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Author Topic: 0.9y-4 - w00t  (Read 125179 times)
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Fidelio
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« on: October 06, 2009, 09:46:41 am »

Sorry, but fucking hell Peter clapping

y-4 sounds spectacular. Using XXHE is like getting major free upgrade to your CDP every other week.

Two things that have bettered drastically: timing and bass, which are obviously connected. The bass is supertight and very present and melodic. Toe-tapping is greatly increased. Moreover, individual instruments are more distinct, but seem to be playing better together. Well done sounds good !

Running Win 7 x64, 24/44.1, no upsampling, 14/20/20/0/0 (works with this version) and appointment scheme 3.
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« Reply #1 on: October 06, 2009, 04:31:03 pm »

Arc upsampling is in a word... "Amazing!"

In between loads of international work travel, a house refurbishment/move and... oh, a new baby(!), I haven't had much time to listen to music these last few weeks. But I just had to try 0.9y-4. And then I just had to write something here. But I only have time for one word right now. So, once again, "Amazing!"

Mani.

PS. I believe my DAC is non-oversampling at 4fs.
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Leo
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« Reply #2 on: October 06, 2009, 05:08:07 pm »

Where will this end ???

sitting spellbound listening to Sergeant Pepper (remastered) an album I must have heart a couple of hundred times
doubling works and is wonderfull , 4 times is even better I think. But that is vulnerable in my setup as I RME gets in Stampede mode and gives the music also in 4 times the original speed. Sometimes that stops again.
Have to stop now, want to listen more more more
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« Reply #3 on: October 06, 2009, 05:10:39 pm »

 unhappy

Dreaming about a 24 bit DAC.... aggressive

 Happy

(ps Have got my collection sorted out! Works fine  ok)
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« Reply #4 on: October 06, 2009, 06:00:20 pm »

PS. I believe my DAC is non-oversampling at 4fs.

Well, that is the idea really. So, for those who can reason out that their DAC may be upsampling, but not further than it's capapility for input, you'd be receiving the same as how it is intended, and I think this is more than spectaculair.
Even if you can do 24/96 only (or 18/96 which is enough to let it work), it already does its job (removing harmonic distortion *and* upsample the GOOD way).

For those who also have the possibility to shut off all the filtering (besides upsampling, which is also a means of "filtering" as such), do that. But, YMMV here, and with filtering (as long as this is not upsampling) may work out even better, as long as it is an analogue filter. This removes the images beyond the frequency the DAC outputs, which I (Arc Prediction) don't touch. This latter is deliberate, because digital filtering can't remove those images anyway. So, it is just a matter of your amplifiers being able to cope with those high frequencies (stay linear). If you're a real NOS user and you were satisfied before, this should bring extasy ...

Peter


PS: I still wonder what this may bring to OS (Oversampling) users (like in sigma-delta). In that case : don't get fooled by "different is *thus* better". It should bring something which you never heard, that realistically. But I don't think it can happen with OS. nea
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« Reply #5 on: October 06, 2009, 07:49:17 pm »

For me it is 176 as I solved the Stampede error mentioned before. In the RME Fireface software I had to untick DDS active. Now it is very very nice sound and no funny fast music bits anymore.
HA !
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« Reply #6 on: October 06, 2009, 08:14:37 pm »

Hey Leo,

For me it is 176 as I solved the Stampede error mentioned before. In the RME Fireface software I had to untick DDS active. Now it is very very nice sound and no funny fast music bits anymore.
HA !

On a side note though (and I never quite understood it myself), if you tick DDS to be active, but also tell the Frequency Muliplier to x 4, the Fireface *must* be mastering, because you can change the speed now with the coarse and fine slider if you tick those "Active" checkboxes at the bottom (this even is so with SPDIF passthrough and another DAC connected to the Fireface). This might bring you better sound again.

Also, be careful : With your Altmann DAC I think you are fooling the system by telling it is more than 16 bits (because it isn't, right ?). I am not sure what anomalies come from that, but it should be allright ...

