Hi LydMekk,

Thank you very much for your description. If you only didn't copy mine I sure want to copy yours. And your description of the bass is better than my own; my "more separated" indeed intended to say something like more deep or maybe "in its own subwoofer world" (while I don't mean 20Hz regions, but something like the individual low vibes of e.g. bass guitar strings so good audible that it is like they sound through an undisturbed own speaker).

[...]

But I don't understand what you mean by this ! or ... what your implied impact is. I mean, if I "set" my soundcard to e.g. 96KHz (instead of let it auto-adapt to what it's fed with e.g. 88K2), the song is played faster. Can you please explain ?

But now take this :

There is plastic and there is metal (see earlier in this topic). Tonight, for the first time I listened to "decent" (because before it was just wrong) Double/Upsampling through my audio (nos !) DAC. Now :

I'm under the impression that the roughness (or rawness) as described earlier (btw not by me) coming from the nos DAC and XX since 0.9t (this latter is my own perceivement) is just compensated by this. All is very fragile, but still ... cymbals are more singing, and the sole thing ever occurring to me throughout albums was the better seprated bass (instruments) from all. So was this better ?

Maybe Yes, maybe No. Because I also noticed a downside (similar to yesterday with the Fireface) :
When the music gets more loud, things get too much shouting. This again is (IMO) about the higher frequencies no being able to fill gaps from the lower frequencies, that by itself possibly caused by the lower highs (volume) output. And so :

As how it worked always so far, it will work the same now : since I clearly perceive better things from upsampling (only Double this time), it will be a matter of finding a means to retain the positives, but eliminate the negatives. I don't know how yet though ... could be a matter of the pre-processing thing maybe ... I hope to get that ready tomorrow (for myself at least ).

To clear things up maybe :


OF COURSE not.
I gave it as an example how we can get fooled, with maybe the most important message : who actually can tell which sound stage is the most real for width and depth ?

I must admit though, that "laid back" is another phenomenon than "less deep". Or IOW : everything on the foreground (compare : right in your face) is different from everything in the background. Or maybe even better (though a negative) : everything being too loud cannot be compensated by a volume control.

One other thing to think about :

Since this is a topic that makes me brabble around anyway, take an MP3 if you dare (take the higher bitrates though).
So far I did not spend much attention about it (too much ), but you may find yourself finding it *better*. Maybe the pinpointing is less, but for sure the stage is wider and deeper, the bass is better ... Now what ?
And yes, I even have theories for it why it can be better, and 0.9t (and u for that matter) is a bit based on this theory. Mind you, a theory I had to come up with afterwards, so nothing I could reason out in advance.

What does this tell ? well

a. do never think that an e.g. wider sound stage is better, because how can an MP3 be better ?
b. the fact that, say, noise is added, not necesserily means we perceive it as worse. It should be worse though for theories.

Anyway, this kind of explains why we also can perceive the nos DAC to be better, although there's some more to that.
For those who know : I've been always referring to the Kodac DCS whatever camera for comparison. There is *NO* camera showing its resolution better than that one. There is NO camera more difficult to deal with at making photo's and avoiding the things it just can't do, like a roof with overlayed stone cover (sorry I don't know the name in english) which will show moire all over. Even with trees you must be careful. And you know what ? there is just no AA filter in there. It is the only camera which allows to see the 14MB picture at full size, and it wil show no degradation on sharpness. Also it is the only camera that will not need sharpness appliance ...

With these things in mind, it is more easy to see why we can perceive things better, while actually they are not, or why we *should* perceive things better when we only comply to the rules.
This is a great deal of XX ...

So far the discussion is not about NOS, but sure the lacking filtering in there comes down to similar stuff (/ perceivement ?). Otoh, I don't think it is good to bring that into the mix, because it will be different for inherent anomalies and (listening) results.

Then, I think it is important not to mix up with unrelated things as to what the alias at the other side of the mirror might bring for audible anomalies on one side, and what obviously happens on the left side of the mirror which SHOWS anomalies. It will be 100% sure that what shows there, is audible too.
Since we left our books covered with dust, we could wonder how the right side of the mirror (which makes perfect sence that that aliasing  happens) influences the left side *if* it is that what is going on. And I am not so sure about that yet.
Probably right now we are a bunch of stupids with to few knowledge to talk about this seriously, and IMO the only reason why it could be interesting even to the knowledgdeable, is the possible situation that an AA filter not only removes the aliasing, but *also* compensates for the known highs roll of implication, and just nobody saw that the roll of is already there when the aliasing is just left alone ...

With above fuzzy lines I only want to say this : I don't think we should try to work out (or "find out") the obvious, for those who just know. We'd only make oursellves more stupid to those who do know.
On this matter it is my idea to ask a person of which I'm 100% sure he knows ... Bruno Putzeys. Why him ? well, because he has been all over into this stuff, *and* we actually know him. He might not recognize it, but it was me asking for the highest grade Hypexes as I think are around by now, and hybride, you might know him more personally while I only talked indirectly to him (via MM-Audio). Send him an email, direct him to this topic, and ask politely ...
Btw, for those who don't know him : he once DIY created an amplifyer taking DSD for input ...

