XXHighEnd

I have the following settings:

Special: Q1=1, buffer=32
Adaptive: Q1=1, buffer=32

Wow - Special sounds totally different to how it used to! It's drier... much drier than it used to be. My main criticism of the old Special was that it sounded a little too 'thin'. This is no longer the case.

Adaptive sounds... wonderful. Better than before? I'm not sure. But certainly as good. I likey, likey very muchy.

Mani.

This is 1 sample internal latency. Is this correct ?
(yes, I am asking you, not myself haha)

Ah, slight misunderstanding I think.

The Q1 'knob' is set to 1, but the caption above says 'Q1 32'. So, I guess it's buffer=32 and Q1=32, right? If I take the knob down to Q1=0, then the caption reads 'Q1 16'... and I get distorted sound.

Sorry, my mistake.

Mani.

No, it just is confusing. But I can't get the "32" on the button itself.


With a net latency of 32 samples, yes.

But you agree on the dry sound ? If you're only not placeboed by me (earlier today) ... :)

Absolutely not placeboed by you.

Special sounds much drier. So much so that I think you should check the code - I reckon you may have got Adaptive and Special the wrong way around! The overall sound is totally the opposite of how I had it only earlier today - Special was the 'sparkly' one and Adaptive the 'organic' one. Now it's the other way around... although I'd call Special 'dry' and Adaptive... well, just 'perfect' actually. It is very, very listenable.

Mani.

This is the best sound I've had a chance to listen to from my PC - congratulations Peter! However, I'm not able to go lower than 1024 samples in special mode, still it's an improvemnt from 2000 something in 0.9y-c. Haven't tried adaptive yet, I'm afraid that I may hear sth worse :D

I think so. Need to listen more.

I am sure it sounds better, at least in my system. The difference is clear to me after a second. Now I had trouble choosing Special/Adaptive. Special is sweeter, great for jazz/vocals, but heavier stuff sounds better via Adaptive, I suppose...

What a cr*ppy driver, it doesn't matter if I play with QAP or at 24/44 (even 16/44), I can't go below 1024 samples and that is for Special and Adaptive mode. Never mind, it sounds terrific

... Not to this post, but to the sound. I mean, don't get fooled because it sounds different (which it 100% sure does). And, enable yourself to come back on your own first findings. That is not stupid you know; you don't want to know how many times by now I came back on something I thought was better, at the fifth day of listening.

Why do I write this ? well, because I (again) have problems with proper judging myself because of doing two main things next to eachother : this software and the Phasure NOS1. I mean, at times I can't tell what causes what, while I have to proceed on both projects. Also, each has their own negatives (and positives) at changes, and when e.g. XXHighEnd gets worse, it is not said that this expresses through the current setting/setup of the DAC I am listening to.

If you all keep on saying that XXHighEnd got better, I will have to believe that, and take it as the standard. If with me things got worse for the net result, it has to be the DAC.
Man, I hope you guys can believe what differences hardware changes can make (like one resistor) if bit perfect software already does *this* to us. Sometimes it drives me crazy and I don't know what to revert to.

There's also - I am fairly sure of that - the physical boundaries which just are there ... somewhere. I mean, if we are capable of getting out the waves better, then "better" means in general better outlined. Better outlined means more "sharp", or more transient like if you want. This by itself puts constraints on the hardware. So, one of the things I noticed, is that the step at going from 176.4 to 352.8 does something in unexpected areas, like capacitors suddenly can't cope (read : you start to hear them and they start to bring a flavour).
You could think "but this happens to all DACs with a higher output rate", but I don't think this is so because of the ringing always happening (and that smears hence removes those (micro !) transients).

So you see, I can't do this alone. All input is welcome !
Thanks all,
Peter

Peter, I don't know what you have done, but...

Special sounds totally diffferent from before. This is to be expected, right?

BUT...

It's Adaptive that's amazing me. What did you change? Anything? It's like you've read my mind - I wanted a combination of the old Special and old Adaptive... and to my ears, this is exactly what the new Adaptive brings.

