normal Arc Prediction program code.

You didn't see any anomalies to this (16 - 24) respect in DigiCheck ?
Or maybe now you know there could be a theoretical problem, you recall something strange afterall ?

Thanks,
Peter


PS: So yes, we both defined a clear difference between 192 and QAP within two seconds. But today the difference seems larger. Difficult to compare with the past for me though (DAC changes).

Hi Peter - Yesterday I compared (with my ears) XXhighend against Foobar with SoX Resampler - mod version (with resampled to 196) + Convolver with correction filters created by Audiolense.

I am not sure if others have experienced the same thing, but as usual XXhighend had better imaging and a more 'black' background. On the whole it is more musical. However, there were 2 things which XXhighend was weaker: 1) it was not as lively, in that it seemed to be putting its effort into creating a beautiful sound but it seemed to lack the 'oomph' for certain pieces (especially symphonic pieces such as Schubert's 9th or Beethoven's 7th and did not sound as dynamic).  2) it also lacked a certain detail in the middle register (for jazz pieces with a lot of details in it such as the 1st track of Chick Corea's Ultimate Adventure, or Corryell, Asad and Abercrombie's Three Guitars). I think that for 2) was a combination of Convolver and the higher resampled frequency (compared to XX's 176hz) but I can't explain 1).

When I switched to XX last year it was clearly better than Foobar but recently with these comparisons I am not so sure. I will experiment some more and update with more of my experiences, but perhaps if you have the time you can also try and let me know what you think.

Well, first of all thank you very much for your extensive reporting and explanations. Really good.

Of course, at this moment I can't have explanations myself, and the only thing I can do now is trying to put things in (some) perspective;

First of all, it obviously is very hard to judge any setup which uses third party (additional !!) solutions occuring on one side (Foobar/Convolver), with not such a solution on the other side (XXHighEnd). Of course, you speak relatively (a comparison with the past), but it still is as difficult to judge, unless you are goingto tell me you never changed those convolution files (which you can of course :)).
I could add to this that you could have created that solution for XXHighEnd just the same, if you had the hardware to do it.

But is it important ? ...

Your suggestion that the higher frequency used (192 opposed to 176.4) is a nice one, but I would forget about that as a reason. The whole method is different,, and since this too (SoX resampling) is a third party product and you didn't compare with that ... anything may happen.

But is it important ? ...

No, I don't think so, because it all will be too much apples and oranges. What *is* important though, is the relative change, when the past is compared to the present. And it is just this which IMHO can only flaw too, because too many things will have changed on both sides.

So is that important ?

In the end there is only one thing important : what sounds best to you with your environment and all the tools you can find to do that job, right ? And if that is Foobar with its possible additional tools, then it is that ...

So, having said this all, for now there is really one thing for me to do, and this is the attempt to make clear when apples become oranges, and comparison is not allowed. Or better the other way around : what to do to compare better, with in the end just the best means of playback. So, assumed we can agree upon this, here is my advice for today :

If you (with respect !) think playback gets better from room correction stuff, well, you better keep on thinking that, but I never will. It is just sheer impossible for me to think that, because all what I do at the far less than micro level to get things right, is destroyed at world level by such a thing. And maybe I am allowed to remind you about my thinking that GOOD playback removes everything which looks like a standing wave or anything, and no room correction is needed anymore. Maybe in a completely undecorated concrete room, yes, but then the base is just too wrong to begin thinking about improvement.

In between the lines : I know, when you are using XXHighEnd you won't be using the room correction (well, I think you don't), which is priceworthy for XX of course. So, by letting you know I understand this, I hopefully indicated that this isn't a debate or such; instead it is about how to approach things ... uhm ... in my view. :)

When you are using SoX resampling, you may wonder what to do that for. If you think the sound gets better from that, well, you must be wrong somewhere. And more nicely put : "It" must be wrong somewhere. The point is, SoX resampling doesn't improve on one single thing looking at it technically. IOW, if you think it does, maybe you can explain to me where that is. At this moment it is (IMO) one of the best resamplers, but it still rings as hell. And what are your resampling ? well, something your Weiss does in the first place. So, it is my firm conclusion that your SoX resampling has only chances of making things worse, and it had *no* chances of making things better. Again, if you differently, please let me know. However ...
There are so so many things which can be wrong in your chain which "allow" such process to make things sound better, that there is no way to even begin with explaining how it can happen and what could be the cause. BUT :
When Arc Prediction Upsampling from XX is wrong at this moment, this is always the explanation of course ! And so, we are never allowed to forget this possibility. And of course, it is in this context you wrote your post, and before you forget it : good ! and thanks ! :)
But still ...

