XXHighEnd

Great, absolutely no L/R phase problems, with or without upsampling :smile:...

... but my preference is still no upsampling. Sorry Peter :(

For some reason, the best SQ seems to me to come for selecting 'DAC is 16 bits 44.1 KHz'. I've just been comparing with Foobar/ASIO... and in this configuration, there is simly no comparison. XXHighEnd is simply more musical... almost more 'stable' and 'solid', if that makes sense. The way I described it to my wife was; XXHighEnd is 3-dimensional, Foobar/ASIO is 2-dimensional.

I could listen to it all day long... and still believe it is the best investment I have ever made in hi-fi!

OK, as far as this upsampling melarky is concerned, well it sounds very 'nice' and 'listenable to', doesn't it? But even though it is now properly implemented (well done Peter), I still believe it robs the music of dynamics, transients and presence. Yes, no upsampling is definitely more 'in your face', but I prefer that to 'laid-back'.

That's my subjective opinion, though I'm probably heavily biased, having never liked upsampling on my P70 transport either.

Now, objectlively, I have even more issues with it. As I've mentioned before, aliasing has a definite effect below 22.05KHz.

Sorry, but I couldn't help myself - have a look at the following graphs.

Would love to hear your subjective views on the sound though - please feel free to tear me down.

Mani.

For those of you not familiar with aliasing, any signal above 22.05KHz is not 'real'. There is no information above this frequency (known as the Nyquist frequency) on a 'red book' CD.

Any signal above 22.05KHz is simply a mirror reflection of the signal below 22.05KHz. The problem is that it tends to interfere with the signal below 22.05KHz, if an anti-aliasing (AA) filter is not used. This leads, in my opinion, to a 'laid-back' though very pleasant sound.

Don't get me wrong, there are some very good reasons why you would want to avoid using an AA, but ultimately, it's all a trade-off.

IMO, the only real solution is getting hold of some high-resolution files... though with XXHighEnd sounding so good set to 16/44.1, I'm not sure if I'm going to bother for a while :)

Mani.

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

Yes, selecting 'DAC is 16 bits 44.1 KHz' is what I do and what I want to say.

To me, this gives the best SQ for 16/44.1 files. I haven't done extensive listening, but so far, this seems to make an obvious and consistent difference. I will admit though that I have only really compared it to 'DAC is 32 bits 192.0 KHz'. I have not listened to any other bit depths.

For hirez FLACs, I select to 'DAC is 32 bits 192.0 KHz'.

Would love to know what you believe might be causing what I'm hearing.

Mani.

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

'DAC is DAC needs' only gives the possibilitys of the DAC in XX isn't it?
When no upsampling is choosing and playing 16/44.1, it doesn't change anything to the output when choosen 'DAC is DAC needs' = 16/192.

My experience with upsampling is that the advantage of upsampling is also related to the noisefloor coming from the powersource wich feeds the equipment and the sensivity of the speakers. With high sensivity speakers and clean powersource, i prefer 192khz upsampling. With lower sensitive speakers (<90db) en bad powersource Non oversampling sounds more dynamic, more 'raw'.

1) No. At the moment, I'm using either my passive vol. control with my Genelecs or the vol. control built into the Stax driver unit. XX vol slider remains at -0dB in both cases.
2) More difficult. It just seems more 'alive'. I tend to tap my feet more... if that helps. I'm happy to do some more serious listening... when I have more time (maybe over the weekend).

I've never used the XX vol control. Let me know if I should try playing with this...

Meanwhile, why does selecting 16/44.1 sound different to selecting 32/192 when the vol slider remains at -0dB?

Mani.

Well I am sofar pro-upsampling. I listened to it with exactly the same system as I use for 16/44 and I hear more. And don't find it giving more fatigue so far.
Wasn't this perhaps the longest time that you ever listened to the Fireface as a DAC Peter ? So in your situation I think it is remarkable that Twindac + digital volume is even challenged by Fireface plus passive, when using upsampling.

I went between the sampling rates quite a few times and for me the preferred sound was with quad. I will do some more listening tonight. I don't mind that at all.

And Mani how are the 'gonio's' with the version 6 I was wondering.