Downsides with Adaptive? Well, the 'standing waves' in room seem to have shifted, so I'll now need to reassess my speaker and seating positions. I've just lost a bit of bottom end. But the top is beautifully sweet. I really don't think my DAC is 'struggling' in any way whatsoever.

I don't think Special is for me... with my current setup.

Mani.

Good that you added "with my current setup". I believe that the differences between different setups, especially interfaces, could be vast.

Mani,


To be clear about things : this is not what I suggested. I was only talking anecdotally what happened to me a while ago, which looked as serious as Arc Prediction being wrongly coded for Octo. But I could solve it at the hardware side of things.

The new means of KS Special Mode (which is new indeed) again brings things down to another level, and I too don't think it's the best. But this is mainly because Adaptive is so much better. That too changed, but by means of a less explicit fashion (but it did anyway, and I know it).

Combine this all with what I said yesterday about cardboard stuff PLUS my added bass power, and you can see that all fits perfectly, including the bass side of things at Adaptive Mode which just falls into place at my site. The bass is hardware though, and what you said about it enables me to better judge Adaptive.

By now I think this must be the hardest job of the world -> try to see through everything and see the elements involved. So again, your help is the most valuable in order to see those elements.

Oh, if all is right, which the NOS1 you can move your armchair back to where it was. Hahaha.
:swoon:

OK, after a series of comparisons, here are some thoughs regarding different modes:

Special:
- richer midrange and that is why I liked it when I first hit the play button
- better bass punch
- gentle top end
- overall I get the feeling of easy listening

Adaptive:
- more details, especially at high frequencies
- bass could be better
- overall feeling - natural

Each has its advantages and disadvantages and it depends on album which I'm listening at the moment, but for me and my current setup Special is better at the moment.

Now i'm sure too. Did an a/b too.
The distortion is much lower, I feel wanting to raise the volume, which is always a positive thing in my book. 1024 samples as well. It's good that the same exact settings work with just... better SQ :)

Another remark from my side. Adaptive always sounds good and coherent, no matter what kind of music we're listening to. This is not the case when it comes to special mode. Yes, there are tracks where you think it sounds better, but it's just an illusion  :grazy: Just set a playlist with tracks of different genres and you will know what I'm saying. Special adds warmth and colour, but it's not a good thing in general (IMHO). I updated my signature and it's stays this way for a while.

My system is unfortunately not quite settled at this time as I am still burning in my new DAC.
What I can tell from initial listening is with Q1 set to 1 for both, adaptive mode is more relaxed,
beautiful on vocal music. Special mode sounds tauter, a bit more dynamic but a bit less extended top,
I think.

Hi Peter,
I think your new version sounds more clear and more "analog" than the previous. It is a good step foreward. In my ears special sounds more coherent and more clear than adaptive. Also adaptive has less "punch" in the whole frequency range. I prefer special in every way.
That is my first impression.
Thank you!

Does anyone notice this? With Q1=1 in adaptive mode, with large buffer, the sound is very relaxed, sometimes a bit too relax and mellow for my taste. When buffer side is reduced, the sounds get tauter. Somewhere between 128-256, the sound becomes too lean and bass is too thin. My ideal buffer is probably somewhere between 128-256 unfortunately my Lynx card buffer side does not offer anything in between.

Holy sh*t! Again, WTF!

Just did an A-B with 9y-8. Played Imagine, Eva Cassidy, in 9z louder than ever before. More resolution in the top end than Iv'e ever heard. Then played the same 9y-8. Can't believe the distortion that was there. Both versions set to Signature.

Awesome!

I regret I posted so quick. For now I have to say I am not sure if adaptive mode is better or special. I have made a tweak on my system because sound quality has become worse. I made a second user and started in debug mode without any startup programms. Now I was listening to 09z again. I feel better over all. It realy could be that the top end is more "cristal" with adpative mode and everythings just the other way round I felt before. I will wait a few days and report back again. All I can say is, the new version is better than the one before... :grazy:

^{I'd call Special 'dry' and Adaptive... well, just 'perfect' actually. It is very, very listenable.}

oh manis.. what would Romy the Kat say if he were here. ha ha!