Now notice how all starts to come together ...

What you most probably don't know, is that AI upsampling is just ... SoX. Aha. It may be more optimized than what is used in Foobar, it may be less - I can't tell that. The only thing I know is that I set the parameters myself so there's an optimum result for at least going from 44.1 to 176.4. Going from 44.1 to 192 shouldn't differ much, but it uses the same optimization as going from 44.1 to 176.4. I wouldn't bother about a possible difference anyway.

Now read back what Mani told a few posts back, and see the potential of the same technical merit (hence anomaly) you found. And you see, no matter you incorporated convolution, you still can be very much right ...

With this maybe unexpected change, you too could try AI (4x) with the knowledge the convolution isn't there. So, assumed the optimization of the filter in Foobar is equal to what I did, you will perceive the difference of the convolution not being there. From there you could hop over to QAP, and now judge what the actual merit of that is, in today's form ...

Now, and only now -but with some absolute memory on how things sounded in a "before" situation- you have created apples and apples.
And now you know why I thought it was necessary to spend a few lines of writing on this, just because it is important. And you also know I am not here to defend XX or anything, but to improve it, or keep ahead if you like.

All 'n all, if you would be able to just agree with Mani's findings, this is very valuable to me. Not that I won't believe Mani in the first place, but he too is not sure about his observations, because they are not consistent (why does QAP with 24 bits still do the best job ?). It is my finding too, though from another angle : just plain hires which is too much better compared to before.

Long story short : I think I must agree with you, but try to have the oranges out of the equation. Then I won't think it, but just know. The latter is because I believe you all unconditionally (but with the condition of leaving the orange out :)).

Great thanks,
Peter

Well, QAP has CERTAINLY changed!

Have a look at the spectra graphs that I posted earlier this year: http://www.phasure.com/index.php?topic=1017.0

Now have a look at the three graphs below:

The first shows the 192 file. This is exactly the same as it was before.

The second shows the 96 file with DAP applied. You can see that DAP does a really good job of recreating info above 48KHz.

The third shows the 16/44.1 file with QAP applied. It is different from before. There is a lot less HF content. How this could lead to a thinning of the sound, I have no idea...

(I still can't get QAI to play for more than 30 secs or so.)

More later...

Mani.

Hey Mani - Super !

And now a super stupid question : What do you actually do to create these graphs ? I mean, wouldn't this be dependend on the file played ? And that assumed, did you again use the Beethoven from before ?

Peter

Peter, yes I used the exact same files as before: Beethoven Piano Concerto No4 in G major Op58 - Andante con moto.

I would be cautious about these findings, if it were not for the fact that the 192 file is identical.

EDIT:

But maybe you're asking for more info on actual procedure? Nothing complicated. I just set 'peak hold' on and play the whole track. So, these are just the peak signals at the various frequencies.

Mani.

Really great. Who saves some, has some (dutch expression, and about your graphs from before).

I can't thank you enough for the effort. So, I will setup the same, and I *will* find it.
I hope. :)

Peter

It may or may not be related to this, but when i switched on 9z with my *studio* PC i preferred the non upsampled version to the 4x AP I was using before (hence making also the computer more responsive since i WORK on it at the same time ;) ).

I didnt make precise AB testing, and I havent done any on the real system, just feelings.
It could be very well that NOS playback improved a lot.

Hey Peter,

First of all: Thanks very much for really great and amazing sound coming out of XX.
Last weeks I have been listening to 9zb and to 9z-1 and here are my findings (16/44 flacs in QAP via original HiFace):

HF: 9z-1 seems like a bit thinner and clearer and a bit less harsh, meaning harsh is at very minimum anyways.
MF: 9z-1 seems like closer to listener so making the overall soundstage more round and full. Sound is even further away from the speakers.
LF: 9z-1 seems like deeper and fuller. Finally in my setup I really feel the bass.