Peter


* FirefaceDDS01.png (10.6 KB, 409x528 - viewed 1562 times.)
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A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
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« Reply #7 on: October 06, 2009, 10:19:48 pm »

This is the biggest leap forward in SQ that I can remember.  Everything in my system has just taken a major step toward 'real'.  Articulation I've never heard before.  Distortions which I didn't know were there before, well, this version shows that they were there and they are now gone.  Bass is unbelievably real in a way I've never heard my system do ... real growling.  Gone are cymbals sounding like escaping steam (I mean this has always been getting better with XXHE versions),,, but now, 9y-4/arc prediction we have real metal being struck.  I am just floored and thank you for having this released during a vacation period... or I'd be hung over for work from no sleep last night.  Going to start listening again today... the music room is an addiction (if I believed in such things) and Peter/XXHE is my dealer.
Thank goodness my Stello dac allows for bypassing its upsampling (which always sounded like cr*p) and lets me input up to 24/96 with no dac manipulation ... someday I'd like to hear what quad/arc would do, but for now, jezlooeez, double/arc is simply amazing.
This is a huge deal.
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« Reply #8 on: October 06, 2009, 10:20:07 pm »

OMG,

Peter you did it again,

It sounds as a whole now, all sharpness is gone.
bass response a great, soundstage very big.
And really dry super-dry, dry is GOOD!
More sounds to discover, much more.

I have to `learn´ an album again (this is strange)

It sounds like I've put a wordclock in the DAC.

Petje af !
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« Reply #9 on: October 06, 2009, 11:33:11 pm »

Nice gadget that album counter Wink

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« Reply #10 on: October 07, 2009, 12:41:22 am »

I've just been listening to Keith Jarrett's Koln Concert. Wow!

I've never heard the ambience of the concert hall like this before. And for me, this is the key difference between 'Arc Prediction' and 'no upsampling'. The latter gives the impression that everything has been recorded in an anechoic chamber. With the former however, you can hear the natural reverberation and decay of the hall. Stunning!

So, where exactly has this extra information come from???

Also Peter, when are you treating us to a 'real' CD ripper? When you do, I might seriously give up on hi-res material and stick to 16/44.1 for a while...

Mani.
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« Reply #11 on: October 07, 2009, 04:11:36 am »

Wow P, nice sound with the new ARC pred.

Reminds me of the old days when a couple of XX versions where upsampling was better than vanilla.

Bass, macrodynamics etc. are very good in this version. Ambience is very nice.

Overall one of the best versions over the last year, P. Extra nice IMO because, frankly, I have been a little disappointed over the last versions and had started looking at other players as I felt we were on a wrong track SQ wize lately. Safe to say I've been hauled back in...

Btw., sounds like ARC isnt delivering when running in attended mode so un-attended is req. for getting the full new sound IMO. Sounds like its only doubling in attended, not upsampling.

Therefore, I have startet to run all in un-attended!  Cool

Please, hope further versions will uphold the same SQ or I for one will revert to this version, it's that good.
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« Reply #12 on: October 07, 2009, 09:02:30 am »

Btw., sounds like ARC isnt delivering when running in attended mode so un-attended is req. for getting the full new sound IMO. Sounds like its only doubling in attended, not upsampling.

Uh no, Arc Prediction is definitely working in attended mode, else I'm hearing things!

I'll get around to trying unattended one of these days...

Mani.
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« Reply #13 on: October 07, 2009, 10:20:01 am »

Before getting back on some of the responses : There is one measure really : your wife not complaining for 5 weeks in a row about any single track being too loud, the music being no music, or whatever it is I just might agree upon.

Happy
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
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« Reply #14 on: October 07, 2009, 01:23:02 pm »

[Sorry for a long post, probably full of typos ...]

I've just been listening to Keith Jarrett's Koln Concert. Wow!

I've never heard the ambience of the concert hall like this before. And for me, this is the key difference between 'Arc Prediction' and 'no upsampling'. The latter gives the impression that everything has been recorded in an anechoic chamber. With the former however, you can hear the natural reverberation and decay of the hall. Stunning!

So, where exactly has this extra information come from???

I guess this needs one of those 3 "papers" I have written, and I guess some day I will have a nicely worked out (and understandable) version. But generally it comes down to so many things being wrong. Here's an excerpt of one of the versions of the paper. This starts with a picture (not shown now), and explains about how wrong things actually are :


This is just the analogue wave as it will be output (from the player) to the DAC. But hey, this looks nothing like a sine ! And indeed it does not, and this is because when the samples were taken (think from left to right on the x-axis = the time domain) the sample points did not meet the peaks of the sine, but instead they were taken somewhere at the slope of the sine. You can see that occasionally the peaks were taken (look at the maximum values), but this is more often not the case.

