Haha, you guys talk about version a,b,c,d,i without mentioning whether it's about 0.7 or 0.8, 0.9 etc. That's why.

The changed slider takes effect when a new .wav would be loaded. Or when (stop and) play is pressed (or Next, Prev).
FYI : A next track is nothing to this respect for a big cue pile. But you know that I guess ...

Hi John,

Actually ... no. A combination of things caused this :

1. Edward, the only one needing it at the time, reported he moved to Vista 32. This would not have stopped me, weren't it that
2. I have a licence to download anything with is more or less in beta phase amongst which Vista, but at the time I wanted to download it it was not available anymore.
3. Once I knew were to get it from, my normal Vista development PC crashed from a lacking driver which was unrepearable, and I had to use the -then- free (Vista) PC to swap OS disks and restore from the original PC all the bits and pieces of over 1 week of downloading it originally took.

Ad 3.
The supplier website of the backup software (Adonis) needed for Vista did not "allow" me to pay for the version I needed. So, no backup.
In the end this caused the need of the original PC (dual booted under XP now) with data to be online for several weeks, and actually 2 weeks ago I could get rid of it.

Is this all a good explanation ? probably not. But sometimes even I can get fedup with stupid computers, and again, afaiknew nobody was waiting for it at the moment.
Right now ? ... I don't even have the computer anymore. And besides all, I think chances are about zero that it even *can* work. It's hard to Google for whatever it is it lacks (Vista64) but the gaming environment seems to have enough problems with it.
Currently I put my money on SP1 and hope it solves "my" problem.

Peter

http://www.phasure.com/index.php?topic=67.msg827#msg827 (starts here : http://www.phasure.com/index.php?topic=67.msg823#msg823 )

That's the same thing ... 

No, it won't be bad for your speakers. Also (if at all) it will happen only with tracks that contain "full volume". If the recording is right, this won't be in there are all, and if all is "maximized" without clipping the data (!!) there would be one small "crackle" (IMO hardly audible) throughout the track. However ... with bad "recordings" (digital masters), the sound can be clippen for even a second long, and that several times throughout the track. It still won't damage the speakers, but it is just not good for listening obviously.

I will do some better testing myself now, and let know whether it's really wrong or not.
Peter

Gerner, start reading here : http://www.phasure.com/index.php?topic=131.msg664#msg664

99% sure your problem is caused by the same.
Hasta la Vista,
Peter

Also ... more than 16 bits currently is not supported generally, but I did not even test it;
Might you feed XX with more and it seems to work, then with Inverted Absolute Phase it for sure will go wrong (at the peak levels).

All,

I think it is confirmed that when Inverting the Absolute Phase things can go wrong badly.
I had this myself, and cracks can be very loud and can be persistent. It occurs at the peaks of sound (full digital volume level).

I thought I solved it, but appatantly it can behave differently per track.
Save your speakers and do not use it !

If all is ok ... fixed in 0.9i-1
(but mind Vista and just not allowing it ! like with me ... )

A few bugfixes opposed to 0.9i :

• RealTime priority could not be selected for Engine#3.
  Note : It seems that not all Vista systems support this properly. Check Taskmanager ! (the program reports it works, but it really is not with some (?) Vista systems).
• Player Priority could not be selected.
• Changed wrong wording "Highest" and "Lowest" into High and Low for priority settings.
• Possible error message at startup about an illegal 0 value.
  A new value 0 was added to the Q1 Parameter because of this (actually works one tad lower than 1 which was lowest before).
• Built in messages for presumed faulty Absolute Phase twist calculation (causing loud cracks during playback). If you have the loud (very brief) cracks at Inverted Phase, please check whether they are there with not Inverted Phase. If so, please report.
  Edit : ... which does not help !? : http://www.phasure.com/index.php?board=2.0
• It was intended that Vista would start with Engine#3 as default when not selected earlier otherwise. This did not work (checking this is a tough job, so I hope it now works).

It is clear now that the current boundaries the Q1 parameter provide, can exceed peoples' systems. It depends on the DAC and theoretically the CPU plus the combination of both.

As a reference : I use a TwinDAC+ with in between a Fireface800. How the latter exactly is related is hard to tell, and it is even possible that that one is determining for my system; my buffer settings for the Fireface are 128 currently, and it is a leftover of the 0.9h version, where I found this better sounding than the 48 it can be set to ultimately, with further theory that for 44K1 it can technically handle that (another RME user (DSP Hammerfall) pointed out that 24 still will do).
As a further reference my Q1 settings would be able to play without cracks and pops (skips) between a virtual -30 and +45. From this we could learn that for those who find one of the current boundaries (1 resp. 30) the best, let them speak, so I can extend the range.

When going towards the boundaries, both can have impact on the CPU. For me this is just visible, for others ... it may choke. I use a Core2Duo 2.4Ghz (E6600), and really one core is used by XXHighEnd.

With these more or less CPU constraints, the remainder is related to the DAC.

Because each DAC is subject to jitter already within itself (and the electronics around it), it is hard to predict the influence of the Q1 setting.
Elsewhere I said "the lower the better" but this is not necessarily true and it would depend on the amount of jitter the DAC inheritantly has.
Also, when the DAC and its electronic surroundings contain too much jitter I don't think the Q1 parameter can get it right all the way. It still would influence but I can't tell how and it would be different per DAC.

When you start to hear clicks and pops it means that the data is overridden because of bad timings or it is skipped because of choking. The latter seems clear because of CPU impeeded, but this already is not necessarily true. All is one major interaction of things and in the end it comes down to, say, computer technology. According the first, bad timings, I won't even start explaining. But for you, in order to visualize things a bit, suppose the data is fed too late to the DAC (which would be when it consumed the previous bunch of data) you'd have a skip. A silence.