The genios are perfect with 0.9u-6 - the correlation remains at +1.0 througout... as it should :)

Upsampling/aliasing has no effect. And of course, it shouldn't - aliasing won't introduce any phase differences between L and R channels... just within each channel and hence the interference above 5KHz or so.

You know, this discussion reminds me soooo much of a discussion I used to have with a 'hi-fi nutter' friend of mine. He has the full monty dCS digital rig. I used to have a Sony SCD-1 and Marantz SA-1 - I jumped on the SACD/DSD band-wagon as soon as I saw it.

But DSD never seemed to be right to me in the top end. So much like I'm hearing here with upsampling (especially quad). You probably all know that DSD has less resolution than CD above 8KHz. What do you think that does to transient edges... even of something like a bass guitar?

And here, quad upsampling (without AA) is down some 6-7 dB at 20 Khz...

Mani.

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


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).

"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 :yes: 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 8)).

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 ...  (http://www.phasure.com/index.php?topic=372.0)).

Thanks Peter.

Just to make sure I understand correctly:

1) DAC is 16 bits 44.1 KHz -> data is pre-processed
2) DAC is 32 bits 192.0 KHz (or any bit depth > 16) -> data is processes in real-time

And this is what is influencing the sound, right?

The next improvement will be to pre-process data for all bit depths, correct?

Mani.

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 ... :grazy:


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. :yes:

All correct Mani. Including the last one. :grazy:
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 ... :swoon:
(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).

I'm in the same camp as Mani on not using the upsampling option. When activated, the precision looses a little something. Mellower and not as articulate.

As a typical test track, I refer to "In The Hollow" from the album "Asian Roots" by TakeDake with Neptune. For those who have this album, listen to the shakers just left of centre stage at the beginning of that track. In normal playback there is a crispness to the sounds of the contents of the shaker as it gets moved backwards and forwards. Go with the upsampling and the sound loses it's clarity a tad, almost as if they changed the material within the shaker itself. I could quote other examples, but I spent several hours this afternoon switching between non upsampled & upsampled and my preference was the same each time, regardless of music type.

Nevertheless, it's a good option to have within XXHE for those who have a preference to use it. I daresay different DAC's and equipment paths may well sound better with it activated.

Cheers all, and continued thanks to Peter for the effort being put into XXHE's development,

Russ

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 ...

Hi Peter,

I'm assuming c)

In any event, here are the results with Foobar/ASIO (running 24/192 shared mode). There is obviously an AA filter being applied.

Mani.

I think it's time to brush the dust off those old undergrad books on our shelves... or at least on my shelf.

Mani.

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 ...

It's my lunch break, so I can do this quickly... with pleasure.

Find below the two curves. Just be aware that I'm using my system 2) at the moment - i.e. no external dac and no spdif.

I have no idea what Foobar's SRC does with the bit depth, though in shared mode, Vista pads it to 24 bits, no? (I have it set to 24/192).

Mani.

Thanks Russ. I was beginning to think I'd lost it for a minute...

I totally concur with everything you say.

The only thing that doesn't make sense to me is this: why does quad upsampling sound as good as it actually does. No, IMO, it doesn't sound as good as no upsampling... but it still sounds pretty damn good. If you read the literature on aliasing, it is supposed to sound awful, even in moderate amounts...

Mani.

I don't get this. Upsampling means more precise reproduction of the original waveform. That precision looses a little with Upsampling cannot be true in theory. When you hear the opposite effect, there must be another reason.         

My limited understanding of upsampling is that it has a single potential benefit: it allows the use of a less-steep anti-aliasing filter to be used. This is turn reduces ringing effects in the audible range.

I do not understand how upsampling can create more information than is recorded onto the CD. The only technique that I'm aware of that helps with increasing resolution is dithering...

I kind of remember Bob Katz and others doing some tests on this. They found that in an ideal world, an anti-aliasing filter would have a low pass band starting no lower than 50 KHz or so. In which case double, but especially quad upsampling might make sense.

But what's happening here is that we are getting a load of (I assume) nasty digital artifacts above 22.05KHz, interfering with our beloved music. This cannot be good. But I don't understand why it doesn't sound worse than it actually does...