Summary: 9z-1 is better than 9zb in all categories. The main advantage is that stereo is wider and sound is closer to listener, so making the room fuller of sound. Like it.

...and do not brake it please :)

I just can use 2XAP but I feel the same. It sounds unnatural and the sound seems to lose dynamic. With DAI I am still not sure what to prever. It has a better resolution than no-upsampling but it seems to cost a little dynamic. AI I would say is definitely better than AP. The only hook is, that it does not run longer than a couple of minutes, even in unattended. So I am using no upsampling.

Yeah, sorry about that. I am working on that right now (as one of the last things for 0.9z-2).

Thank you Peter! I am curious further testing DAI.

It's probably worth weighing in with a contrary view regarding QAP.  This (0.9z-1) is the first version I've liked it in.  In earlier versions I found vocals disembodied and ethereal.  Now they have more weight and are no longer a distraction for me.  Of course the added spaciousness that QAP brings is a real treat.  So Peter, please don't throw the baby out with the bathwater!

Danny - I will change nothing unless I can proove by measuring that something is wrong. This is what I am going to do next. I mean, the AI stuff has just been solved for 0.9z-2 :heat:

Maybe for Mani too :
In the end I have no problems with QAP whatsoever, BUT, I think it was Danny here who indirectly pointed me at Peak Extension ... and that being off in my case (beause of all kinds of testing it had been off for a couple of days). So, I activated it again, and all was right (again). Now :

Can it be so that today you tested with Peak Extension while before it was Off ... or not even there ? (I didn't look it up, but I would think the latter)

In any case I will explicitly test it. And compare with AI too, since it's working anyway. :)

Thanks,
Peter

Given the posts in this thread I finally listened to 9-z1. (It had given me problems before cause I had 2x arc prediction ticked and it got all tangled up in a 24/96 file). Well, it seems that the sound of z1 improves zb, but I'm really wondering if anything was explicitly done to change the sound of z1? Afraid I was being seduced by the power of suggestion in this thread, I ran the WIFE TEST. Didn't take her long to conclude that Z1 is more "natural" than zb. Now, as we all know, the wife test is not always correct (don't tell her I said that) but in this area fresh ears are helpful.

To me the mid range of z1 is more resolved or firm, less "breathy" sounding (distortion?) with the female vocalists: Patty Griffin, Tomorrow Night, from 1000 Kisses; Sarah K., Vincent (16/44 version), Alison Kraus, Baby, Now That I've Found You. It seems that something is lost in z1 but it does sound more "correct."

I A, B, C'd between 9y-8c, 9zb, and 9-z1.

Hope I'm not crazy..... you can keep your opinions to yourselves.... :)

I've finally gotten my act together and done the things that I've been threatening to do for a while: 1) determine how transparent my 'PC chain' really is and 2) compare QAP applied to a 16/44.1 file recorded from vinyl vs. a native 24/176.4 file from the same vinyl.

You can read the results of 1) here http://www.computeraudiophile.com/content/How-can-influence-SQ (conclusion at the bottom).

Well 2) has just been really interesting. Firstly, I never knew how good 16/44.1 could be! It's pretty difficult to distinguish it from the source or the 24/176.4 file... but there is a small difference. The 24/176.4 file however is, to my, my wife's and to some friends' ears, identical to the source. Another surprise was how much better my 16/44.1 file sounds than the CD rip that I have (which happens to be HDCD-encoded).

But one thing I have verified for myself is that the current QAP (with Peak Ext) seems to be changing the sound. The native 16/44.1 and 24/176.4 files sound rich and full, like the analogue source. QAP on the 16/44.1 file sounds slighty 'thinner' in comparison.

[Peter, I'm happy to send you these two files - if you're interested, let me know how I can do this.]

Mani.

PS. All listening was done with HDCD-encoding in XX switched off, Adaptive mode and with the buffer set to 4096.

Hey Mani - Thanks ...

FileMail and problematic files (http://www.phasure.com/index.php?topic=721.0) should do the job ...

