XXHighEnd

Hello all,

I installed xxhigh end last night on my for-audio computer, which is built using the cics cplay/cmp suggested cpu and mother board, along with many of the bios tweeks he suggests.  it is not connected to the internet, and is on an audiophile rack, and powered with a very good power cord.  i am using an essence stx pci-e sound card, with an outboard linear power supply feeding it 12v and 5v thru the molex connector.

i have been using it via analog outs, but recently have been trying a ps audio perfect wave dac, on native mode....

normally i use winamp with the steve monks ks plug in, or a recompiled otachan asio dll, by running winamp in cmp mode.

xxhighend sounds good so far, but I have a number of questions about configuring it, and what to look out for.  there is no provision in the stx driver to set buffers, only in the xx control panel.  should i aim for a high number or a low number?  when i click special mode, i get nothing but gritty noise.

i have not played with the q controls yet, but having 5 of them allows an enormous number of possible settings, and I am daunted by the task of trying them all......

i am going to stick with xp, so i would like to find out what xxhighend can really do before i decide to buy it.

i also have the prodigy hd-2 sound card (two of them, and advance and a standard- the standard is modded with better caps, better jacks, and with better opamps.

hope this was the right subforum to post in- I am not looking for someone to dictate what I should do, but to give me some guidelines as to how to "optimize" this software.

thanks for responding.

mark

Hi there Mark, and :welcome01: of course.

Well, I'm not quite sure it is your intention, but I'd concentrate on the PWD. But, and pardon my ignorance, how is this related to the pwx driver ? So, I just don't know about that. However, you say you have sound from the PWD, so let's continue with that (at least for now), and I assume your "native mode" runs through the stx, right ?

So, about those additional 4 Q controls, just leave them be, and set Q2/3/4/5 at 0. Q1 though is important;

As the release notes say, if you don't know the buffer size of the sound device (which in your case seems to be the stx), set Q1 to 0, and choose each Device Buffer Size from the largest (as put forward by XXHE) to the smallest, until you get a Too many Buffer Errors within a second. Next choose the before trialed buffer size, set Q1 to 1, and this is your lowest "latency" setting possile (Adaptive Mode that is !!). This lowest Q1 value (thus always 1) is supposed to sound the best, although it is completely up to you to find out the real merits for your system, soundcard and DAc (etc.). In fact, there's nothing more to it.

Allright, what you just did, is finding out the soundcard's buffer size, and in the end it comes down to "matching" the by you chosen buffer size with "your" Q1 value. Thus, for the stx this is one buffer size only (because it can't be set (??)), while for the Prodigy it may be settable - and now it is up to you to a. match the setting in the soundcard (buffer size) with the Q1 setting in XXHighEnd (which is simply done by entering the chosen soundcard buffer setting in XXHighEnd) for the best sound to your perception (thus : each soundcard buffer setting may sound different, but always denote that setting in XXHighEnd and use Q1 to compare) - and b. find the soundcard's buffer size that sounds best to you (and never forget to denote the chosen setting in XXHighEnd). Now, once you found the best buffer setting, you can fine tune that with Q1 (2, 3, 4, etc.)).
Here too, the best sounding option will be the smalles sound device buffer setting, BUT, it may be not usable for you (clicks, ticks). So (and again generally) choose the lowest setting which doesn't produce clicks and all.

I hope you could get this a bit, and that you understand that when no buffer size can be set for the sound device, you can always use (better : let XXHighEnd use) the one setting which applies to it. Even if you don't know what applies (see the procedure to find it, above and in the release notes).

If it is not clear it will be me at lousy explaining, so please ask further if you can't understand it.
Regards,
Peter

Thanks, Peter.  I only had a sort while to try it last evening, and it was tricky to set up, since the pc did not have the net frame stuff installed, so i had to d/l on another pc, and then move it over with a usb stick.  what i did notice was that if I slid the q1 control all the way down to the bottm, the playback either stopped, or the sound became completely unrecognizable - just nasty squeals,and a "grinding noise" like a horribly blown speaker.

i will try your suggestions tonight, and see what happens.  I can also put one of my prodigy cards into the computer to try it as well, although it is toslink only, and i have a philosophical preference for coax spdif.

i will let you know what i hear after i experiment a bit.

best regards,

Mark

Mark, to be sure you get what we're doing here currently :

With Engine#4 (Kernel Streaming) we are seeking the limits and crossing borders of what normally is possible. Thus, those around somewhat longer even expect sound to be distorted at first (to be sure to be over the limit), to next find the limit without distortion, that situation bringing the best sound. This may look as a strange operation to you, but when you grow with the "evolution", you will take it for granted, and will even be very happy to find your limit and your best sound.


