double & upsampling

[astacus21]

The right is that when i check upsample & double i get no sound and xxengine stop working.
The some with the new 96.dat.

The above posts are wrong because i had the older xxengine in the folder.

Ps If i check only double it says that the dac doesnt accept the data sequence (2|32|88200)

[PeterSt]

Oh help, I am getting lost on this one.

when i check upsample & double i get no sound and xxengine stop working.

Ps If i check only double it says that the dac doesnt accept the data sequence (2|32|88200)

These both do not combine for me. Double is exactly the same as Double/Upsample regarding the "setting" of the soundcard/DAC. So, if you are correct about both above, I should be able to find something really, but I don't think it can be correct ...


Would you please be so kind to

a. install XX in its current consistent version (i.e. no mixture with other XXEngine3 versions etc., and use the normal 96.dat)
b. act as if you didn't post anything about this yet, and explain as accurate as possible what does not work, and which works in Foobar WASAPI.

I really want to (and will) help you, but currently I am lost a bit.


PS: Your idea of using an old XX version in order to try, is actually quite good. But, you must use consistent versions always (I mean : as how they can be downloaded), and when such an old version works it really could be useful to know which one. But please don't do that right now because I first must know the exact situation at your side from the current version.

[astacus21]

OK peter sorry for the misunderstanding. I will try to be as more accurate as i can.
forget about the above posts.

Using the latest version 09v-3 as it is after download (i don't know if any registry value must be deleted before i use it)
with the new 96.dat file.

If i set dac is 96/24 (engine 3, double & upsampling checked) and play a wav, i get endpoint create failed > the dac does not accept the byte sequence, and engine exe doesnt start. The same with only double checked.

If i set dac is 96/16 (engine 3, double & upsampling checked) and play a wav, my external processor locks at 88.2 but the output is distorted.
nothing hangs in this case.

If you want any other information please ask.

Now lets go to foobar.

After a clean installation, and using the necessery plugins ( wasapi out, and a resampler), 
i check as output wasapi, buffer 100 and output format 24bit. I set the dsp resampler at 88.2 and play a wave.
My processor locks at 88.2 and the output is ok. the same happens, if i set as output format 16 bit.

I think that if upsampling in xx was at 16 bit, everything would work fine for me. for some reason there is a bug in xx or in the driver that doesn't allow 24 bit with it.

thanks again for the help.

[PeterSt]

Ok, thank you very much. One small (but important) thing though :

with the new 96.dat file.

Do you mean the one I uploaded yesterday, or the one received with the normal download ?

[astacus21]

the one you propose to download yesterday.

[PeterSt]

Ok, here's some thinkering ...

I looked at Foobar again, and I found a difference. Foobar will not output in 32 bits. That is, I receive an error from that setting. Now, XXHighEnd ONLY outputs in 32 bits (hence not in 24). Why ? well, because

a. my Fireface soundcard (/DAC) won't operate when fed with 24 bits
b. I didn't hear from even a single person that he couldn't use 32 bits (per the way I do it), or the other way around, that 24 bits was *needed*.

Ad a.
Somehow it must be my lacking knowledge that I can't (or couldn't at the time) get this going. I conclude this by
1. Foobar throws an error for 32 bits hence the 24 bit setting really must be different;
2. Foobar plays those 24 bits through the same soundcard/DAC which I (in XX) can't get going.

Ad b.
You might be the first.
Also, how could one ever tell without the comparison means, which now exists (Foobar). Thus :
More people might be suffering from this, and think it is their driver.

Despite the above, and assuming it is true that your sound device really needs 24 bits (and can't use 32), it still will be rather "impossible" for me to create something that works, because I can't test it as long as it just doesn't work (kind of your situation). Also, Foobar not allowing the 32 bits looks fishy to me the least. Why ? well, because the error thrown doesn't come from Foobar but from WASAPI, and it is just the same as the one you receive when you receive the Endpoint Create Failed message. Now here we have a problem, because this implies that both (XXHighEnd / Foobar) work mutual exclusive the other way around, and that can't be ... Added to this, is that I of course know what I'm doing, so without doubt I am outputting 32 bits. And this implies that Foobar is wrong, but I can't tell in what.

The latter isn't much important, weren't it that we both try to use Foobar to work out why XX doesn't play in the same situation. So, it *is* important.
Btw, sadly, I can check the bits used only up to 24, so I can't check whether Foobar by accident has things twisted (plays at 32 when set at 24 and the other way around (which then indeed doesn't work, like with me)).