Peter


PS : I have been testing today with test tones and the DigiCheck Vector Audio Scope, and funnily enough, the only really 100% in phase filter is (Double and Quad) Arc Prediction. So on that matter nothing seems to be wrong with it. I also saw a VERY nice surprise from it :
Arc Prediction shows a relation between Peak and RMS Voltage which is exactly correct (1.414 factor difference), while Anti Imaging showed both equal !! The latter can be reasoned by me but is very hard to explain (but think that only a square can do this :)), but about the former ... I never knew how to see or calculate it, while it would be the 100% proof it is just right. So it is ... haha (tested with an 18KHz test tone). Notice this is all about the two shapes a wave can have at producing the exact same result (which is what I "found" ...).

Hi mani,

Drastic changes, from 32 to 4096 samples? ;) Did you get the PCI Adapter Case? Any other things on your mind?
I've switched to my second PC, completely dedicated for audio purposes. I've removed almost everything (according to cics' recommendations and my own) - unnecessary services, drivers, registry entries. This is on a Biostar TA690G mATX mobo with dual core X2 3600+ underclocked to 800MHz (at 0.72V!). CPU temperature is only 19* at the moment :D Anyway, the sound is still a lot worse compared to my new PC. Most probably this is due to inferior motherboard's PSU section and PSU itself + regular HDD instead of SSD. Sound cards should have separate ps inputs and feed from external power source. That would make things easier...

[moderator action]

Marcin, I removed your 2 x 3m picture, and I hope you don't mind. :) It couldn't be handled really. You may repost it in a smaller version ... if it is on-topic anyway (I really couldn't see what it was about).

Thanks Peter - I'll do this when I'm back home later this week. I'm really interested in hearing what you think of these digitised vinyl files.

Mani.

Yep, got the case. Will report my findings when I'm back in the UK later this week.


Interesting question! Actually, I'm very chilled right now. I've proven to myself that my hardware is transparent. This is so nice to know this. No more messing around putting PCs together, playing around with interfaces and/or digital cables. This is cool.

But I'm still waiting on Peter's findings regarding AP... If it's currently working as he intended, then that's fine. But for whatever reason, it no longer 'works' with my DAC. Don't get me wrong, it still sounds good. But I'm just losing the 'body' and 'weight' of instruments and music in general. Maybe this has always been the case and I've only just realised...

Mani.

Transparent, you say? ;)
Give me a favor and try this:
1. Run msconfig (from start menu, just type msconfig in search field)
2. Go to Boot tab -> Advanced options
3. Tick Number of processors, select 1, confirm
4. Restart
5. Play some music
6. Go back with remarks :)

You may be able to set number of cores in BIOS, but I'm not sure if your mobo supports this.

Hi Marcin,

This isn't speculation on my part - I explained how I arrived at this conclusion here:  http://www.computeraudiophile.com/content/How-can-influence-SQ (conclusion at the bottom).

The 'analogue -> AD -> AES -> PC -> AES -> DA -> analogue' chain sounds identical to the direct feed from the analogue source.

Furthermore, XX playing a 24/192 recording of the analogue source also sounds identical to the analogue source.

ERGO... my PC is OK. Nothing more to be done here.

Where I still have some work to do is in optimising the playback of 16/44.1 files via XX.

But look, I can easily change the number of cores from the BIOS - I'm currently using 2 and not 4. I'll happily try 1 core once I have some time.

Cheers,
Mani.

Mani, but that doesn't proove anything, at least not when it comes to digital playback with XXHE, right? :D

Mani,
I read your post at CA with great interest, but one question bothers me: first you hear all kinds of differences with different SPDIF properties and in the end everything sounds the same?
What did you do to arrive at this stage?

Huh? When did I say, "Everything sounds the same?"

If I make a change to the AES voltage, to the AES cables or to buffers, the sound changes. It's taken me a while to find the 'sweet spot' and I probably wouldn't have found it if it were not for Keith Johnson's input. And here was the issue:

From the Model Two manual:
“The AES Ground Isolators should be connected to all AES digital signal inputs and outputs at the Model Two whenever they are in use.
The AES Ground Isolators consist of a number of turns of precision 110 Ohm balanced, shielded cable fed through multiple ferrite cores, each having a different selected permeability.”