Now, try to imagine that the max peaks as you see them are at full scale, and that those you see just a little under already are twice as soft. Notice that yo must think "dB" here, and a little under as you see it here is approx. 6dB, or twice as soft. If you look in the area of sample number 79 you can see that only half of the normal full scale is reached, or IOW this is at -48dB. This will give as good as no sound to your ears.
If we think decimal, that -48dB is at half of the full scale which is half of 32768 (for either plus or minus side), which means an error of decimal 16384 on to 32768 which of course is 50%.

This is the job : Turn that error of 16384 into an error of 0, and see that there are 16384 possible steps of improvement !
And so you see, while we may have thought digital is quite OK, it is actually the most wrong, with the too low sample rate as the cause. And thus what (software) engineers strive for, is getting that analogue wave back to normal.



So what I am telling here, is that everything output to the DAC is so hugely wrong, that, well, it can be hugely improved. Of course, this is a normalish task of the DAC itself, but the means used (technical : sinx function) is even more wrong ...
... And it is here where the paper(s) start to end in needing much more attention at laying it all out (with proper graphing and comparisons), and where they thus all end, for now.

As I am always telling that so many things are wrong, likewise I am telling for a longer time now that official measuring and measurements is/are wrong. This in very brief now : it is no big deal to show a 0.0001 % harmonic distortion on a continuous single frequency (or even two frequency) wave created by the sincx function (call it "filtering" the DAC applies), just because it is such a continues same wave, while sincx loves that. It works by "averaging" (*very* roughly said !), and averaging an ever same wave will do no harm at all -> 1+1+1+1+1+1 / 6 stays 1 = the original. But what if it's 3 + 1 + 7 + 2 + 1 + 2 / 6 and this 2.67 not being equal to the original(s). The average is correct allright, but each single input (meant to be output !) has been destroyed. Now, the first case is measurement, the second case is playing music, and sadly that can't be measured.

What it (again in brief) comes down to, is that NOS is way wrong, because that would measure (single frequency) something like 1+1+1+1+1+1 / 6 must be 1.2. And since 1.0 is 20% off to the expected 1.2, this is 20% harmonic distortion. This is measured like that, and it *is* true. What I do, however, is something like 1.4 + 1.1 + 1 + 1.3 + 1 + 1.4 / 6 = 1.2 and now it is conform the expectation, and thus 0.00001 (or 0.0 in this example) % HD.
The big difference with normal "filtering" is that I do this on a per sample base, while sincx filtering can only do it with a (theoretical) infinit number of samples. But say 100 samples, and get the grasp of now one 100 samples all being of on *another* way. It measured good for theories, but all the samples are all over the place, and actually this is more off than "just NOS" (and it is this "more off" for me to proove by numbers).

Back to the actual question "where exactly has this extra information come from???", I could say "so very wrong it was !". And actually this *is* the answer. But, the stupid thing is that this indeed brings a kind of information which is completely new to us audiophools. I must say, at working at the DAC similar things happened to me, but this is still in a quite different leage;

On a side note, and this is important to ever get there I thinnk, keep in mind that my ears seem to be good enough to know that NOS/Filterless sounds better than any type of OS/Filtered, no matter how hugely bas NOS/Filterless measures. And might you not know, this plainly comes down to even 40 THD+N at certain frequencies (this starts at just under 5KHz, and which is completely logical once you're into this all a bit). So, I sat down to *proove* why NOS still sounds better, and it is all about this. So :

In one of the papers I wrote more extensively about the types of smearing. With my brief explanation above, and the "averaging" type of filtering, it may come as obvious to you that this is an explicit smearing type. It is nothing less than spreading the information of 100 samples into the one sample currently output. Literal smearing that is. However, NOS/Filterless smears just the same, although the type of smearing is a completely different one. It "smears" the one original analoge piece of a wave (expressed in a snapshot sample) by its own harmonic distortion because the samples are at the wrong place. So, harmonic distortion smears, and looking at the longer period of the wave, a clean sine is captured (on the redbook CD) as not a clean sine at all. This is of course what we knew already, but it is the imagination of smearing which is important here;