I must emphasize that yesterday I forgot to tell you that my suggestions of operation spring from 0.9-y7, while you seem to be using y-6. That (Adaptive Mode) won't work there ! So, although Special Mode (which is what you will be talking about I assume) sure has its own sound, Adaptive Mode (in 0.9y-7) is a 100 times more easy to "calibrate". So, if you are new, it is my strong advise to start with that. But, as it seems now, it may even bring the best sound, so you could just stay at it.

Peter

Peter,

I tried your suggestions, and it worked out, once I figured out that Q1 ranges from -4 to +???   my monitor is not very good, and I thought the lowest I could go was 4, and could not for a while figure out how to get it set to 1. special mode did not work at all,

once i figured that out, I was able to get it to be stable at 128 samples, q1=1.  , i set q2 to it's mid-way setting, which balanced a nice natural warmth with good detail and accuracy.  I left q3,4,5 set to 0 so far.  it does take about 12 seconds to start playing a flac file (with lowest compression) from an internal sata drive, and there is a quiet pop noise or two just before the file starts to play, but there are no glitches or pops/ticks once the file is playing.

i did get a few errors, as i was experimenting- engine 4 did not start in expected time..... so i just closed xx and restarted it.  I still have a slight sonic preference for winamp running thru the steve monks ks plug-in, so my goal is to beat that sound with xxhighend.  Are you suggesting that version 7 will work in xp, and sound better than version 6?  Also, what settings only available in the paid version will improve the sound, and is there a way to pay for the full version, and get a refund if it does not work out?

thanks for taking the time to read this and make suggestions.

Mark

Well, the only thing I really can say (again) is that you should use y-7 because of its Adaptive Mode. I'll bet you 72 euros that will make "the" difference. haha.

The only real thing of importance in this case that's not in the Demo version is the Core Appointment Schemes (which allow you to find cpu schemes with better SQ). Sadly (and this was unintended) by now, for Kernel Streaming, the Core Appointment settings are important to achieve the ultra low latency. But, if you don't have a really underpowered processor, it won't matter much. So, including this unintended combination, still no refunds. :)

But please, try y-7 and Adaptive Mode. Let's not bet then, but let me promise you it will get better from it. Ok ?
Peter

am trying 7 now, in adaptive mode,  had to go with 256 samples, with q1=1. sounded a bit dry and edgy, so i am up to q1=12. it is sounding better.  the main issue i have, after brief listening, is that the soundstage is wide, but very shallow - there is less layering of depth, than with my winamp configuration.  i cannot find a data file to report on, but cpu usage is at 0% most of the time, and goes up to 3-4% for a short time -- when loading a large flac file it gets as high as 50%.

also, it stops playing after a file from the playlist ends, but i would like to play the whole list in order.  also, when it has an album with 10 or more tracks, it puts track 10 before track 01.

confusing.

Hi Mark,

It looks to me (not sure though) that you are using XXHihgEnd in Attended mode. Wrong ! For the best sound quality you should use Unattended mode (see checkbox at the bottom). But don't forget to setup autoHotkey first, or leave the Services On (which is again bad for good sound).

The fact that playback stops after a track (which to me indicates your Attended mode) should be coming from an underpowered cpu; can that be the case ? Anyway, please use Unattended, and you will be surprised.

