Btw, you might think I'm a little irritaded ... well, I'm not. But when you don't follow, say, "instructions", I just don't know what to do. Also, I don't link you can let me chase and solve a wronly working driver *if* that is so. And I don't say it is, but it has to be proved.
May you be afraid of indeed proving that, then I don't think I can do anything.

Ok ?

I think you have quite some problems in your system, and the one I asked for several times now to eliminate (18 euro) JUST TO TEST, you won't even start trying.

As far as the status is now, you are asking me to solve Dogber's problems. UNTIL PROVED OTHERWISE.
Now please, shops are still open.

You do recognize that you are the only one, right ?

In the mean time, stop using FLACs, because THEY can be wrong as well. Then :

Please grab a normal WAV, look at the number of seconds it should run, take a stopwatch and look whether it plays as long as it should. Anyway, tell me how many tenths of a second it is off, okay ?

So now your are saying that all the time you get these errors but never mentioned it ? Come on now please ...

But to prove this ... what happens when you set DAC Is to 96/16 (!) and play the same file as before ?

Ahh ... I know now ...

This is related to the trick I apply to get Exclusive Mode let play modes it otherwise won't. But I didn't think of any 16 bit DAC (setting).

Actually, yes. I now take the physical length of the file (track), to be independent of often wrong header data.
Now what ...

Ok, I think I can find another means of doing it right. I hope ...

Thanks a bunch Edward !
Peter


PS: Didn't you see a lot of zeros near the end of such a track ? I saw them before (actually my glitch detection saw it but now I wonder whether that should be skipped for some reason ...

So to be sure : You let XX Double / Upsample, *also* set the soundcard explicitly to 88K2, and then it sounds better than without the latter, right ?

Maybe your soundcard resamples back otherwise ? is that checkable ?

In my case I could simulate similar : when I set the Fireface to 88K2 it would pull on the data; when I set it to "nothing" (for Mani, I don't use DDS) it will be "pushed" to 88K2. This looks like being different for sure, and whether it changes the sound ... must check that.
Keep in mind, this is the soundcard, and not the DAC behind it and it will follow whatever the soundcard tells (which can be e.g. 79.070 with the Fireface (1Hz steps)).

Yeah, that's the general idea.

I had been following that thread a bit already, and his basis is OK I think, but halfly put through, and for the other half applied wrongly. If I may play God here :

It certainly is not easy to look through the real merits of this, and the good example is given by soundchekk overthere (of course being the same as soundcheck overhere), who's response is allright for content, but not appropriate to the subject. The fun is, that *I* know exactly what soundcheck has been through hence talks about, him being the DTH 4 ME and me being the soundchekk (maybe 18 months ago over at bd-design);
People over and over come with great sounding tracks where special "technology" has been applied to, and the only thing I do is rejecting it because of colouring or whatever it is that's wrong with it, reading the file into Wavelab, and point out how dead wrong it is. Hahaha, over and over again.
Somehow this is about the art/skill of being able to hear that things have been manipulated indeed. Of course it needs some experience in looking at the digital wave form, but by now I can easily see at a glance what is wrong with it (if it is). Just think of the example of left and right channel occurring a few ms shifted from eachother. Now, would that create spatiousness or what !? of course it does. But it's manipulated.

What I actually start to learn (not sure though !) from that AA thread, is that it could very well be that no software player is applying what XX now does : using the bits where they should be used. Mind you, this is just the most normal, but you have to know quite a few things. Just relate this to :

a. The fact that Doubel/Upsampling as how it has been in XX until a few versions back, was just not real. Remember, it applied the higher sample rate only, and I actually never heard someone say that it did half of the trick only (not with other players as well).
b. The suggestion of people that a device like the Burwen Bobcat improves the sound.

I said it elsewhere, the "upsampling" must be in balance. Double the frequency, then double the bit depth. Use more than double the bit depth, and it will be wrong ! Or the other way around : double the frequency without doubling the bit depth (doubling is 1 additional bit) and it sure is wrong again (you'll get harmonic distortion from that).

This all must be seen to the respect of the plots from Mani, showing things way wrong (if really so, but let's assume it), and *that* actually not causing real harm. Again, this is about the nos principle.

In the end, what does it tell ? as said before : that a 1 zillion things are wrong anyway, and we hardly hear them. But wait till they're all eliminated ...
And that's why I keep on saying that XX is still in the beginnings of sound improvement. Just give it time ...

In that case I really can't think of what happens. If I only had it myself ...

As a side note with hardly a chance that it's related : not only me but others too reported similar, but exactly the other way around as what you describe : at the first initialize of the Soundcard/DAC of a higher samplerate setting, there's a 100% volume pop. This is not related to the fact that the soundcard/DAC already was set to that setting; it keeps on doing it, though randomly. This probably has been so always (with me anyway), but only occurs now because of the pre-amp being eliminated.