Without creating 8 pages again, the net result of what Arc Prediction does is that "dryness" as I describe it myself. It is one of the first noticeable things, and dryness can also be described as shortness, the best perceived at bongo like instruments, because their attack is so short (in synthesizer terms : there's hardly an envelope, which is the shape of the attack).
While it is nice to perceive a bongo as a bongo, all becomes *really* apparent when you found yourself disturbed by one of the most uses instruments : a snare drum, and not being able to map your mind onto the snare drum used. I mean, if you have a drum kit yourself (like I bought one 2 years ago, just to compare reality with music reproduction), it is just that instrument that makes you wonder what's actually wrong, because it seems so simple. Today, and relative to how it was, you can just hear what was wrong : the hit on the skin smeared the snare. So, the hit on the skin (even more firmly spanned than a bongo) is now as short as reality, which a. gives it its color, and b. makes the snare sound as how it really is.

Once you got the above (I mean, listen whether I could be right), it becomes more easy to see how everything got smeared, and how the accidental hit of the rim of a tom now allows to be all over, while before it was smeared into the (hit on the skin of the) tom itself. Keep in mind, a hit on a rim is one of the shortest sounds available (attack is as short as can be, and decay is also as short as can be), and today it just has died out before the skin is it. Nothing less than reality, and caused by the shortness/dryness which now is "available".

Again once you perceive this indeed, you can go further in the "stage sounds". So, indeed it is true, and not by a small amount ! There's voices, placements of microphones, walking, laughing, but just the same there's the now all over hit of the (spanish) guitar cabinet during playing the strings. ALL sounds different just because these sounds now are there.
In the end you could say that all now sounds short enough not to get smeared into the nest sound.

The recap is simple :
a. no HD in the audible band;
b. no ringing.

Ad b.
I am sure this is the most important element of good music reproduction. All is a tradeoff between this kind of distortion and the harmonic distortion of NOS/Filterless. And (still assuming good ears), while NOS/Filterless takes prevalence over OS/Filtered, at the first measuaring near 40% distortion, we really may wonder how OS/Filterless measures once it can be done as should.

Lastly for now on this subject, and a real technical matter : Those who are really into this (signal processing) may state that reconstructing the analogue wave on a sample per sample base is not possible. Although at this moment I must see the first one telling *why* that is (which I by now can), the net result is very different, and explaining *that* requires a small book by itself. For later.


Quote
Also Peter, when are you treating us to a 'real' CD ripper? When you do, I might seriously give up on hi-res material and stick to 16/44.1 for a while...

As said elsewhere, 0.9y was created for just this. Thus far, however, I just didn't get to it, although the most of the base software already has been written (this must have been April 2009). With ripping as the base, and which might come down to merits of great importance again, I guess I fell from the one thing into the other, which -you may recall- started with measuring. And, this (detail of) measuring is obviously needed, while normal measurement means won't allow showing the differences (while we sure hear them !). So, to make this small subject consistent with the main subject here, let me tell this, which by now is completely consistent with the knowledge from today, which knowledge (on my side) sure was *not* there back at the time :

The measurement means I created (remember, a.o. needed to judge good/bad ripping) worked for NOS/Filterless. I could judge the data as in the file onto the result coming from the DAC. However, trying this with an OS DAC, and there was no head and tail to anything. Thus, the beautiful graphs I made (actually in XXHighEnd, but all disabled to you), showed completely unrecognizeable waves cmpared to the input waves. It is my luck that I first tried it with NOS, because that proved it could be done. But, and as a side note, already that shows how *again* everything is way way off to what's supposed to be output (which should be 1:1 without any filtering means). Anyway, the waves coming from OS could just not be recognized at all, compared to the input. I now know why, and I guess it just required more knowledge on my side. In the mean time, this is just so; what comes out can't be recognized from what goes in (both graphed). With this Arc Prediction it sure can, because the original samples stay in place. Note though that for OS this is not only about the original samples, but it is even about the "original waves". So, at the rough wave level all has changed. In the end what it comes down to, is that the solution for reconstructing a single frequency, doesn't count at all for complex waves (containing music). Ah, I said that before. Right, but this time my measurement already proves it. What comes out can't be recognized with what goes in. But still it bring you music ?? yes. And thus again : so much can be improved.