The tracks not sorting correctly will be because of the track numbers in there not being 01, 02, 03, but 1, 2, 3 (and I know a(n official) download site which presents them like that). Notice that this automatically implies a wrong alphabetical sorting sequence, like 1, 10, 2, 3.
This is solved when you use cue sheets, or -for example- when the FLAC files contain the normal numbering. So all 'n all I guess you have a special situation at hand (because I really never see complaints about it), with the missing cue sheet as the most obvious. Notice that the cue sheet (.cue) must be in the same folder as the tracks are. Note : If you recognize that you are just doing this, than don't bother much, but tell me you do, and I will look into it. I mean, I can be doing things wrong, and I estimate this is a not much used solution (so, not really tested, so to say).
In any case, renaming the track numbers to 01, 02, 03 etc. will solve it.

Let me know what happens with Unattended !
Peter

i will try unattended tonight--- the flac files are numbered with a leading 0, as in 01 02....   the config i use in dbpower amp is to name the directory artist-album, and the flac files are track number track title.  winamp sorts them correctly, but i will check the tags and see if something is up there.

if i use cue files, which i can do very easily, do i add the cue file to the playlist, and let it find the tracks?

also, the cpu is an intel e7200, using 1 gig of kingston ram, and with the bios tweeks suggested by cics, which means it is underclocked/undervolted.....   I had been using winamp within the cics cmp environment, but since i have switched over to using an external dac, i cannot get the xonar stx card's drivers to use asio via the coax digital output (at least in winamp).  i switched over to using the steve monks ks plugin, and the digital out works flawlessly, and allows me to compare xx to winamp both via ks (so, apples to apples).

I may try to use another sound card, but am inclined to stick to the xonar stx, since it is pci-e, and i have built a nice dual linear psu for it, supplying 12v and 5v, with choke-filtered supplies.  this was a nice improvement via analog outs, compared to powering the card's molex connector via the computer's psu.  i have also done some capacitor bypasses on the dual 12v lines for the p4 processor power feeds from the computer's main psu.

also, the dac i am using (cary xciter dac), internally upsamples to 32/192 (cannot be turned off) - if this gives any ideas as to which settings for the q's may be good to try.

thanks again for the suggestions....

Mark

Hmm ... can you show me a screenshot of the Playlist Area with the wrong sort order ? BUT :
Notice that you must test this with selecting an album in the Library Area (just click on the Coverart) and next click the Load button.

Please keep in mind : if you don't use cue files, it is not my intention to start using them. If only the track numbers are in the track name allright. So, something else must be wrong then, and I rather solve *that*. Ok ?


... maybe you were just explaing what you are using (for sound card and all), but nobody is telling you to change ... (or I have missed it :))

If you feed the DAC with 16/44.1 theoretically you can go lower with Q1 (Kernel Streaming Special mode) than with 32/176.4, but if I were you I'd still try Arc Prediction (Quad). Or ... or would your DAC resample that to 192 also ? (brrrr)


Ok, one really important thing : you have too few memory. You really must have 2GB ... I am sorry ! ...

Peter

i will try those ideas.  i have been choosing music using the explore button --

i also have 2 2gig ram sticks somewhere, but at one point i listened to the corsair 2g ram vs the 1g kingston, and thought the kingston sounded better, and left it in. 

Hello all, hello Peter. My first post.
Have built dedicated PC similar to fzman (E7400/mATX mobo/1&2GB RAM) - rest of the system in signature.
Wanted to try XXHE with WinXP SP2 - no success. Getting error message:
"The application failed to initialize properly (0xc0000135). Click OK to terminate the application"
This happens with 2 different mobo's (although very similar, G31 chipset, integrated graphics).
On my other machine (XP SP2 as well, non integrated graphics) XXHE starts w/out complaints.

Any clue?

Thks, George

PS: XXHE -y6 & -y7

Hi George, :welcome: here !

Can this be because you skipped the dot-net install stuff (needed for XP) ? (notice I have no indications for this, but is *is* something XP needs).

Although the Demo version, Install notes and getting an Activation Key  (http://www.phasure.com/index.php?topic=11.msg14#msg14) is somewhat old maybe, I guess for XP the majority in there is still true.

Can that be it ?
Regards,
Peter

Yes, and Googling helps as well... :blush1:

Thank you!