Now, since I actually don't even have those books (yeah, you knew that) there is much more going on which I found well over two years ago, and for which I don't have explanations. In brief this is about the whole audible spectrum being FULL of aliases, if you only know how to look at it, and how to incur for it. I can tell you, only nos DACs show this best, and actually it is quite unbelievable we can hear music through that mess. This too is about the Nyquist frequency being a mirror, and "colliding" parts of the mirrored frequencies being able to create the exact same SPL as originating frequencies elsewhere. Believe it ot not, but e.g. a 16Khz (somewhere around there) tone can create an as high volume at 20Hz. Mind you, no spurs of 20Hz are in the original tone in this case.
This concluded for now : the nos DAC is as bad as can be (for matters you can't guess and which I did not explain), BUT WE LIKE IT.


All 'n all (again, for now), assuming I can't hear the anomalies on the other side of the mirror, the only thing I see is a roll of of the highs (but careful, because you can see a drop a 2KHz already) and an anomaly at the low end. If this indeed would be all that's the matter it hardly harms sound but for more dullness (and what comes from that of course -> should be less detail). Mind you, apart from the thingy at the low end, it's rather lineair.

Last thoughts : if this is a phase impeeding thing, I'd say that we can't capture it with a *digital* loop back. But I can't believe this is phase related ...
The strange things I found I talked about in the above, are in the pure digital domain also ...

And oh, let's keep in mind in advance of things : theoretically it can be that what we see is not real. Not so much because RME software would be wrong, but because the representation of it all actually also needs an AA filter. How this can be made consistent with software applying an AA filter (like most probably Foobar and its used plugin does), and all looking good, is another thing.

Peter


PS: This didn't incorporate hybride's last reply, above.

To a certain extend, yes. Now you can't set the Core Appointment thing. This *does* differ, although it depends on your system as a whole, how.

The only "useful" thing I can say for now : take your time on finding what the problem is, and during that time you'll know that in the end (when you have succeeded on solving it) the sound quality most probably will improve.

Of course I'll do anything to help, but I really can't tell anything else from what has been said in that specific topic.
Maybe you have a friend around who is a systems manager and who wants to help you ?

g. http://www.epanorama.net/documents/audio/spdif.html

uhohh


Thank you for the Foobar test Mani.
Now I wonder what happens at double/upsampling with "DAC is" at a 16 bit setting. If this looks normal, then

1. It is most probably the SPDIF limit;
2. Foobar (SRC or whatever it is you were using) doesn't utilize the bits (only uprates the sample frequency).

Btw, I am not asking you to try this, but can't do it myself at this moment (tonight I can).


Oh, when things still don't look normal, I'd try ADAT instead of SPDIF.


PS: I was Googling for an hour or so on the SPDIF matter, but couldn't find a clear answer. Keep in mind that "we" are using SPDIF over Firewire (400 in my case) and that things may matter there.

PPS: I got the idea here : http://www.avsforum.com/avs-vb/showpost.php?p=13353422&postcount=897 which was singing in my head the whole day because I couldn't believe that, while on the other hand, that (MS) guy should know. Suddenly the combination with our topic here popped into my head ...

Ok, I wanted to put this in a separate topic, but actually it is 100% related, so here goes :

Mani, I don't know about your signal processing skills, but personally I don't understand what I'm seeing, looking at your plots.
Remember what I said elsewhere : "I don't like to apply an AA filter, because it would roll of the highs" (similar). Mind you, this thinking of mine is taken from what can be read all over the place;

Your plots show (and mine obviously will too) that the roll of is already there *without* an AA filter. Now what ? was everybody looking only after the AA filter had been applied ? Or IOW, what is actually going on here ?

a. there's a problem in the (XX) software;
b. the DAC isn't capable to cope with the high frequencies at the higher resolution (that's what it would come to) (for others : keep in mind that this goes over the analogue output of the Fireface, with or without loop back cable (the loopback it can be done internally);
c. the aliases influence the phase of the audible side of the mirror, and therewith attenuate;
d. the method of measuring is faulty;
e. this is just normal signal processing theory stupid, get your books !

Please keep in mind that I look at this all from the context of knowing what the program does. So unless a. above is applicable, I can't think of why a sequence of samples coincidently implying a higher frequency, would decrease in volume (that's what the plots show) other than c.
I don't change that volume, nor is there any arithmetic that implies it ...

All correct Mani. Including the last one.
There are some more combinations though, like playing a 96/24 over a 16 bit (set) DAC. That too is processed in real time, and that too can be setup differently.

Note that at some stage the advised 2GB of internal memory will go low ...
(just think of 96/24 files being pre-processed; they are really over 2 times the size of anything which I had  in mind of needing to be in memory ONCE MORE hehe).

Allow me to say this latter is a good remark, or one I try to deal with explicitly anyway. On that matter, keep in mind the explicit change I wanted in 0.9t, which was about "more metal" where things had gone too plastic.
In all cases this can be compared with (digital) imaging, where added noise perceives (nothing more than that) more sharpness, while the opposite - denoising makes the image litterally plastic. Ok, this is not abpout (de)noise, but for "sharpness" things come down to a similar matter. The key to my last referred to link above, just *is* in the area of sharpness ...