Long story short on the last question : I guess it needs quite some knowledge to "just in between the lines improve ripping". But I guess I'm almost ready for it.

One last thing for now and here, and related to the last sentence :

There is more going on than what I currently can reason out 100%;
This is related to hires material not sounding as good as this upsampled 16/44.1; I won't say that this is caused by all hires being upsampled (by the same wrong means) from 16/44.1, but something is going on with "native" e.g. 24/192 material which makes it hide those on stage sounds and everything. Personally I don't think we can end up at "but Arc Prediction is wrong afterall", just because it does so right for such a long time (5 weeks in my case). Notice the difference with "faked high detail", which is very easy to create by just removing some (lower) mid frequency output. Result : "Ah ! you can hear her spitting in the microphone now !!". Yeah, right. But this won't last for long, because it is "created" by removing something else first. It won't last for 5 weeks.
So, it intrigues me that it is just sheer high resolution I perceive from this Arc Upsampling, while native 24/192 (or 24/96) already shows explicit lower resolution. So, "as good" would already be strange (but according to my own applied theories it can be done), but how "worse" falls into place ? ...

Peter
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« Reply #15 on: October 07, 2009, 01:53:48 pm »

Wow P, nice sound with the new ARC pred.

Reminds me of the old days when a couple of XX versions where upsampling was better than vanilla.

Bass, macrodynamics etc. are very good in this version. Ambience is very nice.

Overall one of the best versions over the last year, P. Extra nice IMO because, frankly, I have been a little disappointed over the last versions and had started looking at other players as I felt we were on a wrong track SQ wize lately. Safe to say I've been hauled back in...

Yea, I know. Now try to imagine how tough this all is while bit perfect = bit perfect and everything can only workout the most indirectly, and hopefully the same for everyone !.

Quote
Btw., sounds like ARC isnt delivering when running in attended mode so un-attended is req. for getting the full new sound IMO. Sounds like its only doubling in attended, not upsampling.

Therefore, I have startet to run all in un-attended!  Cool

Please, hope further versions will uphold the same SQ or I for one will revert to this version, it's that good.


Well, the least I should do, is just stay out of that sound creating program. It may help ! Happy
Anyway, I guess you just proved how important Unattended is, because like Mani said, it definitely is active at Attended too. In general though, it is so that the "better" (or more detailed) everything is in Native XXEngine3 (= Unattended) the bigger the difference is with Attended (= XXHighEnd active) is.
So I assume you now know once and for all ! Cool
haha
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« Reply #16 on: October 07, 2009, 03:05:46 pm »

Now that I've had the chance to play all little with Arc Prediction, I'm not that sure I am all that convinced.

Some records, especially orchestral pieces, sound more neuanced, detailed and clear with less congestion with Quad/AP. However, all other music suffer from Quad/AP in that it ruins the timing and groove. Especially rock suffers I think. Double/AP is better than Quad/AP, but vanilla 24/44.1 rocks much more on my system.

AP makes me wanna change song whereas no AP makes me wanna dance.

Mind you, this is on my system, the whole computer sourced playback-thing is so tentative that I'm sure a different system will produce different results. Love y-4 tho, 14/20/20/0/0 with scheme 3 and 24/44.1 sounds jolly good, with tons of toe-tapping, dance-inducing PRaT.

Edit: Damn changed my mind about Double/AP. Will run it for a few days before I try vanilla again.
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« Reply #17 on: October 07, 2009, 03:54:44 pm »

Ok, thank you for sharing. But as I (I think) suggested in the other topic, you have an Oversampling/Filtering DAC, and in my book this Arc Upsampling can't do its job then; This would invalidate the filtering used in the DAC.