... maybe my hearing isn't good enough to hear the mess that going on up there :sad:

Mani.

Can it be so that the audible effect of artifacts and distortion with NOS (without filtering) is heared and interpretated as more details? In other words; that what we think to hear is not what we really hear.
 

Do you really want to upset the NOS crowd???

Mani.

 ;) no!, because i also like to listen to my diy (Monica) NOS dac. Sounds very nice. But....

My Stello 220mkII Dac is connected to 100% clean power (Kemp powerstation).
The Stello can be easy switched from NOS to OS.
The audible differences I hear are more in agreement with the theory, wich i suppose as the truth;
OS (192khz):  More subtile, a little bit less dynamics,  much more microdetails in sense of dying-out sound wich makes the sound image more complete and enjoyable . 
NOS:   A bit raw, more dynamic sound.  Very good musicality but ‘not complete’


Moderator's note for future reference :
The "NOS" referred to above, is unrelated to the official "Non OverSampling" phenomenon. The Stello 220MKII is a Delta-Sigma Oversampling DAC.

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.

One other thing to think about :

Since this is a topic that makes me brabble around anyway, take an MP3 if you dare :blob8: (take the higher bitrates though).
So far I did not spend much attention about it (too much :blush1::blush1::blush1:), 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 ...

Hi Peter,

Do you know the site src.infinitewave.ca?
They show interesting testpictures of a lot of upsampling algorithms, most of them look very bad.
Even the ones that are reported by some to sound good look awfull.
I was hoping you could make one that sounds good and looks good, but that must be a very difficult challenge!
Maybe you can do the same tests they did, or find some clue what the competition is doing.

Well this topic is about sound quality..  :)

I agree that 'what sounds good' not always has to be a 100% agreement to theory. 
If it was, our hobby would have ended after Philips made his first cd-player  ;) :).
But 'impressive' mostly gets boring through the time. Although, i think that kind of phenomenons keeps the highend market alive.
Don't you agree that its more about the natural level of sound with stable precision in more complex parts of music (like orchestra's).
It is very hard to achieve real natural sound. 1 'wrong' capacitor in a circuit can screw things up.
Mp3's may sound impressive, but will i.m.o never fit in the framework of highend.

An evening of comparing quad and 'uno' SQ keeps my judgment as before. More to listen to, no lack of transients, no fatigue, new things to hear on well known music with the quad upsampling. Whether this joy and judgment are caused by the nonoversampling Altman  DAC in close cooperation with XX or by Dr Placebo I don't know. I am happy withit anyway.
And while I have my fun, Peter is working on his next improvement for the oversamplers of the world. Great division of labour :)

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.

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 :grazy:).

Ok, have been playing with version 6 for some hours here now. Preliminary results, in MY system, with MY DAC.

Bass goes deeper with more impact down low (sub-regions) when I use both upsamp and double together, xx is set for 32/96 DAC. More physical bass deep down. In this case I also set my soundcards S/PDif to 88Khz for best "impact". Sound is clear, detailed and separated around each instrument, maybe too clear and detailed? Hmm. Difficult to say, but the better bass complements it well. Very wide soundstage, don't stop until it reaches my sidewalls.

Same settings but with the sound card set to 96Khz gives a kind of "ringing" in the high frequencies, a lot of SSSS-sounds in the treble etc. Not good.

Setting the SC to 44Khz, XX to 16/44 for DAC and turning off double and upsamp, gives a somewhat "kinder" and "round" sound, a little more "warm", not so much impact in the real deep bass, and a somewhat "simpler" image, not so much air on the stage. The bass impact seems to have moved to around 50-100hz up from deep down. The image seems a little more truncated against the middle of the stage and is limited sideways to the physical location of my speakers.

All of this is with different 16/44 material but using Fourplay-Journey-Play Around It for some detail comparisons more than other tunes.
 

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 ?

Feel free to copy me!  :)

Hey, it's not only me who's up at 03:30 in the morning! Cool...  :prankster:

The 96Khz setting for my SC: When set to 96Khz and using double/upsampl, the DAC shows 96Khz when playing back 16/44 material. What happens is that 44 material is played back at 48 or 96hz depending on double or not. And you're right: it doesn't sound good, especially in the high freq.