Am loving Special mode so far. Now all XX needs is VST support so I can mess about with the phasing and X-overs.

Peter - how about instead of me going out and buying an RME AES I just pay you the money instead and you build in the support??

Call me crazy, but I think that specific order of playback settings matters (besides psychological side). Peter, is this remotely possible from a software/system side? Let's say that during one system runtime I'm trying 10 different settings in XXHE (for adaptive and special mode). Can you tell that each setting is totally independent?

I don't know what Romy would say if he were here. But I don't think he'd take issue with what I've written. Here are a couple of quotes from his website:

"What greatly surprises me is that fact that all playback software sound very different."

"So, I wonder what makes a WAV player to sound good. I would write up my own WAV player if I have any idea what makes a player to sound good…"

Mani.

For those having troubles playing with low buffer settings, try higher Q1 values. Endless path of combinations... Peter, what's your statement about recommended settings - keep Q1 at 1 with lowest possible buffer, even if it means 2048 samples or try higher Q1 values with low buffer settings? For me it's either Q1=1 and 2048 samples or Q1=7 at 256 samples? What's the difference in theory?

Looking forward to your reply,
Marcin

Haha, I guess if you put your finger over an RME card, a ToolTip pops out. Man, I can't even do that with software !
So it seems. :cry:

NO !
I discovered this myself a few days back, and discovered that a certain setting wouldn't play anymore at all, as if all the albums were badly ripped. I really needed a shutdown of XXHighEnd + restart.

I dedicated this to the same boleary reported a few weeks back, after which he started changing settings and got nothing to play anymore. Or something ike that.
IIRC I did something to prevent this, but could never really test it because I couldn't repeat the situation. Generally though, quitting XX and restart it will refresh any "memories" left, and start fresh. Some things are really complicated here.

From what you told I don't think it is related to you problem (can't play anymore at all), but from the quote above I'd say indeed there is some problem somewhere about things which get stuck for a certain (not working !) mode/setting.

So take it that you are right (I do too), but try to proove it is solved by quit/restart. If that doesn't help in your situation, it must be something quite different. beyond comprehension. Well, mine. :)

Peter

Good sound is back again, I guess. I've found the best settings from a system/XXHE side, but lowest buffer size is still 2048. I have no idea, how it worked at 1024, never mind... Now it's time for further hardware improvements and tweaks:

- second PSU
- HDDs out with external power connected via eSATA
- kuroutoshikou no-pci cards (http://www.kuroutoshikou.com/products/etc/no-pci+fset.html http://www.kuroutoshikou.com/products/etc/no-pci-express.html http://www.kuroutoshikou.com/products/etc/no-pci.html)
- linear PSU (maybe)
- good power cable and socket outlet

etc

To be honest Peter,

I DON'T like the sound of the latest version, :(
or something else is wrong, but i dont think so

I use same settings as previous version. (see sign.)
The sound is less full and rich. (less bass in mid region)
I hear no (lesser) wood in the acoustic gitars (Armik) for eg, I also played Beef (ska, reggae band from Eindhoven) wich I totally dislike on z version
Armik - Isla del Sol    is also a HDCD, i still like it better played with y verion even without HDCD coding.
I also find the "background" more dead, so less spacious

So I did some testing,

1- original cd played directly with plextor thru FW (Texas Instruments), used playback software foobar wasapi 24/44

2- Played with xx y version

3- played with xx z version

z version always finished last, how come ? (1 and 2 match the most)

Peter, I will send you some files, do me favor and compare them for yourself.
Let me know what you think
 :)
Best,

Roy

Hey Roy,

I will for sure try the files (if you only allow me not to A-B anything :)).

But I don't think using Adaptive Mode with (1)/48 is a good thing ... Better try (1)/1024 ...

Peter

I just send you those files, just play some tracks, its very obvious,
you dont need trained ears to hear the difference. (I also dislike A-B anything, as you know)

Thnx for advise, will take a listen to 48 vs 1024 (or other settings)
Should I use xx buffer settings or soundcard settings. (or is it the same)

Thnx

Soundcard 1024, and Q1 = 1.