Note that the filtering in the DAC (which is part of the oversampling) is based upon a band limited signal (the band of 16/44.1 is limited to 22050Hz), and the so called sincx function is able to reconstruct the wave from that point of view. Now, in this case you are feeding the DAC with an already reconstructed wave, although to the DAC this will look like native 24/176.4 (for Quad). But, in practice this latter is not working (out) like that, because of the small variances Arc Prediction expresses compared to a native 176.4 (to 88.2) band limited signal. Too explain this in detail goes too far, but let's say that roughly it comes down to the very precise knowledge of how that band limited signal looks like and how the sincx function anticipates on that in detail. Maybe it's best to think of the equations I put for example in my earlier large post, and many of the 6 numbers leading to 1.2 for the average result. Twist two of those bease numbers (like 1.4 and 1.0) and the result for sincx will not be the same, because those base numbers have weights (like the further away the actual sample, the less it weighs in). Now think of I have already changed those numbers, but sincx doesn't know, and now the result of 1.2 becomes 1.3.

It still could have worked, but the longer I think about it, the more I think it just can't for OS Dacs. For OS users this is just a pitty.
But hey, if you are happy as can be without Arc Prediction, you just are. I sure don't want to change that !

Peter

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A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #18 on: October 07, 2009, 05:28:52 pm »

But my DAC isn't oversampling, is it? It just converts the digital stream that is presented to it to analogue. I don't understand what kind of DAC that would benefit from AP then?

If I feed it 44.1, it reads 44.1. If I feed it Quad/AP, it reads 176.4. Thus, the DAC isn't doing any oversampling, but XXHE is. Am I wrong in this?
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« Reply #19 on: October 07, 2009, 07:11:10 pm »

Arc Predicion 2x and especially 4x brings a significant SQ improvement. The goosebump effect is undeniable.
And I'm at headphones yet for today.

Tomorrow plain/4x arc -attended vs unattended.

After a quick check with q1/2 at 20/20 i think i still prefer those at 0, sounds more realistic to me (in headphones).
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PeterSt
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« Reply #20 on: October 07, 2009, 07:14:41 pm »

Quote
If I feed it 44.1, it reads 44.1. If I feed it Quad/AP, it reads 176.4. Thus, the DAC isn't doing any oversampling, but XXHE is. Am I wrong in this?

Fidelio, I am afraid so, yes. secret

What you see on the DAC (driver) side, is the *input* it (currently) takes. So, 176.4 at Quad upsampling by XXHighEnd. The envy24 chip your soundcard uses isn't a multi bit DAC, and thus it is an oversampling one (http://www.datasheet.org.uk/pdf/Datasheet-040/DSA00104728.pdf -> if I see it correctly, 44.1KHz x 512. Btw note this is a kind of a strange link; you must save it first before it opens as a pdf).

sorry
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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GerardA
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« Reply #21 on: October 07, 2009, 07:51:02 pm »

Well Peter,
I'm sorry but my oversampling DAC sounds definitively miraculous with Arc.

First I tried double without Arc, like before, and this sounded better, more defined, but voices had an electronic hardness.
So then I checked if Arc was on (no) and after I switched it on the sound is almost frighteningly good.
Bass is very strong, deep and rithmic. A lot of new details in the background, so yes where did this come from?
My dac uses a CS4398 with 120 times oversampling and in between is a CS8416 upsampler (88 to 176) but this ARC sounds
completely different from what the DAC can do in upsampling mode.

And the wife is very happy with the loud music, even singing and dancing!!
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« Reply #22 on: October 07, 2009, 08:17:13 pm »

roflrofl Well, I guess I am as happy with posts like yours Gerard(A), but it takes more brainpower for me to reason that it can work, than it does to reason that it can't. Unless ...

Quote
My dac uses a CS4398 with 120 times oversampling and in between is a CS8416 upsampler (88 to 176) but this ARC sounds
completely different from what the DAC can do in upsampling mode.

Unless you mean with this that you can switch off oversampling afterall (I would say No with your Terratec).

But if sufficiently more people with OS DACs come up with this, I am willing to force my head better into thinking of why.

Btw, in any case this won't invalidate Fidelio's perception on this, because anyone sensitive to "timing errors" is perfectly allowed to indeed perceive this all very differently. However, the very first thing I would dedicate someone like Fidelio, is NOT an OS DAC. If anything is wrong regarding timing, it would be OS (think like : the lower the frequency, the more it siples through to the higher frequency samples -> hard to explain). But maybe this is another kind of "timing".

