Hi Dave,

Maybe I don't understand all of your post, but


No. You can just leave it at 44.1/16 and it plays the file (and you won't hear the difference). Mind you, the DAC needs 96 KHz for this. In your case you won't be able to play the file at all (why ? because XX doesn't downsample from 96 KHz to 44.1 or 48 Khz -> it could, but it doesn't seem the right thing to do).

Guys,

I am not sure what you are experiencing. Anyway apologizes, really !

a. I think with the new way of assigning the DAC (that avoiding rejections by Vista) things have changed. However, I myself never noticed that at 44.1/16 playback (which should be Dave's situation).

b. When the format of the file has changed there should be a hold on the next file's playback and a message as how the Releasenotes on 0.9u-0 (I think) announces. Tonight I witnessed (outside, doors closed) the family jumping at the sealing because the format change wasn't captured anyay.

I'm really really sorry, and I guess testing of all the combinations (of changes) takes ages to test thouroughly.


That's what I noticed too, assuming the max level "tick !" which -no matter how prepared you are- drops you of your chair. This would be #a above. #b is even worse because it lasts (a few seconds ?).

Please note that -so far- all came to me at changing from 44.1 to 88.2 or 96, so it seemed relatively rare. However, what Dave claims is that it can happen, say, always.

Btw, I checked for this earlier today, and it happens in the DAC, not in the (SPDIF passthru) soundcard.

Possibly this was happening always, but was attenuated by the analogue volume ??

Racing accident !
Afterwards it looks funny of course, but I really couldn't make anything of it.
But I think the real one to be sorry is me. About the subject and "wacky tracks" anyway. All, of course, given to me by that first track which indeed doesn't work and the OLD text which was still in my mind.
As said, racing accident.

Edit (hehe) : ... and sure not to degrade your findings, which are ever so right as always.

And if you'd only know that yesterday (just before the upgrade) I tried to make that slider useable after the restart of XXHighEnd at UnAttended Playback and I couldn't manage ... So I undid everything and put up the upgrade.
Now I understand why I couldn't manage ... it just doesn't work in the base anymore.

Yes, it definitely is related to Attended Playback. Unattended is allright.

Hmm ... this has a high degree of strangeness. I've played quite a few albums yesterday (with the exact same 0.9u-2 version as you have). But today this Ayo woman indeed goes wrong after track-01. And it's not even the same album release ...

One difference maybe : now I play Attended ...

PS: From 02 to 03 it goes wrong just the same.

Yes, I just saw that in advance of you (trying the same album hehe). But do you (and Gerard !) imply that things go wrong only at using the slider ?

Oops. I didn't read that properly (but caused by your changed text, sorry). But then I wonder what you mean by "going from track 1 to track 2" ... this wouldn't be possible (?).

Are you sure you are using the same files as yesterday ?
Btw, my responses to Gerard are kind of wacky ... say, as wacky as a few of his files ...  (Hi Gerard, but you know what I mean !)
About yours Mani, I don't know.

If the're not the same as yesterday, what kind of files are you trying to play ?

Well, if you at first post only this :


and nothing else ... what can I say ...
But okay, you added "the explanation" later ...

Your errors come up "more" lately, because you try more wrong files lately. Hahaha. But it's true (because XX allows for the formats).
That first SACD whatever thing is not right for starters. I know it ...

I'd better change the message "No track given" into "File format is illegal".

Ok ?

Well, you get a prize for the most useful post of the whole of 2008 !

I think you can find some answers in this post (from 09.U-0a setup with external USB DAC).


Less dynamics is what everybody would say. This is stupid theory though (IMO), because *or* you are not able to hear the full range, *or* -when you think you do- you will hear distortion at the low end instead.

Just try it with XX which very officially presents you the attenuation in dB. Take a 16 bit random file, set XX to -0dB, the preamp to any volume you think is bearable. After this, set XX to -90dB. Officially one bit is left now, and no music can come from it (and no dither is applied). However, when you are able to hear it, you'll hear it as distortion. Now, can you hear anything ?
Keep in mind, the volume of the preamp should be kept at bearable listenin levels !

My suggestion is : no. You can't hear it.
So now tune up XX until you start hearing something. Ok, in the before you obviously were with your head in the speaker, but keep in mind this is not much realistic for obtaining "dynamic range". So better step back to the listening position, and find out when you are starting to hear music at tuning up XX' volume.

Say your answer is -66dB. This would mean that the dynamic range you can obtain is ... well, 66dB. Keep in mind that the 0dB is the "bearable level !".
Sidenote : "bearable" might be 105dB SPL (listening position !). So what you -in my example- actually still can hear is 110 - 66 = 39dB.
With -66dB you'd have lost 11 bits so 7 remaining. Way too few for good music. But *now* you'd have to search for music that allows playback at 105dB SPL *and* has music information at -66dB. Of course this exists but it will be more rare.