Peter,

I am running with unattended mode now - I did not notice that the settings tab had a scroll bar, and that there were a number of settings I did not see before.  I am just starting to figure out how to add to the playlist from the library tab.  (is there a link with explicit instructions so I am sure I am doing it corrctly, rather than guessing and trial and error?) Things are a bit slow, but I have not yet put the 2 gig corsair ram in yet, but that is easy enough to do.  I noticed also that autohotkeys crashes winamp, so I have to remember to unload and reload it as needed.

I have also turned q2-q5 down to 0, and have backed q1 down to 4, adaptive mode, with 512 for the buffer....  the sound is getting better, and I am closing in on deciding whether to buy or not.  I switch back to winamp to have a comparison, and because I can run it continuously as I am burning in new dac and cables.

I also switched to unattended mode, and that does seem to sound better as well.

Thank you for all your help- I must say that the feedback and assistance are very valuable, and that you are earning your money.

Best,

Mark

Hi Mark, thank you for your nice words.

When you set a proper Music Root folder (Settings Area), your albums will show up in the Lirbrary Area. Watch the textbox at the bottom (which should show that music root foler), and click search. A maximum of 100 albums will show up there because of the demo version. If you have coverart in your per album music folders, the Library Area will show that Coverart (instead of backish crosses). It is there where you can click an album for selection and do many things with it (or all the selected albums as a matter of fact). One of them is that Load button I talked about. Much more you will find under the rightclick context menu (which thus shows at rightclicking an album).

Notice that there are more means to load an album (or individual tracks) in the Playlist Area, and I only pushed you towards the Load button because of that strange sequence you have, which easily can come from the way you e.g. drag files from Explorer onto the Playlist Area (which is just one of those other means).

In any case, take your time for it all, and just try to get your best sound. I think it could take you days to only find that, but with some little experience you will perceive something better compared to what you are used to soon. You must also learn what to listen for, and without someone "teaching" you, this takes some time also.

So far for now,
Peter

Peter,
Thanks for the tips.  I have been involved in high end audio for a long time, am a trained musician, and sell high end audio for a living, so i do have a lot of listening experience.  The system I am using is very good, and I have attended to many details of making the computer sound good - plugged into a shunyata v-ray with shunyata power cords, magnan interconnect from dac to sim audio preamp, audioquest digital cable, power supply tweeks in the pc, along with the custom diy linear power supply running the sound card.  The room is acoustically treated, and it is sounding very good indeed.

The computer I am using for audio is not connected to the internet, so has no antivirus running, and has very little software installed on it.  If i buy the software, how would i transfer the "license" to that computer?  would i have to install the xxhighend from scratch, or just copy a file over on a flash drive?

I have .9y6 on this computer, can i pay for that one, and then transfer the payment over to the audio pc running 9v7?

thanks again,

Mark

Hi Mark,

XXHighEnd doesn't "install" a s such, so it is only a matter of copying to the folder where you want it.
The license is for your "household", so you can have it on several computers (at the same time). For PCs not connected to the network there's a special procedure which I will PM you.


Versions really don't matter. You can also always fall back on old versions (which keep on being available for download as you may have seen).

I'll send you that PM.
Peter

Peter, I am still using the 1 gig kingston ram chip, and it seems to be playing fine, and responds more quickly with adaptive mode, and unattended.  is there any sonic advantage to switching to a single 2 gig corsair ram module? 

Mark

I suppose you achive this by setting your Split File size parameter relatively low. This works allright, but it not the intention of it all. But if you are satisfied with the sound like it is, why not (and I sure don't guarantee you will be perceiving changes to SQ when the parameter is set larger).

Notice that I am not familiar with low voltage, underclocked etc. memory and everything, which is a thing I leave for cics. :)
The same for 1 vs 2 RAM modules. So, to respond to that question : no idea !

Peter


PS: By now I am not sure what your settings are (you will have told it somewhere, but please put it in your signature !), but supposed you are not using upsampling (or Double only) then this really matter for the memory usage !

I will add to my signature.  Last night I was able to audition an Audioquest Signature digital cable, with the Boccino rcas.  Unbelieveable how good it is sounding.  The speakers dissapear, tonality is wonderful, very dynamic, and you can hear the spaces between the musicians!