So naturally, I have been using these AES Ground Isolators since I got the Model Two. But in his response to me, KJ said:

“... I find that sometimes a ferrite core on the interconnect wire or more drastic changes make improvements for that particular hookup. However things could be better or worse for a different one."

(The highlight is mine.) After reading this, I decided to try without these Ground Isolators (going against the strong recommendation in the manual)... and voila, I found that they were making things worse and not better. [EDIT: Having removed them, I found the 'sweet spot' by using an AES voltage of 5V and my 'cheap' Sommer Dsub-AES breakout cable.]

So what exactly is the same? Well:
1) the analogue source
2) the analogue source being fed through the Model Twos AD section and then internally to its DA section
3) the analogue source being fed through the Model Twos AD section then through an AES cable to the RME card, through the RMEs mixer, and then back down an AES cable to the Model Twos DA section
4) a 24/4fs recording of the analogue source being played back on XX

These four sound identical, all else being equal.

Mani.

Aha, now I get it.
So with the newfound sweetspot your DAC is completely transparant!

I did not try this experiment yet but I'm not shure I will like the result!  :(

Do you have an AD/DA converter, or is it just a DAC? If it's the latter, you're not going to be able to do exactly what I did.

Mani.

I agree that it doesn't prove that my PC is necessarily optimally configured to play back 16/44.1 files through XX. But it does prove that my PC is optimally configured to play back 24/4fs files through XX. I mean, how could playback be any better than the analogue source from which the 24/4fs files were derived?

But I doubt the PC is strained any more when playing back 16/44.1 files than when playing 24/4fs files, even with QAP and/or HDCD - my understanding is that this is all done in pre-processing.

Of course, I totally concede that my analogue source, amp and speakers may not be of high enough quality to really hear nauanced differences. It is therefore here that I would now like to focus, rather than on the digital side of things. As I improve the analogue components, maybe I will start hearing deficiencies in the digital recording/playback chain...

Mani.

Yeah, that was my next question!
What turntable, cartridge, preamp are you using for this?
I guess if it is to 'low-end' the comparison will be veiled by this.

I'm going to record with a terratec phase 24 FW and play with a HongKong Dac.
First have to find a good program to record in 24/192.
And hope the needle is clean and undamaged enough...
The good point is that recording vinyl this way allows you to play the records loud (through XXHE) without any acoustical feedback.

Exactly.

My vinyl setup is:

Technics SP-10MkII on a solid metal plinth and mounted securely onto a supporting brick wall  (i.e. the most rigid support available)
SME 3009 ('pre-improved') arm
Denon 103 cartridge
Rotel RHQ-10 phono stage
Henry Matchbox HD single-ended phono to balanced XLR converter

This is not a particulary expensive vinyl setup, as far as they go. However, I know many vinyl aficionados who would claim that this is a very good setup. As I have already stated, 16/44.1 files that I've recorded from single-sided 45rpm vinyl on this setup trounce the equivalent CD rip.

Good luck with your recordings!

Mani.

Yes, I received the 'PCIe to 4xPCI Expansion Box' from VirtuaVia a couple of weeks ago. It was a disaster!

Not only did my latency go to pot (I couldn't even play Adaptive with a buffer of 4096 samples!), but it made my PC very unstable also. I was regularly getting the blue ‘screen of death’. I installed the ‘ExpressBooster’ program that came supplied with the box, which just made things worse – my PC would not boot up at all. I had to boot in safe mode and uninstall the software.

In any event, as you've no doubt already have garnered from this thread, I'm satisfied that my RME card is performing well plugged directly into the PCI socket of the mobo.

Please learn from my mistake - even though I received a full refund for the box (€189), this episode has cost me >€100 in shipping costs. I strongly recommend anyone to AVOID this box. However, I commend VirtuaVia for providing good customer service and a 'no quibble' money back policy.

Mani.

Thanks for sharing! You save me huge portion of frustration.
Did you think about feeding RME with linear PSU?

Well that was really the motivation for trying this in the first place. So, I didn't manage to test this.

But as I said, right now, I'm satisfied with my PC/RME/AES setup... SMPS 'n'all - I think Zalman must have done good job with the design of the TNN300 case and it's integrated silent SMPS.

Mani.