A bit depending on the outcome (and your system), you can imagine what happens with a 24 bits DAC. This means 144dB so called dynamic range, and there is NO WAY to unveil it.


The noise floor too, IMO is a phenomenon which does exist in theory, but in practice works out slightly different.
My point would be this (disclaimer : by lack of real signal process knowledge, haha) :

In a DAC there is nothing like analogue noise with will show underneath its capabilities. That is, not derived from the digital capabilities (from the analogue output stage there is, but this is not about the "noise floor" as we speak about it). You must look at this relatively :

When a 24 bit DAC is fed with 16 bit material derived from a 24 bit file (like you can do now since 0.9u-2) the cut bits will create noiSES. This is nothing else but harmonic distortion occurring at a certain level, and that level will be just around the noise floor of 16 bits.
Now here we go again : the noise floor of 16 bits would be -96dB, and I think in the above we determined that there is no way that you can hear that. The dynamic range you'd need to obtain it, is too large.

The same noiSES will appear when you feed a 16 bit DAC (or any bit DAC for that matter) with, say, 12 bit material derived from a 16 bit file. Again, the relativeness is important, because again the *cut* bits will create noiSES, or better, the remaining 12 bits actually needing the 16 bits will do. So the noiSES floor is 12 bits, and this is -66dB. See the story above.

What actually happens ?
A more fluent sine is cut into more square cutoffs because there's not enough bits anymore to keep it as smooth as it was. It is those squares causing the harmonic distortion, or maybe better : squares have a signature of over tones that don't belong to sines.
A side effect of squares is that they bear more energy than sines. And this is why this "distortion" can be audible above the noise floor. They *will* be when the needed attenuation is so large - and which follows from too much gain - that they're in the audible level.




PS: for fun, look at a 24 bit file opposed to a 16 bit file. When you follow my reasoning, you'd conclude that a 16 bit file is full of (relative) harmonic distortion opposed to the 24 bit file. The sine in the 24 bit file is rather endlessly more smooth opposed to the 16 bit file.
Now, who says - or what determines whether only 24 bits is good enough, 16 bits can do fine, or 14 bits are okay just the same ?
The dynamic range you can perceive does !
Now I think you can better understand why I said in the Releasenotes of 0.9u-2 that to my theories you wouldn't hear the difference between a 24 bit file played as such, and the same 24 bit file cut to 16 bits (when played with XX ).

Hi pedal,

(btw, is there any importance in the picture underneath your text ? I see a read cross only ...)

The answer is not all that simple (for understanding) I'm afraid;

Let me first say that if you do not want to use the Digital Volume from XX for your own reasons, for 44.1/16 playback it doesn't matter at all.
The point is in the if though ...

I determined that applying the digital volume is better than the analogue volume. Never mind why for now, but I reasoned it out en so far it looks ok. Additionally this - by accident - allows you to get rid of the preamp. And, as this is a major smooth-operator holding back transients, creating noise and all, it slowly becomes a matter of "needing" the Digital Volume to just have a better sound quality.

Btw, that a preamp holds back and therefore better is to be avoided is notthing new, but -so far- we couldn't really see how to do it right with the ditgital volume and really too few bits available for it. Now, what is applied in consistency in XX is that with 16bits input and 16 bits output all is - or just is not sufficiently legit.
Btw, keep in mind that with 16 bits in (say, the CD) and 16 bits out (your DAC as it sadly seems) everybody on this planet will say STOP !
And when nothing "special" is applied at digitally attenuating, they are very much right ...

And mind you, all who apply official digital attenuating (like in the pro world (which includes RME and the like) will do so by means of internal 40 bit or more float processing, and XX does nothing of the kind ...
With this I only want to say that we're way off the usual tracks here, and my own theories including the practice as of now, tell me it just can be done. Ok, just, or just not;

As I said before, the just / just not is highly related to the gain you apply and the sensitivity of the loudspeakers;
At first, when I couldn't exploit my own 18 bits of the TwinDAC+ (XX could not do that yet) I determined that as long as I kept above 48dB attenuation (so, towards 42, 36) all was okay. But mind you, this was absolute listening without the possibility to compare the 16 bits DAC (useage) to the 18 bit DAC. I thought I could hear anomalies at 42dB, but I took them for granted NET. Together with that I would never play that soft, and it's merely -30dB or -24dB. At -30dB too, I again I thought I couldd hear the anomalies, and again I took them for granted, and again NET.