I am liking the software very much, but can;t figure out what all the commands are for the autohotkeys- so far i have found alt-x to start xxhighend, alt-e to exit, and alt-n for next track - i cannot find the command for previous track... 

also, when i set q1 back to 1 (down from 4) the hard drive light came on, and stayed on for a long time even while music had been playing for minutes...  so i put it back to 4.

I can't imagine how that would be related ...
Maybe you just try it again ?

Notice that changing Q1 will restart the audio engine. So *that* implies other stuff. But only then, and it should fade out.

I am still learning, and only have a short time each evening to experiment and listen, and I would much rather listen, as the sound is very captivating now - and i need to do more system tweeking before I do a lot more experimenting with the settings.  But I will keep you posted.

Peter,

I switched another setting.  I changed my dac is   from 16/44.1  to 24/192.  i am using the Cary xciter dac, which uses the akm 4399 dac chip.  I think a very slight nasal quality in vocals has gone away with this change - does that make sense?  i may tri 32/192, which I believe is the limit for the dac, i will check the spec sheet tomorrow - gonna listen more before bed.

Allow me to guide your fingers on the keyboard.

( see below )

http://www.phasure.com/index.php?topic=1058.msg8923#msg8923

Hi Mark,

If your DAC is 24/192, you just *have* to tell that via the settings. So, even if you are palying normal CDs only (redbook) the additional available bits will be used, although it depends on what you are doing. E.g. using the digital volume of XXHighEnd is enough. Or an upsampling means.
Btw, I don't think the DAC will be 32/192, but I didn't look it up.

Peter

From the AKm press release:
High Performance 32-bit Stereo DAC Optimized for Sound Quality
San Jose, CA December 19, 2008 — AKM Semiconductor (AKM), the world’s premier
supplier of audio ICs for professional and consumer markets, today launched the AK4399, a
high performance 32-bit stereo DAC that achieves 123dB dynamic range for stereo and 126db
for mono outputs. It is the newest member of the Audio4proTM family, and it is the highest
performing device in the 32-bit DAC series. An innovative design technique utilizing a
symmetrical layout for the left and right channels prevents signal quality deterioration, and a
32-bit digital operation block provides full 32-bit processing.
Achieving very high dynamic performance, the AK4399 features -105dB THD+N and 123dB
dynamic range in stereo mode. The device features fully differential outputs, digital attenuator,
de-emphasis filter, soft mute circuit and zero detection. The sampling rate ranges from 30kHz
to 216kHz, and the operating voltage is 5V.
Two digital filters are integrated into the AK4399: a sharp roll-off filter for high fidelity audio
and a short delay filter that has no impulse noise pre-ring for live performance applications. In
addition, 32-bit digital audio data can be input directly to the delta-sigma modulator without
passing into the integrated digital filters, allowing for use of an external digital filter.
The AK4399 supports DSD (Direct Stream Digital) or PCM data input. Since the AK4399
contains an integrated low-pass filter only for DSD data, the required performance in the
SACD format book (Scarlet Book) can be easily realized with a simple external analog filter.
The output level is the same for either PCM or DSD input data, as well as external digital filter
input. Volume control is independent of the type of data input.

thanks Calibrator!  that's a big help.

Peter,  regarding ram, what I was asking is whether a single 2 gig ram module would give better performance than a single 1 gig ram module....

Not sure if you have any opinion on whether it would sound any better, but whether the system would respond more quickly, run smoother, allow lower latencies,  etc.


Also, how does q1 and the buffers setting interact with the dual and quad sampling options (sonics, mostly)?  are there specific recommendations, using ks/xp yr7 as in my signature?

Wish I had more time to experiment myself, but would like some guidance so I am not randomly trying every possible permutation!

Best,

Mark

Peter, et al,

I have done some brief listening with linear, arc prediction, and anti-image at both 2x and 4x settings.  I have had mixed results, and noticed that the driver for my essence stx soundcard has fixed sample rates of only 44, 88, 96, and 192, which matches the indicator lights on my cary xciter dac.  with any of the upsampling settings, the sample rate light on the dac does not light. Any thoughts on whether this is good or bad?  How do you think it will effect the sound?