But I also have better SQ steering the SC to 88Khz setting instead of letting XX upping from 44 to 88 automatically when using double/upsampl. God figure why...some of the "impact" gets lost if XX does it automatically.

Btw., not hearing much difference between only double or together with upsample. Earlier versions had more difference on this.

Ok, just wanted to say that as of now, with the SQ I have from XX and the promises from Peter about even better SQ in the future, I will NEVER buy another discplayer. End of story.

Of course, I forsee a lot of possibilities for better SQ as of this version (6) but we're definitely on the right track.

 :clapping: :clapping:

Anybody who's hearing a balance shift to the left side, e.g. less volum/level at the right side? Or is it my slight nosecold who's the culprit?

I went to bed afterall. You aparantly did not.


You mean : when you check the Double/Upsample checkbox, right ?


How much did you :party: by this time of night ?

:drinks:

Hi y'all,

I've just got back from a really hard day's work and thought I'd relax to a bit of music... through XX, of course.

I started with Claire Martin's 'Too Damn Hot!' album on 24/96 FLAC files. I couldn't stop listening - it just sounds amazing through XX...

I then compared this to my 16/44.1 wav files of the same album.

There is absolutely no question that no upsampling sounds closer to the 24/96 FLAC files than double or quad upsampling. In fact, the non-upsampled files are remarkably close to the FLACs. However, the 24/96 FLACs sound more musical - in short, they sound simply stunning.

... BUT...

... I'm not necessarily pointing the finger at upsampling.

I am now convinced that selecting 'DAC is 16 bits 44.1 KHz' is making the difference.

As Peter has already explained, in this mode the data is pre-processed.

My hypothesis is that for 24/96 FLAC files, the data is also pre-processed. The sound is simply too similar to 16/44.1 wav files with 'DAC is 16 bits 44.1 KHz'.

Peter, am I right?

If so, I think all of you upsampling guys are going to have a treat when Peter forces this to be pre-processed also.

Mani.

Shoot Mani ... :rofl::rofl::rofl:


You're just Gold ...
At first I thought you couldn't be right, because at this moment I couldn't see how. Had my answers ready to justify your thoughts anyway, poored a special whiskey, and checked to be certain. Below snippet is from the preprocessing part ...

It actually says that the only thing it takes for the actual playback is adding the last byte (totalling to 32 bits) which needs no further processing, and FYI looks the same as something that needs no processing at all.
I really can't say how much I like someone who actually dares to state as you did at this super micro detail stuff, *and* is correct at the same time. Cudo's, hats off, anything. :love: :love: (can't recall I ever used this :)).

Coming from this, I would like to say :

If there's someone in this world doubting that stupid software can make a difference to sound, might it be Queen Mary, O. Bin Laden, mr. G. etc. Bush, Bill Gates or anyone else thinking of ruling some parts of this world, let them read this, let them read into all the combinations possible here, and let them guess something which was just proven right in this occasion.
I know, my english is far from expressing what I want to make clear, and I sure hope it comes to you all as how I intend it.
And I wish to go further even :

Please allow me to quote you from the place we have met :


It is not more than two weeks ago - and after you were slammed a bit as usual on the Internet - that following from this I tried to explain how to listen and what to listen for. I personally think it is more than quite amazing how you could evolve to, well, someTHING like this. But don't say I promised ... like in


So, apart from the valuable contributions from your hands (like the plots) now you can come up with this ? I really can't comprehend ...
I know from myself that the far most important on these kind of things is the, say, absolute memory which allows you to relate things, or better, make them relative to the reference (from that moment) I talked about in the quote. In fact this IMHO is about the ever so much importance of being able to listen through things, which comes down to taking distance from feelings, getting into technical details for comparison, and only when that fails, start with tapping the feet more differences. 8)


I may sound rather stupid or wimpy-like in the majority of the above, but I just feel like expressing some feelings as an obligation to you Mani. I take it for granted when I'm a woman from now on. Can't be that bad anway. :)

Peter

Nah, I wish - but not a good cognac in sight...maybe intoxicated on music, had this "just one more CD" train going through XX - not hitting the sack until 6.00...yuck...tired today.