But I'm also afraid you are comparing apples and oranges ...
If you play this (HDCD) album in 0.9y-8c it will be (QAP) upsampled, while in 0.9z nothing will happen to it. So, I can listen to it, but I will do it in 352.8 anyway. This is not much usefull ...

But I will listen to it because I like to listen to music, haha.

You are right, Peter

Its the buffer settings, much better with 1024
Now I have my bass and mids back

Thnx

And enjoy the music

But still don't listen to HDCDs if you are used to QAP !! (you may have missed it, but it is a bug that HDCDs don't upsample).

Anyway, good that it helps. And you heard it pretty quick ! (as you should)

This makes me curious.  If Audiodidact can play at buffer size=48, Q1=1, why is it better for him to play at 1024? 

In my case, I can get my HiFace to work at 256, Q1=1.  Should I too be playing at a higher buffer value?

More fun now  ;)

Can be heard in the first seconds, just listen to Beef and it is pretty obvious !

There is a hdcd led somewhere?
It's funny because i'm listening to one of the HDCD albums courtesy of Roy and i'm loving the sound (using QAP on the computer where i always use it).

But sadly it doesn't do that (QAP).


Maybe I don;t understand the question, but under where normally the FLAC logo appears, there will be the HDCD logo, if HDCD encoding is detected.

I think it depends what your preference is : KS Special Mode or Adaptive Mode. With Adaptive Mode I (personally) don't think the lower settings are better. Higher settings are more warm and of better color (all IMHHetc.O).

Peter,

With 9z I'm preferring Adaptive, and I used to favor Special.  I'll play around with increasing the buffer size to see how the sound changes.  Do you have any objective way of determining what the optimal setting is - what I should listen or look for if it's too high or to low?

Thanks,
Dan

Danny,

Just try some settings between 48-1024 samples,
Think as 1024 is maybe to fluffy in bass region
512 is more firm.

I stick with 512 for now

Roy

Thanks, Roy.  I'll give it a try.

Dan,

Apart from just listening (like Roy tries / just did) I would always incorporate the cpu behaviour. The higher the buffer size, the more normal it gets. Normal = consistent.
But it completely depends on the driver.

In the very end it is listening of course.

Peter

Let me check. I dunno if the album was hdcd or not. The numbers in the top left still show the 176400 upsampled bitrate.
Checking...
OK, so it shows FLAC and below hdcd.
This means that it's not upsampling, correct?

Yep.
Unless you use Octo/Double Octo. :)

And notice this is a bug. So, it will be solved.

Hi to you all!!

I've just purchased the XXHIGHEND player!!
This new 0.9Z was that amazing good that I just made my descission!

I am listening to it right now in Special Mode Q-1 /0/0/0/0

My system is a Gryphon Tabu / Ensemble PA-1/Profundo speakers and a Xindac dac-9 upsampling 192Khz.
I am just starting to know what is my system capable of!!!!!!!!
Laptop HP 1,66GHz 2,5 GB Windows 7 professional.

Peter, you did a very good job! I've tried J.River 15 / Foobar / I-tunes etc.and did not expect this kind of difference in sound. The Ensemble / Gryphon combo is obvious capable of a lot more I could ever dream of!!

Also my wife, whos is the best reference in listening, noticed the better quality immediately.

I am wondering what setings are better for my dac, wich is an usampling dac 24-192.
Am am playinf 16 / 176400 right now / Invert / Arc Prediction(does that work on 16 B.

Any helpfull information is welcome!!

Al the best regards Peter,

ELFUEGO also known as Ronald (from the Netherlands also)



 

I know that I'm hearing with a very good loudspeaker. Until version 9z I didn't know that I own a little gem! The main change from my point of view is exceptional coherency of the music. Playing quiet or loud - the air is full of flageolet, breathing of the musicians, noises of live performances (moving chairs, scrabble of shoes), plucking with the plectrums and much more. I disconnected the bass horn for a lot of music because it isn't necessary anymore!