I switched back to straight 44.1, and the lock light came back on, and the sound is a bit warmer, and there is more precise imaging - no "wandering"

Thanks for reading this and offering any thoughts.

Mark

Hi Mark - So I *should* have looked after those specs. In that case, set DAC Is to 32/192 of course !

But now your sample rate light ... I don't know ...
Btw, are you sure your mentioned list is correct ? I mean, 48 should be there too, and when the DAC does 88.2 it should do 176.4 as well ? and ... aren't you playing Quattro ? so ... ?

Anyway, it would be interesting to know whether you receive the same result (light doesn't come up) with Engine#3 ...


I can imagine this is a tough job when you just start out with it all. But do you follow the forum topics ? I mean, so much has been said about this the past few weeks ... I can't do better right now than referring to that ...
In any case I wouldn't be bothered about the mixture of upsampling on one side and Q1 settings on the other. First choose whether you like upsampling or not, and next play with Q1. Not the other way around, and not both at the same time. Upsampling is more important, and if I were you 'd only try Arc Prediction. If you don't like that (or better : when your DAC doesn't like that), just don't. It is nog needed, and the fact that 98% of people like Arc Prediction doesn't mean something is wrong or whatever when you don't. It *is* DAC dependent you know (and BTW made for filterless DACs, which your DAC is not (again I didn't look it up, haha).


Isn't this a typo somewhere ? -105dB isn't all that good. Actually, it is quite bad ... :yes: (just a technical thing, you won't perceive anyway IMO).

Peter

Peter,

Thanks.  I may have mis-typed.  it is 44, 48 96 ans 192, on both the sound card and the dac.

I can't do engine 3 since i am only running xp, not vista or 7- unless I am wrong about engine 3 on xp.

I do not have a problem with whether the light comes on or not on the dac, what i was wondering was whether the fixed rates, and the lack of 88 and 176 on the sound card meant that dual and quad upsampling were then being resampled in the sound card driver, and that the driver may be the bad guy!

I ordered a 2 gig stick of ram (kingston valueram) which should be here today, and I will install it.  As I type this, there are 2-second pauses in playback as xx is loading a large playlist, with constant hard drive activity, and the xx screen not switching over to the wallpaper in unattended.

Do you think the extra ram (2 gig vs 1 gig) will help this out?

Best,

Mark

Mark,

Aha, so it is the other way around ! Now I understand ...
So indeed, when no light for locking to 88.2 or 176.4 is there (and 96 and 192 also don't lit) plus your description of the "fixed sample rates", we must assume that your sound card is resampling. Not good !
In this case you might try the AI Upsampling, and be sure to set the "96/192 Only" checkbox in the Settings Area. It might become better from it, and in this case 96/192 should lock (lit). Notice though that the AI upsampling is rather time consuming, plus your 1GB from this moment may play a certain role in it (more slow).


Did you install the 0.9y-8a patch for this ? (Re: No Wallpaper at first track in 09y_7 (http://www.phasure.com/index.php?topic=1137.msg10163#msg10163))


For some (systems) this is unavoidable, and certainly for the ultra low latency Kernel Streaming modes (Special and Adapative) it is. In that case it may be better to untick "Start Engine3 during conversion" (Settings Area, Memory and Disk utilization section), of which I now realise this should be named "Engine3/4 ..." :). Then playback won't start before all is loaded/converted. More memory will make a difference to the aspect of things going (somewhat) faster. Ah, you will see it yourself soon ...

Peter

the 2 gig ram just showed up on my desk, courtesy of ups - i will let you know how it works out.

Should I try special mode instead of adaptive?  any thoughts what sonic differences there may be?  special did not work at all in 9yr6, but that may have been a too little ram problem?

Mark

Hi again Mark,

Special Mode is tedious to get going (to say the least), so I would keep it to Adaptive. It think is *is* so though, that Special Mode can just give that special experience (I am going to try it myself again in an hour or so). But again, better use Adaptive Mode, because it is easy to get going, if you only follow the guide (the Tooltip on the KS Mode combobox).