Please keep in mind that there is a "balance" between more attenuation and the SQ getting worse on one side, and the less attenuation and less distortion (that's what it plainly comes to), or IOW the louder you play the better you can hear distortion obviously, BUT the distortion by itself gets less. At 0dB there's no distortion left on these matters ...

With the repeated "NET" I refer to the overall improvement, and *if* I can hear distortion somewhere a. I'm not even sure it is "that" distortion I hear and b. because of a. and the overall better result, I don't care.

Mind you, this is all with the 16 bit DAC.
With the 18 bit DAC though, the distortion (if audible at all at the given level) officially comes forward 12dB "later". Thus, when with the 16bit DAC audible distortion is there at -42dB, with 18 bits the very same distortion would be there at -56dB. Now here the mentioned balance comes in again, because when you normally play at -30dB, there is no way there's audible sound at all at -56dB. Try it (not with your ear in the speaker, but at normal listening distance).
To make a long story short, I am sure that I hear the difference between the 16bit (used) DAC and the 18 bit DAC both at -30dB. Also, when you A-B this, it becomes more apparent what it actually is you hear with the 16 bit DAC and which you took for granted. It's a kind of roughness that I can *not* really explain from missing bits as such, but of which I know by experience that they are needed to fill out "holes" at certain frequency levels (or better : from certain instruments). This is similar to how I eliminated "roughness" by just adding gain to the highest frequencies in my loudspeaker filter (which lineairly goes up from 5K to 20K by 16dB !).

With 18 bits I'm satisfied at -30dB, but maybe with 20 bits I'm more satisfied ...
Suppose that 18 bits would be the "standard" for legitimally play back at -30dB, then from that follows (it's just math) that with 16 bits all is as legit at -18dB.
When you are not able to play at -18dB (because it's too loud) the relatively simple solution is to adjust the gain of your amp. NOT by means of attenuating the existing gain !

The latter comes down to using the digital attenuation in the first place. So when you just don't use that, there's really no problem with 16 bits files that a more than 16 bit DAC can solve. Not that I know of.
Doubling or Upsampling to 88.2 KHz was created for better SQ (mainly Doubling !) but if it works out like that ... YMMV. For me it does not, because it needs a larger buffer in the soundcard (from 48 to 96 samples), and -again- NET it is not better because of that.

Please keep in mind, all of the above is related to 44.1/16 files only.
Once it would be a standard to have 96/24 etc. files, the "problems" are obvious, because you just won't be able to play them.

I'm not sure if you'd need any advise on bringing back your DAC to April Music, but if you spent that amount of money with the idea it could do 96/24 ... well ... The proof that you could be mislead is sufficiently available in this topic. . Along with that, the person involved not really to blame for it as well.
For the future : when the specs of a DAC do not specifically mention input samplerates and bits, better assume 48/16 as max. For that matter the CD10 does mention 96/24 for input explicitly. Whether *this* then is true (over USB !) I don't know.

I hope this is something for an answer pedal.
Peter

One of the disadvantages IMO of the DEQX is that its digital input determines the number of bits it uses for its internal processing. Thus, where it seems that CDPlayer -> DigIn-DEQX(DAC) -> AnOut-DEQX is best, Analogue-In really should be better for overall results because then it uses maximum bits for its internal processing.

Of course, both internal 16 bit processing and DCPlayer -> DAC -> (ADC)AnIn-DEQX(DAC) -> AnOut-DEQX come down to a huge compromise.

Now please keep in mind that when you set XXHighEnd to "DAC needs" to 32 bits and you feed that DigIn to the DEQX, it will apply internal 32 bit processing.

Peter

Allow me to first extend the list with one other option :

1. XXHighEnd>Volume slider at   0 db>Passive -18db (resistors)>AMP
2. XXHighEnd>Volume slider at -18db>AMP
3. XXHighEnd>volume slider at -18dB>Passive 0 dB>AMP

When you are afraid for the 2nd option you can always use the 3rd. I know, it is the most peculiar way of doing things and I don't think you will find anyone stating the same, but I say it is better. Just give it a listen (and A-B with the 1st option !) but keep in mind the noise the preamp may (or will) add with the 3rd option.
Might you agree with me and you find the 3rd option better indeed, you can always go for the 2nd option at the time you got used to all and are confident that the "emergency" volume knob of the preamp is not needed.

The 2nd option, or IOW eliminating the preamp, will give you the best transients your system can unveil, BUT, now you are fully dependent on the AMP and DAC. Thus, any "smoothing" the preamp will apply otherwise, now will not be there, and e.g. harshness of AMP or DAC will show, if there. In that case, do not think your preamp made the sound better ... it only masked bad things from the rest of the chain.
Without the preamp (passive or not) the sound will be totally different ... that's a guarantee !