Georg

Hi,

Have not yet updated my sig, but here is how I do it:

Windows7 HP, C2D E6300, HiFace USB device, KS mode, Special mode, Dev buffer = 32, All Qs=0 (the upper 2 values show 0 and 32), QAP:
The sound is amazing, there is bigger bass than before, more details and more space. The room is fuller than ever. Even my wife noticed that quality has grown since before versions. Have not yet tried Adaptive mode. Will report for sure.

One question: if I load in XX the FLAC file which is not from HDCD, should I still use Peak extension feature?

Ivo

Hi there Ronald. A warm :welcome: here !

Arc Prediction works on 16 bits allright.

I am afraid I can't advice you about the best settings for your DAC, which merely is because I don't know how exactly it works for when does it upsample to what etc. The thing I can tell you is that Arc Prediction was made for NOS DACs (taking 24 bit input), but that it for most of the DACs works for the better anyway. Best would be if you could tell your DAC not to upsample, so you will be sure it doesn't REsample. Thus, if you output 24/176.4 from XXHighEnd (with 16/44.1 source material) it will remain untouched in the DAC. There will still be filtering going on, and if you can switch that off too, it will be the best. Notice : if your DAC is a Sigma-Delta type, don't look for any filtering to shut off (it needs it).

For the remainder it is a matter of trying what sounds best to you ...
Kind regards,
Peter

Peter,

Thanks for your prompt answer!

My DAC is an upsample Dac and I cannot shut this off. Only to choose  96 or 192 /24Bits, but actually a very good sound from a reasonable priced Chinese DAC (899,00).
I havo to convince some High-end friends of the quality getting from a pc/Dac route (W&W audio Enschede) together with XXHIGHEND :-) I think from now an easy job. I will install my HP laptop together wit the Xindak DAC-9 on a very good system available over there and show them that they do not longer need a CD-player!!!
And I will try if I can get a NOS Dac from them to see if it's a better solution!



I will keep you posted about this!!


Best regards,


Ronald

I don't give you much hope for that ! :)
And they are not so digital minded either. Or at least a year ago or so there weren't.

And be careful with porting all to your laptop, while apparently now you don't have it on a laptop. Chances are fairly high it won't sound, or you won't be able to achieve the settings you are used to now. No big problem maybe, but don't think you can put it all on it, and switch it on overthere for the first time ...

Peter

Hi Guys downloaded 0.9z today & got it working. My impression of the SQ, in my set up, is that it is much more dynamic than the previous version. SQ is almost the same, maybe a bit cleaner in the top end, but nothing significant.

In short it sounds really good, as it always has & with a new GUI.

Arvind

The more i listen to it and the more it sounds "right".
My most sincere compliments, Peter. It's since 0.5y that I dont feel such kind of improvement. And this is with an OS dac of average quality.
Got the goosebumps since I dont remember when.

 :pleasantry:

After some listening and fine tuning of 0.9z I also have to give a reaction.

sound is AMAZING!! 

I use Adaptive mode in this version.
In the previous version I used Special Mode but that mode now gives a too hard, sharp high/midrange on my system.
One thing I don't understand is the higher buffer setting, with 2048 I have "too much buffer errors"  resulting in not recognizing my Hiface anymore!
Only a restart of the PC could solve it.  After setting the buffer size to 4096 I have had no problems anymore. 

Thanks Peter, for all the support and the great times we have with perfect sounding music

Thank you Telstar, this is very nice of you to say. I also know what it is worth if you say it.


And Arvind,


I doubt it whether others know what you mean, but I do (unless I am wrong of course) : The very most strange (and apparant !) thing which happened to this version, is the difference between loud and soft passages. It keeps on appearing to me while listening to known music. I actually can't understand how*that* in the world is possible. Unless everybody's "bit perfect" light turned black now ...

So, more dynamic indeed, but at the global level.
I know what I did of course (besides it stops Engine#3 from working haha), but I am totally clueless where this phenomenon comes from. I mean, I didn't even know it existed as something which can be variable from (CD)player to (CD)player.

Anyway Arvind, we know how your judgements are always correct, and of course it had to be you to come up with this strange change as the first person.

Peter

Hi All,

Same here, sound is really amazing.