No, I don't think more memory will make a difference in getting Special Mode going. BUT, if you have tried it before at starting a larger playlist while that checkbox "Start Engine3 during conversion" was ticked, your chances severely degrade (as does trying it in Attended Mode). It merely depends on the soundcard (and currently I do NOT have the experience of you all to really know about what does and what does not) and above all the cpu capacity and the spread of the load (for example : a one core processor will be a no-go).

I understand that you want to be as prepared as possible, but it really is difficult for me to give you the good advises. If you (hopefully) start perceiving the differences, it will be the most easy to proceed from there. Also it will be far more easy to help you. Now we're both a bit in the blind, because so much depends on your PC. But I promise you you will get around with it.
In any case keep asking, but don't be too disappointed about the answers !

Peter

And Mark, to get you on the track a bit (if you don't know already) : very generally in each occasion (Engine#3 and #4) the lower the Q1 the better the perceived sound. I dare say this counts for most of us (for me too), but of course keep in mind : until it breaks. So, assumed the lowest setting indeed *is* the best setting, you must always find a settings which works, and then go lower (for KS Adaptive it works a bit of the other way around because of the guide). Notice though, and now with Engine#3 as the example, that you must be careful not to be fooled by the lower Q1 setting. For example, for me (and I think for most) it is clearly audible that at the lowest Q1 for Engine#3 (which btw is not about latency at all there) is giving you a more accurate sound. But, this turns out to be fake somehow when hopping over from e.g. Jazz to Classical, and the violins appear to sound digital. I am talking about Q1 near 0 (or even under it) and nearly everybody perceives this the same. I didn't notice this with the low latency settings from Engine#4, and there it just seems to be the lower the better. That is, this too wasn't reported otherwise afaik, BUT, it can break (Q1 of Engine#3 generally won't).

I hope this helps you !
Peter

I put the 2 gig module in, and all is working fine. I switched from no appointment to scheme 1, and I think it sounds a little richer and more natural, but with a long play list the hard drive light stays on after playback begins, and stays on for at least 5 minutes, and it takes a while for the wallpaper to come up.

is it better to have a short playlist, for sound quality?  I have many albums in the play list so it will run for a long time so I can break in my new cables, etc..... 

also, when i try to save a playlist, i only have two choices-  currently playing, and played, but i have not set up a folder for data, and wonder if that is required to be able to name and save playlists......


lots more to learn

Mark,

About saving the Playlists : It is better to have a separata folder for the data outside your XX folder (also meaning not as a subfolder). You'll notice this with several XX installs. But look into Galleries. They can behave the same as Playlists but are a 100 times more powerful.

About the disk light ... not good. I could say your Playlist is too long (or the selection in there), but we would be talking about 100s of tracks, and this is useless (you won't listen to that in one go). But I don't think you talk about 100s ...

Did you untick that "Engine3 Start duing Conversion" checkbox ? (I think I "told" you before).

If so, you may have tweaked your PC in the wrong direction, so it now behaves as a laptop (I'm serious here). Disk light should go off within 5 seconds or so after playback has started (but mind beforementioned checkbox), and if not, your PC can't cope with the processED data. This is not easy to explain, but what it comes down to is that e.g. data to be copied to disk first goes to memory (this is always so) and from there goes to disk. Now, it can easily be so that this last part takes 5 minutes (look at a laptop, and there it will be standard for most), while it should be instantly. If not the latter, all kind of things can be too slow. Can be the disk, can be the memory, can be the 2nd level cache, can be the FSB, can be the processor, can be the disk controller.

?

Peter

Peter,

I checked my power management settings, and it is set to 'always on', and only monitor is set to turn off, after 20 minutes....  I did lower the color depth from 32 to 16 bit -- I would have gone lower except there was no setting for 8 bit at the screen size i am using.

this helped the drive light activity very much. 

I also set up an xxdata directory, and put it in settings, and set the gallery and data to point to it- that works as well.

I get lots of errors when I pause a playlist, and it keeps wanting to send error reports, which i do not send, as this computer is not networked.  i have to kill the xxhighend process in task manager to get it to stop giving errors, then all goes fine.

Now that I know how to save playlists, I can chose single albums for real listening, and load the large one when I want to get hours of burn-in on the system, without being there to listen.

I will start playing with the gallery soon as well, to see what it can do.