Important : Might you check things yourself sometime again, always quit Foobar after a change in the bit depth area, and restart it. So, restart playback isn't enough, and you really can't see the difference (I can with some software I have).


What I will do next, is trying to find a way to let my DAC work at 24 bits, or prove otherwise that each soundcard / DAC needs 32 once more than 16 are in order.

[PeterSt]

Now I think of it ... there is another difference :

You would be using Vista SP1 (which is W2008), while I use SP0. And, since all (audio stuff) is so buggy in there, *and* Peter from Foobar says that really SP1 is needed it might just simply come to this :

I couldn't get the 24 bits going because of a bug in Vista, and now I just could. And, because I couldn't get it going, the way it tries it in the current version could be just wrong (remember, I couldn't test it because it just never worked -> chicken-egg thing).


Maybe you already told it, but what happens if *you* try the 32bits in Foobar ?

[astacus21]

Peter i have 2 things to propose. I don't know how difficult is it to accomplish.

First of all you can try with your onboard soundcard, to work at 24 bit. All this cr*py soundcards behaves the same.
They resamples 44.1 input to 48, before send the signal to spdif. If you choose 44.1 as output they convert again to this samplerate.
So we have 2 convertions and this is what i want to keep out of, with upsampling to 88.

The second is if possible, to modify the code of xxhe, so that if someone choose 96/16 or 192/16 (at dac is) the upsampling is kept at 16 bit depth or to have the option of 16 or 24.

Anyway i appreciate your help and your hard work with this program.
For me even at 44>48>44 khz xxhe sounds much better than foobar.
It is the winner by far

[astacus21]

Now I think of it ... there is another difference :

You would be using Vista SP1 (which is W2008), while I use SP0. And, since all (audio stuff) is so buggy in there, *and* Peter from Foobar says that really SP1 is needed it might just simply come to this :

I couldn't get the 24 bits going because of a bug in Vista, and now I just could. And, because I couldn't get it going, the way it tries it in the current version could be just wrong (remember, I couldn't test it because it just never worked -> chicken-egg thing).


Maybe you already told it, but what happens if *you* try the 32bits in Foobar ?

I get an error. i can't use 32 bit BUT for me vista sp0 have bad impact to many other things including speed, latencies etc.
I can't use q=-4 with sp0. I recommend to anyone to use 2008 or vista sp1. especially all the users that have a good soundcard with native 44.1. Give it a try, i'm sure you gona like it.

Ps: vista sp1 is not exactly w2008. It is difficult for a simple user to set up 2008 but for power users is quite easy and much faster than vista.

[PeterSt]

The second is if possible, to modify the code of xxhe, so that if someone choose 96/16 or 192/16 (at dac is) the upsampling is kept at 16 bit depth or to have the option of 16 or 24.

That is exactly what XX is doing by that settings. Earlier I checked whether those 16 bits are really 16 bits (and not more by accident), but it is OK. This is why it is so strange that it doesn't work for you (combined with that it works in Foobar of course -> Foobar does those things correct as well; I checked).

And thanks for your nice words. We'll keep in touch.
Peter

[astacus21]

Peter i think i found something...

Upsampling at 16 bit is ok. I roll back the drivers of the soundcard and now i get normal sound until q=1. at 0 i hear clicks and pops
and below 0 i get the same kind of distortion i had with the old driver. i' m little tired to test right now, but it seems that the previous driver had smaller input latency than this one. So i suppose that the only problem for now is why exclusive mode rejects 24 bit in xx.

Another question i would like to ask you, is if xx take care of the clipping that sometimes creates the upsampler in some recordings.
What is the difference between double & upsampling to double, upsampling and AA?

[PeterSt]

Ah, good ! so the (for me) most strange thing has been solved.

clipping that sometimes creates the upsampler in some recordings

I don't know what you mean ... With the AA checkbox unticked, XX can't create clipping. The music data may already clip from itself of course (this happens in more "recordings" than you like).
If you think there is a problem somewhere in this area, please start a new topic for it, ok ?

Double is just changing the sampling rate (to double the source value) without touching the data.
Upsampling is also changing the sampling rate (double or quattro), but now the additional samples are calculated; without upsampling the additional samples are repeats from the previous.
AA = Anti Alias is a more official way of upsampling but which changes the data all over to filter out aliases which always emerge because of upsampling (because Nyquist gets disobayed then).