Special mode is to hard and direct, resulting in listen fatiqueness after some albums or even tracks. (got even as low as 32 samples at Q1=-3)

Adaptive mode is REALLY REALLY great at 1024 samples, lower as 1024 samples will result in lesser bass, soundstage becomes smaller.

1024 Samples = ANALOG

Never heard such dynamics ever before, anywhere !

 :veryhappy: Thnx Peter

It's all system dependent. E.g. my amp has a little laid back mids and I think that special version adds colour in this range, but I don't like general presence of Special mode, especially top end. Like you said - adaptive feels organic and that's what I like about it.

PS
I've been experimenting a lot with hardware recently, I connected second PSU to 4PIN socket, which feeds CPU mostly, and connected drives to it as well. My primary, better PSU, feeds only 24PIN socket. Very nice improvement! I think I'm gonna try linear PSU

Although,

I think with such dynamics, speed and precision becomes much more important.
Precision by means of word-clock and speed by decent (enough) dacs.
Because sounds/samples are much loooooonger they can only get better and smoother by means of upsampling. (AP)

I think I run into my system (dac) limits,

Roy

I agree that I think a lot has to do with the rest of your system.

In my case, there is just a certain 'magic' at buffer=32 that gets lost for larger buffers. I think I can attribute this to 3 main reasons:

1. My RME card really seems to be happier at 32 samples than anywhere else. I'm not sure if anyone has tried KS Normal mode (why would you?), but my RME almost refuses to play without clicks and pops unless it's set to 32 samples.
2. I'm currenly using my old Pass Labs Aleph 4 amp - 100wpc, class-A and single-ended (not push/pull like 99.9% of amps). This is a seriously (too?) laid back amp.
3. I'm currently using Quad electrostatic speakers, which are nowhere near as dynamic as more conventional speakers.

For these reasons, Adaptive, Q1=1, buffer=32 seems to work well. Yes, not as full or rich as higher buffer settings, but then again, I just don't need more fullness or richness. A little more of the gorgeous, taut and rounded bottom end that I'm currently getting would be very welcome though :)

As for my DAC, well it just seems to provide an open window to whatever is fed to it.

Mani.

After some days of listening I can repeat my first statement. I like special mode more than adaptive. I first used scheme3. Here adaptive has more hf-detail and better resolution in the highs than special. I changed to scheme 2. Here special gains hf-resolution and it sounds cleaner with better seperation. It keeps it's dynamics and the sound is balanced. Adaptive has not this overall punch and there is too much focus on the highs (with scheme 2 and 3).
With special (Q1=2) and scheme 2 the sound is realy amazing!

Is it possible that arcPred has an invfluence to the sound even if no doubling(upsampling whatever...) is used? I feel it cleans things up a little bit but don't know if this is possible.
@Peter: Is there a "manual" for all the functions of XXHE? Some are not explained by the tooltips.

If all is right, not. With Attended, very maybe ... :)

For a manual, at this moment I am afraid this comes down to reading the Release Notes from the newest till the oldest, until you are fed up with it. But all should be in there ...

I am not using unattanded yet, so it is possible. If it has no influence, it is a nice placebo ;)

Better late then never...
Thanks to you boys I tried the higher buffer settings (1024) with adaptive mode and normal Q1=1 and wow!
The first evening I tried to lower buffer and Q and everything went fine and low, thinking this will give the best sound.
A little disapointed I stopped to go sleep and never felt like trying anymore, maybe back to vinyl?
So now it is back to enjoying the music, and yes the best ever!
Maybe a little extra tweeking and maybe let John Kenny do some surgery on the hiFace and you'll never hear from me again!  ;)

This trip never ends...,  :grin:

Remember me writing about buffer sizes and losing my rabbit? Guess what? After I deleted XXHE config files, made a clean install, I'm able to listen at 1024 samples again!!! Why? What happened? Peter would probably say - your PC is faster-  but no, it's not faster, I got 4 apps running, 40 tabs in Firefox open and my dpc latency varies from 200-350us. Maybe to low os latency doesn't allow for low buffer sizes with my drivers? That's magic for me. I won't reboot ever, afraid that my rabbit'd run to hole and I won't catch it again... The sound is 100x  better now. Again - why???