Thanks,

Mark

The player seems to stop part way thru the playlist - I am not sure when exactly, we left for some errands, and when we came home 2-3 hours later, the playback had stopped, and xxhighend had closed.

I have the settings in xp set for "always on" and I switched to giving priority to background vs programs -- I will see how that works.

Also, I click on the "x' to close the player window while it is playing, but still processing the playlist - this brings up the wallpaper screen well, but should I wait and let the program switch displays on its own?

thanks,

Mark

Hi Mark,

Please stop with these playlists before you are confident with what you are doing. Or, the least thing what happens is that I don't understand.

This :


and this :


is not consistent IMO. You either play Attended (seen checkbox at the bottom) and the GUI stays, or you play Unattended and the GUI disappears automatically (!) after the tracks in the Playlist Area have been prepared for playback.
Also, the first quote above makes me tell you again that you should UNtick the checkbox "Start Engine3 during Conversion". Why don't you do this after I told it a couple of times ? Of course it is not necessary, but then please leave out descriptions related to a problem (solving) because I really can't follow anymore what you are doing.

The same for the playlists. Not "tomorrow" as you say ... stay out of them because I advice you to. Not because they cannot or should not be used, but they imply you get stuck.
Just play albums for now. One at a time. When you get and see what it takes to do that, it will be your decision whether it is "allowed" to load more. IOW don't load dozens of tracks in the Playlist Area (!!) before it is clear to you what will happen with that, and which is very different for Attended vs Unattended. Ok ?

Here is a small list for the fastest means of loading tracks (this is per track, but for Unattended this is for all the tracks in the Playlist Area) :

- Do not use Anti Imaging Upsampling;
- Do not use the Normalized Volume;
- Do not Copy to XX Drive;
- Use WAVs (this is a stupid one of course, but it does matter);
- Load one album in the Playlist Area (for now);
- Do not let spin down your disks after 1 minute (use 4 minutes or more).

From here on you can start to change either of the above, but watch what the differences are, and what they mean for your real life.

Hope it helps ...
Peter

Peter,

Thanks for answering - I am sorry if I have been stubborn, but I am trying to learn the program with very little time to play with it- and it sounds so good, that I want to be able to listen to music, rather than just tweek settings.

When I first got xxhighend registered, which was probably yr6, it would run for several days without interruption.  I should have known better to expect that it would work with huge playlists-

I agree with your advice- use it to play albums - I would have used winamp to do burn-in duties but it would crash every time, since xxhigh end had been installed and run.  Turns out it was autohotkeys causing the conflict, and some autoruns had gotten into my registry.  I cleaned everything out, and know how to run winamp when I need many hours of continuous play for burning in cables, etc.

I also played with some of the optimiazations of windows, and found that xxhighend seems very sensitive to video setings, so I will be careful with them.  Many of my assumptions going in are from the cics/cmp/cplay mentality which I had been following, and xxhighend seems to take a different approach to some extent, so I will be more careful and attentive about making changes and asking for help.

Now that it is working well with single albums, or playlists of 2-3 albums, I will try to install and configure yr8, as the wallpaper still sometimes skips a track, although playback continues.

Sorry to have interrupted your weekend.

Mark

Hey Mark,

By no means you interrupted anything of my weekend. I am just afraid that maybe you won't get there because of running out of patience. Please don't. I realize though that the tone of my last post may have been "impatience" by itself. If so, please appreciate it as unjustified.
Just hang on, and I know you are trying the buts out of you. I will keep on trying the same !

Thank you ...
Peter

Peter,

It's been a few days since I posted.  I am now using xxhighend just to do serious listening or enjoyment listening (not always the same thing!).  I am still getting my system and room dialed in, as I have moved into the house somewhat recently.

Anyway, y7r is working fine, and sounds really good.  I am happy.

now i need to get a registration code for the main computer which is attached to the internet (i registered to unconnected one by a temporary hookup).  Can you pm me and let me know how to proceed.  i am at work now and will not be home for about 5 more hours.

anyway, thanks for all of your help.  I am very happy with the sound.

Mark

Oops, I missed your post yesterday. But you have that PM now ...
:sorry:

got it.  thanks!