Peter and Mani: Can't wait to hear the pre-processed upsampling routines! Maybe more of the "warm" and "more organic flow" will come forth, but my latest listening also confirms a pretty nice sound p.t.

Hey guys, maybe this latest combination of measurements of the digital outputs combined with listening is the right way forward. As you said Peter, the amount of code starts to get big with only listening as an correction method.

 :prankster:  :grazy:  :good:

I am lost here.
Are you guys going away from bit perfect with your pre-processing data?
Is that is what is going on. Your posts are very confusing!!!

I'm a little confused too Andrey.

I have a question for Mani and LydMekk - are you saying you play 24/96 material with 'DAC is 16 bits 44.1KHz'? And/or are you saying you are playing 16/44.1 material and are checking double but leaving at 'DAC is 16 bits 44.1KHz'?


LydMekk - I still don't understand what you mean by this. Are you saying you do not check double? What do you have your 'DAC is' set to?

Did I miss something? Peter, is it possible to play 16/44.1 (without checking double/upsample) when your DAC is expecting 24/88.2 or 24/96? What would the benefit of that be?

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.

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".

Sorry, I don't have much time right to reply right now...


I'm not sure how many whiskeys you had before you checked the code, but could you double check?... actually make that quad check :)

Mani.

some input to the upsampling discussion

http://www.audioasylum.com/forums/pcaudio/messages/2/29646.html

this guy gets attacked directly on AA so there must be something good in his approach :)

so far I have found that Islay whiskies agree best with quad upsampling. The jury is still out on the best whisky for double upsampling

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 ...

Thank you Peter,

thinking that we have this ability to listen through all the things that still need to be improved and allready get so much fun and happiness from the music that we play. And with all the non-solved issues around clearly have this ability to make the distinction when improvements have been made (at least quite a few people seem to be able to do that)

have a nice day,

Leo

Yep folks. To try to explain a little clearer: It's the difference between what's my soundcard is putting out (f.ex. 88.2) and what XX is trying to play back (f.ex. 44Khz with or without double etc.).

Problems occur (as mentioned) when I locks the soundcard at f.ex. 96Khz when playing back 44Khz material either normal or with double or upsampling. Then something ugly happens up in the tweeter-region.

And in my case it's also better to lock the soundcard at 88Khz instead of letting XX handle the auto-upgrade to 88 when selecting double or upsamp. It plain sounds better. Don't know why.

Clearer?

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)).

Just to be clear, these are my settings:

24/96 FLAC: 'DAC is 32 bits 192KHz'
Double/Quad Upsampled: 'DAC is 32 bits 192KHz'
Non-upsampled: 'DAC is 16 bits 44.1KHz'

'DAC is 16 bits 44.1KHz' does not allow the playback of 24/96 FLAC or double/quad upsampled 16/44.1 files.

Mani.

Same as Mani.

Have only played 16/44 material though, normal .WAV files, no FLAC or other formats.
Standard CDs ripped by EAC, slow mode (type 2.3X read speed), not hysterical or what EAC calls it.

New AK4397 32 bit, 196kHz DAC. Phase-perfect digital filter, fully differential analog out without the need for coupling caps, digital level control (32 bits, remeber). Direct interface to 32-bit path outputs from DSP's. ;)

Hi Peter,

Firstly, thanks for your kind words earlier in this thread. I really appreciate that. I just hope something worthwhile will come of any new insights you've had...


For me, this is priceless advice. The problem, of course, is in finding a reliable reference... and then being able to compare 'apples with apples' once you have it (what with so many different factors at play).

Can't wait for 0.9u-7...

Mani.

Me neither...

My experience :

Once you can discern technical aspects, like the tout bass, and you are able to keep it in memory once you heard it from your system, it can be layed aside for later. As I said "from your system" is important, because your system can (no, will) unveil those separated technicalities, and all have one best setting : your reference for that part.
All you need to do now is unite those separate best references into one system.
Recognizing what causes what is obviously very helpful in achieving this.