Hi Peter so am i made out to be the villian...he he...

Arvind

Hi Marcin,

Ok, to me it is clear that more might be going on than I can reason out. IOW I sure believe you.
Keep in mind though that I can not judge your "capabilities" of judging a system for its performance and everything. This can go down to stupid details even I don't want to know about, but can work (around) with anyway. Example : If I see my driver is behaving strange (and the RME Fireface can do so without (known) reason), I reboot or whatever it takes to let it behave normal again. This does NOT imply I solved the RME problem, but I solved the result of it. Why ? well, in fact only because I pay attention to nearly everything, so the leat I will know is what is "normal".

For everything is a reason, but it is the sport to find out what causes it. Again, this often means take away the cause, while the root problem never is solved.

So yes, something will be going on, but what it is I can't tell. But let me give you a hint, which is not such a common thing to come up with for a normal PC user, but which is a fact for me :
Your PC "lives" by interrupts. You know, the PCI stuff, but in fact everything. This all works by preset timings. Something like "if I want to read a block of data from disk, my request should be dealt with within 5ms". This 5ms is a setting and preset by the OS or drivers. The PCI latency program (which worked for XP) is an example of that those settings can be changed. But now think like this :
There is *another* setting, and it says that scanning a directory entry must be dealt with within 1ms. The pipe this goes through is the same as the other one, "reading from disk". Now, when the scanning is in order, this causes the reading also to receive 1ms, while the actual setting is 5ms. And thus, putting the scanning in a constant loop (consuming some cpu cycles) suddenly gives a 5 times faster disk access ...

This example is completely made up by me, so don't try any of it. But I always have the real life example of decoding of FLAC, which including the reading the FLAC file from disk, the decoding itself, and the writing of the decoded file to disk, goes way faster than I can achieve myself with the fastest means of a file copy in C++.
IMO there can be one explanation for this : the FLAC decoding keeps alive something (interrupt) towards the disk I/O, making that by itself way faster.

With this in mind, anything can happen up till "the more Firefox instances, the better it becomes" (up till a limit).

Also keep in mind the reporting of someone who found the latency (checker) dropping significantly while XXHighEnd is playing music. To me this makes perfect sense, although it may not happen to everyone like that, as the remaining latency can get worse of it (which would be the expected situation).

This is for your weekend ! :)
Peter

Peter, I  get your point and totally agree with you, but assuming my PC "lives", I can't control it or predict its behaviour for sure. That's why PC audio is so complicated.

Another funny thing, now that's everything is back to normal (in my opinion), I like invert more. And all my findings are backed up by my girlfriend and she knows when sth is right  :)

Have a nice weekend!
Marcin

I delayed writing this -I wanted to properly listen to 09zb before commenting (in the past I have got carried away to soon). First thing is with 09zb I can run the buffer at 384 instead of 768 in adaptive. Why is that Peter? my sound card buffers have not changed of course maybe better interaction  with them.

I checked 09zb versus 09y with the following settings:  adaptive & 1024 buffer.

I really thought with 09y I was there, that I had arrived & there could be no more to come from my system as the sound was so good. So when 9z appeared I thought nice window dressing but underneath same engine same SQ.

How wrong - I started to make comparisons but 9zb is much better than 9y in my system so no need for further comparison. It is the whole sound that is better - clearer more dynamic there is a layer of hash removed. Hash that was not there to remove!! (and I don't mean hash in the Dutch way ha ha!!!). So I then just listened and listened for hours to all sorts of music - because it is just sooooo easy to listen to. I can't say much more the sound is so beguiling I just want to listen and listen.

I am right now listening to some 24/96 files and the sound is crystal, sparkling clear - voices are there in the room, voices oh my!! how good they sound - you can hear the real expression fantastic. A big improvement but how can it be? - I thought it superb before.

I look at my speakers and wonder how bits of paper, magnets and wire can produce such sound. And then I think what will it sound like with a NOS1 and maybe some nice horns? Scary.