With the latter in mind XXHighEnd does its thing and it does it under influence of Q1. Although the stepping of the parameter is kind of linear it is (explicitly) not said that in one direction you'd receive skips. It can happen in the other direction just the same.
It would be true though that the slider "adds more" in one direction and "has less" in the other. That's what I meant with "kind of linear".

Functionally you can't do more with it than *knowing* that once you cross a for you technically working boundary, you won't find another working area above that (when going upwards) or down that (whereever it stopped working).
With "technically working" I mean : without clicks and pops.

Within the working range you should be able to influence jitter.
For me the working range seems relatively large, for others it may be very small.
If you can tell the CPU is choking (which I cannot imagine, but anyway) then things obviously can be solved by a better grade CPU.
If it is not the CPU there is no advise; I could tell you to buy a DAC with more headroom on this but it would be a nono advise already because each has different reasons for a DAC. One sounds better than the other to your own judgement, and who says that it's quality can be influenced for the better by Q1 anyway (yeah, actually I say this, but it would be dangerous to take it as a guarantee as of this moment).

Also watch this line : In a few months time we will have different insights again.

Peter

TorH, this is normal;
If you were using 0.9d satisfactory you must set it to 14 and work around that figure until it collapses.
If you *also* worked with 0.9h satisfactory (without the dropouts) your figures are between 14 and 24. Mind the "*also*". so 0.9h was 24.

I can't tell what is the bandwidth of your DDDac and whether you can achieve the same jitter signature as e.g. me. Anyway try to find the boundaries, and obviously only experiment for better sound in between those boundaries.

Hope it helps,
Peter

Hi Bert, thanks.

What about your Player Priority ? can you change that effectively ? (from within XX).

I can't be sure because things won't be the same at your place, but possibly (even probably) your "hazy" is the refined fragility I was referring to in the other topic. It needs the experience *OR* special music to adopt it. If you have Roger Waters - Amused to Death ... try that. It contains loads of "information" hence detail at the micro level, to easily find out whether you deal with hazyness or fragility.

Both look alike when compared with too fresh sound because of added noise (that's what technicians would say). The difference would be in the level of detail. Fine detail. Not profound, but fine. "Planned" if you like. Air in between the squary waves of a synth. Air in between the distorted waves of a distortion guitar (not a feedback).

IMO (so far) the lower the better, until things collapse. 

You'd be having premium pirority then.
Djeez, what is this.

When I try to change to realtime (no matter what program) Vista tells me it is not allowed. "Set to High instead".

Since you can do it anyway, can you conveniently press Next or Stop ? I mean, does it respond correctly when running at realtime (XXEngine3.exe) ? Because if not, it's worthless anyway.

Sidenote : I had it all build in, and had to remove it because it just did not work. Here ...

Hi Gerner,

So indeed we posted separately and as it appears both our Q1's ended at 7. Hahaha. But maybe there is a difference how we both got there.

I got to 4 (and did not try lower) in order to remove the destortion from the saxophone ( http://www.phasure.com/index.php?topic=160.0 ).
But I found that not all recording did well at 4. This was day before yesterday.

Yesterday I started all over beginning at 24 and inverted phase, stepping to 14 with normal phase, and via 11 and 9 I landed at 7. Mind you, this is just challenging until you (I) think it is enough (and I think with the sax I jumped from 10 to 4 so maybe it can bear 7 as well).
So 7 is enough for me, and throughout I did not hear any of the so well known characteristic "distortion" anywhere. Mind you, "characteristic" had become "distortion". Times goes fast ...

What I cetainly do NOT do, is finding a better setting for poor material. I can't be sure, but I think that would be faking things. Filtering maybe. I'd rather go look for another amp or whatever it takes to represent the material accordingly, better. But again, I am not sure.

That nobody will land on the same value seems logic, while it is jitter impeeded, and jitter is different for everyone. Like you said, it depends on the DAC (too).

Gerner, it is amazing how you express about the "trapped" the waves in the horn, because I actually posted later than you did. Okay, since you are buildig speakers for ages you must know better than me anyway, but/and the other way around : this is how I felt it. So let it be true.

I feel like coming back to this one again :


What would happen, is kind of the opposite from eliminated unnatural squares : adding more. When more (unnatrual) squares are added to squares, you'll get sines again as long as we see that the derived squares are smaller than the originals. This will happen only (I think) because of bouncing or even reverberation of the horn wall. Another explanation I have not at hand (the digital data would not devide by half because of jitter (see my other post and the 50-50-50-50-50 becoming 50-0-100-0-100 ... it can't become 25-25-25-25-25 because of jitter).

Which brings me to the rather interesting next subject, hehe :
(just made this up)
I was wondering why Doubling isn't better anymore. Well, this is completely logic to me now :
Doubling was meant to be a form of jitter reducer, with Upsampling as a derival of it. And Upsampling (I think) sounds worse now. Ok ...
Upsampling is not nature; it's faked nature (calculated resolution, whatever). With upsampling indeed the 50-50-50-50-50 could become 25-25-25-25-25, really ending up as 25-50-0-50-0 because of jitter again (which always comes behind all). Thus, when we would be capable of expressing 50-50-50-50-50 indeed, for sure we should not turn that into 25-25-25-25-25 (assuming that is merely without jitter as well). But what would it do ?

It indeed would round squares. But it would round natural squares this time !! (again assuming we merely eliminated jitter).
Keep in mind the nos-DAC, which is just there to EXPRESS squares, once they are in the material anyway.


Gerner, you keep on saying that your speakers disappeared. All is relative as it appears now.
But this time *I* say my speakers disappeared. Hehe.

Thank you Gerner
Peter