DAC Needs is a technical setting by itself, but when set to a lower rate than actually possible (which would come down to choosing a 16 bit setting for DAC Is (!)), higher bit depth files will be (explicitly) cut.

Your suggestion that it doesn't change anything would be true for your context given, but you are forgetting the volume control. Only for that, a higher bit depth setting is of life importance.

"Tap my feet more" is a perfect description. In fact the best where other descriptions fail.

Ok. Why do the both sound different ? well, for me this is perfectly explainable, but let's say for now that the code for both situations is different. And -so far- (see later) this had a purpose;

The 44.1/16 code has remained exactly the same as before to the point of sound quality. I did this on purpose, in order to not dissatisfy people which were satisfied. By itself this is related to the necessary code changes to allow the larger bit depth. Thus, actually the latter should have urged for different code for everything, but I just retained the "old part". In fact I was hoping for someone to notice the difference (and remember, I myself won't use the old code because I want the larger bit depth to be active, that by itself because of the volume control).

Now, because the old code too is (has been made) subject to volume control, *and* I wanted to pertain the old code for SQ, this is pre-processed for the volume control. After that has been done, te playing code is 100% the same. This is similar(ly woking) to how I can guarantee FLAC not making a difference.
Once a converted bit depth comes into play it was more easy to do this real time, and the necessary code for that influences sound.
And thus, we already can see the next sound improvement coming up for those who exploit the additional bit depth (which already is in order at 96/24 files, no matter your DAC can do 24 bits only, and which is related to all being transported over 32 bits ! ("DAC Needs")).


For a few reasons this gives the major impact on SQ. But careful though, because your Hypexes may respond to the vast improvement on transients, and so far accepted sibilance could become unacceptable. I don't know really, since the only D/T amp we skipped at the time (about one year ago) was the Hypex, but all the others had unacceptable sibilance to begin with. According the principle used in there, I expect the Hypex not to be different, and I only want to say : be careful in your judging on what you actually hear / listen to.

When you use the volume control, you should not avoid the additional bit depth, unless your attenuation will not be more than, say, 24dB.
For now this means using the "bad sounding" (hehe) code, but I will change that since I now know that it makes the tapping feet difference indeed (no matter it is only you saying it ).

On the digital vs. analogue volume control much more is going on than allowing for the pre-amp to be eliminated. What I found on this remains a secret so far (The 0.9u what actually happened topic ... ).

Haha, I can tell you, flat as a (vertical) pancake (I tried that same plots to find all the bugs ).


Well, you got my message right ...  
On that matter it is my idea that anyone who compares apples with apples (which I can't, and I'm not sure whether Mani uses the same DAC in all cases) could come to the conclusion that Quad/Upsample (but into 18 bits at least !) is better. And as I said earlier, the sole fact that I heard more information on known records, kind of makes me urge to explore it further (but see next post !).

For others : keep in mind that we all don't use the same amps. However, when the battle has to be fought over Leo, Mani and me, they are equal regarding the speed (which IMHO is an important factor to the subject).

Haha, that's why I'm asking (and why it is important). Two other questions please :

1. Do you use the volume control (either case) ?
2. Could you describe the difference between both cases (always playing 44.1/16 of course) ?

Btw, I already know my answer, but want to learn the perceived difference in the end result of things. Also note that I never listen to the "DAC is" 16 bit anymore, because I use the volume control (and really want to use my extra 2 bits (from the 18 bit DAC)).

Thanks,
Peter

Hi Bernard,

Keep in mind that the user(id) itself must have Administrator rights as well. Not only at changing the UAC settings, but also when running XXHighEnd.
Otherwise I can't think of anything.

You are not using Vista/SP1 I suppose ?


PS: Quite some fast system you have !

Well, what can I say ...

I based my enthusiasm (in the release notes) upon more perceived detail hence things there which I didn't hear before, things not getting wrongly to my brain, and an important factor - listening though the Fireface as the DAC with the pre-amp in between again.
I did not A-B anything, knowing that the Fireface as DAC already would make the situation a loosing one, BUT, I don't recall ever being able to listen to/through the Fireface actually hearing better elements.

After a couple of hours it started to loose its interest, but (for me) more importently : when I shut down the music at dinner time, my wife said she was glad I did. In the mean time I had been turning down the volume at some tracks (also an indication of the very wrong), and all 'n all ... it can only be so that this is not it. No matter how I want it (like the movie thing), it probably can't work.

Maybe if I do build in an AA filter afterall (of which I'm near sure it will be for the worse).
First some other things though.


Can you please confirm that this is actually what you do and want to say ? I mean, this is different from "setting" the DAC at a higher bit depth (the frequency is unrelated), still listening to the native track (hence not doubled etc.).
The answer is kind of important (will tell later why).
Please, I don't ask you to sit down and (re-)listen (actually you should not), and only want to know what you meant to say.

Peter