From your description I perceive all as the same plop I (and others) have, but in your case it is during playing. When it's not about the Invert I can't think of anything else, and there is really no way the DAC stops in between songs.

Ok ... I have one other idea;
Things have been changed according the calculation on where the music data starts in the file. I did not play attention to gapless, but I recall a few times the song started abrubtly. Like not in the beginning. If this is indeed the case, you should be able to notice it just by a gapless (live etc.) album. Thus, then gapless wouldn't work properly anymore ...
I put my horses on this one.

Edward,

For technical reasons (hence knowing the program) I can't think of anything else than gapless indeed. The strange thing is though, that I've been playing with the same settings yesterday for quite some hours, and nothing of the kind happened. Ok, one difference : I didn't use Invert.
Can you please check without Invert ?

Otherwise the only technical reason I can imagine, is that your (e.g. FLAC) files are converted late (slow system), that holding back playback for a while, and *then* the DAC is kind of re-initialized. Otherwise it's continues play.

But hold on. I think I can reason it out;
If that's an indeed "gapless" playing album you have at hand (like a live album), I could have made the mistake to apply Invert twice. Difficult to explain, but it would come down to one track playing Invert, the next not, the next again Invert, etc. Now *that* would cause the loud plop, because the voltage will go from minus x to plus x in one straight go.

So indeed, try without Invert and let me know.

hybride, I'm fairly sure that you talk about something else. Please don't mix up things in that case.
But if you experience the (100% !) same as Edward just explained, then please say so !!

With what setting (acc. Double etc.) and file (rate etc.) is dat ?

Nah ... Andrey ... I guess you didn't read up through the topic ?

When the DAC expects a word length of 32 bits, but the file is 24, those last 8 bits must be added. This can be done in real time, but this also can be done in a "pre-processing" stage. All plays from memory, remember ?
This 32 bit thing is just one example. The "processing" is needed, but it would be as bit perfect -> when reading back into 96/24 the 8 bits are chopped off again.
Don't confuse this with the digital volume, or inverted absolute phase. That obviously can't be bit perfect as such. And that too needs the "processing".

I think what we all miss is the lack of knowlegde on eachothers' DACs. LydMekk may talk in terms I indeed still don't understand, but you do just the same. The point is, this is not you. It is your expectation from my DAC. Here goes :

If I set my DAC to 88.2, it plays the file twice as fast. Ha ... WRONG. Again ... if I set my SOUNDCARD to 88.2, it plays the file twice as fast.

Done. Hahaha.
So *I* can't even start let my DAC have expectations.
I know from others that they explicitly have to set the soundcard to the samplerate implied by XX. I don't know what happens otherwise with them, but with me ? all automated here.

So it is just hard to communicate over these things.

Well, typo's won't make my english better to read for sure !
"by bass" = "my bass", and what I meant to say is this :

Following the development of XX throughout, one may have noticed that at some stage I measured (IIRC) some 24dB SPL more sub-low (under 20Hz) output on "a" XX version. Please remember, all is bit perfect always (but skip the digital volume from today of course), and as it turned out lateron, this version wasn't quite right. I reasoned out why that version could be better anyway, it made sense to me, and not to forget : all so often, explanations have to be found on why things turn out such and so.

Since 0.9t there is more bass output as well, although in a much higher frequency region than before mentioned example from maybe 8 months ago. Theoretically that is wrong. It "creates" audible standing waves, of which I always have been proud not to have them with XXHighend, no matter officially there can't be a reason for it. Anyway, since 0.9t they are back, and the last thing I would do is tweak my XOvers because it would be, say, wrong. BUT :

My XOvers have been set to a setting I like best anyway, and if you'd see the plot of the frequency response of my speakers, you won't think it can be right for sure. Still all is so on purpose. By itself this is not of the most importance, but for *me* it is important not to tweak like that, while in 8 months time I didn't feel the need as well. Since 0.9t I kind of had to, because after a few weeks time I couldn't get used to it *and* - and this is important- from nobody I heard a perceivement similar to mine. Conclusion : my bass setting must be too "loud" compared to what good playback means (??) incurs for as of today.
And from that point on (turning down the bass by some 9dB!) I'm happy again.

YMMV though, and by this time I can't tell what is and what should be. Just look at this post from the O.P. of the thread here, see our mutual agreement, and just try to "see" that this perceivement IMHO can only come from a not too high bass output (put it higher and things get coloured, fumbling, filling the room to much).

Despite all this, I'm fairly sure that earlier versions had more "headroom" on this, which all is related to, well, just a different representation of the low frequencies. What can you do with that ? I'm afraid nothing more than agreeing if you have the same experience, and hoping for the better at the moment I can again find best of both ways (read : that the highs have changed enormeously is without doubt, but often the bettering of one, goes along with the worstening of the other).