Thus, first you tell what the sample rate must be (double or quattro the source) and then you tell how to output that (upsampled or not, and if yes with or without AA filter).

[astacus21]

With the AA checkbox unticked, XX can't create clipping.

OK peter thanks again for your help.

Is it possible that the distortion i get at -1, -2 etc is from the output buffer of xx?

[PeterSt]

I tried to explain this here :

The Foobar buffer in Playback - Output does not act the same as the buffer in XXHighEnd I refer to BUT at using the Foobar buffer at e.g. 100ms for sure you would feed the audio chain with something which is much more "at ease" than the XX buffer at in fact any setting.
To work as stressless as possible in XXHighEnd you should set the Q1 slider to a higher value (but too high and it doesn't work anymore (skips and things). And I see from your sig that you use a Q1 of -4, which is the most stressful setting (but generally with the best sound !) and which for sure not everybody can use ! (ticks, distortion).

You could say that the Q1 slider towards the lower levels, brings XX closer to the DAC. It influences the DAC more, which is not necessarily better, but in general (I mean for most people) yes, it is better.
It is not said at all that one is able to play at -4 because this highly depends on the latency of the system, but also on the playing software itself (if you followed the development of XX you've seen that at certain versions suddenly people couldn't play at e.g. -1 anymore and had to use a higher number).
Concluded, it is not strange at all when one can't reach -4 because things get distorted, BUT you could say that if your system can't reach -4 your system isn't optimal (and mind "system", which is a very wide phenomenon). The same counts for the upper level of Q1; it is not said that you can play without distortion at the topmost levels, which depends on many things but mainly the DAC itself.

The Q1 is a very fragile thing, and with some experience you can hear that at the lower levels - when too low - violins start to sound digital. This is no distortion yet, but a too high influence on the DAC and you could say it gets nerveous. Now, at interpreting these things one must be very very careful not to draw the wrong conclusions, because the too digital violin IMO is just the better representation of something which is digital from the beginning (the WAV file). You can hear this by sensing that all receives better detail at the lower Q1 levels, and you could say that as long as it is not about violins, it is just for the better. Mind you, this all is so fragile that I certainly wouldn't bother to slide up the Q1 when a violin comes by, but I'd know that violin can sound a bit better. On another note, I already play the Q1 somewhat higher lately (4) which has a reason of course (and I can play -4 without any problem).

Interesting for you might be the fact that the Q1 values of -2 and -3 are a kind of special to the sense of that those values don't workout in a consistent way. Hard to explain, but thinking in terms of the time something takes (is ordered to take) this time varies at those Q1 settings. More at -3 than at -2. But, since there's an averaging aspect in there (think in terms of how long something takes on average) the effect will be that those -2 and -3 values incur for even more stress than -4 (-4 is a consistent working value again). Thus, when your system coincidentally is in the critical range, it might happen (theoretically that is) that -2 and -3 can't play without distortion, but -4 can.

Lastly remember, in general we can say that the latency used is 1ms, and if something holds up for longer than that, you can't play (without distortion). For that matter, the 100ms as the minimum from Foobar could be interpreted as 100 times more headroom for other tasks in the system.
Btw, before you ask, the 1ms XX uses, has a purpose.

[astacus21]

You could say that the Q1 slider towards the lower levels, brings XX closer to the DAC. It influences the DAC more, which is not necessarily better, but in general (I mean for most people) yes, it is better.

On another note, I already play the Q1 somewhat higher lately (4) which has a reason of course (and I can play -4 without any problem).

This has to do with the output latency, i mean 1ms is the same for all Q1 settings?
Or it has to do with data output, for example lower q1 sends, (how to say) more "accurate" signal to dac.

Another question here has to do with phase. When i hear at -4 region if i don't tick invert phase the sound is very unconfortable.
At +2 or +4 the difference is not so huge, I'm not sure what is better. Is there any explanation for this?

The Q1 is a very fragile thing, and with some experience you can hear that at the lower levels - when too low - violins start to sound digital. This is no distortion yet, but a too high influence on the DAC and you could say it gets nerveous.

I think "digital" is not the right word. I saw in another post someone else use the some word, but for my ears i have a feeling of more analog,detailed and comfortable playback. I dont know if my tube amp has any influence in this. I want to hear some seperated instruments sounds to verify that. With music by lowering Q1 i get more pleasant sound.

I hope not to bother you with all these questions