So where do we go from here? how can it possibly get better? - but I know that if Peter has anything to do ** it will ** I don't know how you do it.

Best regards from a very happy bunny

Paul

A big smile Paul. Thank you very much !

I agree with Paul and the others. 09zb is the very best so far. I run Adaptive and 1024, like most of the others. Reducing latency to 512 make the sound (dynamics) somewhat "overdamped". 1024 is just perfect. Never before could I play so LOUD. Unlimited LOUD! At the same time the sound is extremely detailed and natural, without any boomy bass or hard treble.

Congratulations Peter!
 :yahoo:


EDIT: Also the soundstage is phenomenal. I have never had such a wide and deep soundstage. The sound has never before been so detached from the speakers.

Peter, you know, I've found the culprit of that mysterious buffer problems. Windows changed power settings without a notice although my power plan was still set to performance! I had to change to power saver and then back to performance again. Stupid, but that did the trick.
I've been experimenting more with number of activated cores and discovered another mystery. With 2 cores enabled OS' performance is somewhat better - apps load faster and dpc latency is about 20% lower in comparison to 4 cores. I don't know whether it's a system issue or my platform's rather. Anyway, one core sounds the best and I am able to achieve the same buffer sizes regardless of sample rate used. I also added a second PSU for 4P socket and all the dirty components (hdd's). My primary (better) PSU supplies 24P socket only - very good improvement, but that's just the beggining of my pc-audio trip. Dedicated super efficient audio server with linear power supplies to come...

Marcin,

The "number of processors" should NOT be selected. (Thus Unticked)
If you selected 4 processors in the first place instead of unticked, yes, this slows things down.

Roy

Roy, I'm talking about "physical" number of cores running - in BIOS. This has nothing to do with XXHE or OS.

Yeah, I've now a new dilema: Whats the "correct" volume for the type of music playing? Rock, of course is easy, but classical is where I'm scratching my head. Hillary Hahn, Bach for Solo Violin, there is a point where I know that the volume is just so "unnaturally" loud, but it sounds so freakin' good at such high volume that I have to remind myself to turn it down to more natural levels, where it also sounds so freakin' good but different, not as revealing. Its like suddenly having an auditory microscope!

its all a bit dizzying over here....... :wacko:

Thanks Peter.

Yes, I have made som reflections about the same. My experience and taste, say that also classical and acoustic music can be played with higher volume "than normal", BECAUSE most instruments are recorded with mic's very close to the instrument.

Through our hi-fi systems we are listening to the recording from the perspective of the microphones (and how they are mixed). A few record companies use single stereo mics placed quite far away from the instruments. Then the "correct" playback volume is quite low.
But most of the time the mic's are very close to the instruments, and then I prefer to turn up the volume. I mean, if I was sitting on a chair 50cm away from Anne-Sophie Mutter, then the live volume would be high too.

Same goes for acoustic guitar. The pick-up/mic is very close. In my system I can play such recordings at +100dB without boom. It just sounds "big", like it would if I had the instrument just in front of my nose.

So this must be wrongly put, or ?
(I guess you meant to say this the other way around ... or ... not so good with the hiface, I think ... hahaha, I changed my own sentense three times now)

No this is for sure not wrong. But I deleted the reply because I like to listen to it a longer time. I hoped nobody has read it yet but you came before ;) The hiface is not bad but hf resolution is realy not as good. Parts might be not as good like in the hoerwege. the hoerwege is also not a dream and technically it has not this superior asynchronus mode but highs are better that is sure. It is not better in every aspect. I still have to listen more.

Ok, I can delete all the (by now) three posts if you want ...

If you like you can but it is not so important to me. I tried some different settings now. It is very interesing how the sound depends on it. With Hoerwege I used special mode scheme2. The hiface lacked the hf resolution as I said but when I switch to scheme3! and adaptive mode the hf resolution comes back. So the settigns depends very much on the device. Now I would say hifiace is better. It is more dynamic and seperates the instruments better with a more massive sound. The hoerwege has still a bit more "air" but in every other aspect it is not as good. So i still live on the same planet as you, puhhh :)