If Fidelio stays alone (with his wife of course Happy), he should try to listen to this with an NOS DAC.
All is relative, you know.
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W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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« Reply #23 on: October 07, 2009, 08:34:50 pm »

Sill using arc prediction... I don't think I ever have tried just straight 44.1 with 9y-4, maybe I should, but I'm still blown away by what I'm hearing and spending way more time listening than I should.  Holy smokes the system can play so loud now with no strain.

My Stello dac has a two way toggle switch on the front that is labeled Upsample 192 & bypass.  I have never ever liked the 192 setting, I've tried many times, but always go back to bypass.  The stello has a 2704 chip in it, and I guess this means it is not a NOS dac, but it sounds amazing right now with double arc prediction, (dac is limited to 24/96, so can't do quad).

Since I'm using double/arc prediction on 44.1 material,,, why is it I can set the Fireface settings to 64 sample rate?  I thought I'd have to set it higher like I have to when listening to hires material.  With 24/96 material I have to set the FF settings to 128 sample rate to work.  I know double/arc prediction must be changing something because I can't play it at the normal FF settings of 48 sample like I do with no doubling 44.1 material.  Is it really doubling to 88.2 with arc prediction, but somehow I'm able to play it with only 64 sample rate?

Welp back to some listening ... I have Red Nichols on the machine right now... an old 1940s recording haha great great old man stuff.
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« Reply #24 on: October 07, 2009, 08:56:37 pm »

Quote
Unless you mean with this that you can switch off oversampling afterall (I would say No with your Terratec).

The CS8420 can do upsampling to 176/192 or just pass through. I can remove the little board and then there is no upsampling, but in the CS4398 there is the regular Sigma/Delta oversampling.
Luckily I don't use the terratec anymore, I think this was changing the output of it's SPDIF to much.
Now I use an extra outputplate with a connector to the motherboard. (Coaxial sounds good now).

I just found out 192 is working too thanks to another thread here!
Don't hear much difference/improvement between double and quad yet...
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« Reply #25 on: October 07, 2009, 09:00:29 pm »

If Fidelio stays alone (with his wife of course Happy), he should try to listen to this with an NOS DAC.

My wife loves listening, she plays bass in a nice punk rock band so she is cool with loud music, and sensitive to SQ.

What DAC are we talking here? Can you give me an example of a NOS DAC that is nice?
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« Reply #26 on: October 07, 2009, 09:56:20 pm »


Since I'm using double/arc prediction on 44.1 material,,, why is it I can set the Fireface settings to 64 sample rate?  I thought I'd have to set it higher like I have to when listening to hires material.  With 24/96 material I have to set the FF settings to 128 sample rate to work.  I know double/arc prediction must be changing something because I can't play it at the normal FF settings of 48 sample like I do with no doubling 44.1 material.  Is it really doubling to 88.2 with arc prediction, but somehow I'm able to play it with only 64 sample rate?

Welp back to some listening ... I have Red Nichols on the machine right now... an old 1940s recording haha great great old man stuff.

Well, it's weird to me, but something has changed as regards to FF 800 settings sample rate... now I can choose 64 samples on the FF and I can play 24/96 material... I used to have to set it to 128 samples to play, so it's not just double/arc prediction.  Tried this with 9y-3a and it worked there too... so don't know how long ago this change occurred. oh well, happy because from what I've heard, lower sample rate is better than high.
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0.9z-8-3a WAV/CUE files on HDDs via MB FW400>; Win7 pro ttp://www.phasure.com/index.php?topic=352.msg4021#msg4021); [XXHighEnd player  Qs 7, 0, 0, 0, 0; eng 4; adaptive; scheme#3; player priority low; thread priority realtime; clock res 5ms: SFS 420 Wink dac is 24/192 w/32bits; Play Unattended; Stop Services ticked; Wallpaper & Show Back ticked - Mirror Image unticked; Start Engine unticked;garbage collect ticked; copy files to XX-drive; *quad arc prediction upsampling*: straight contiguous:>PCI FW800 card>Fireface 800 DAC [latency 2048 samples for 176.4]; usb/ethernet/mb audio shut off @ MB
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« Reply #27 on: October 07, 2009, 11:13:33 pm »

Quote
happy because from what I've heard, lower sample rate

Uhm ?

Furthermore, I don't know. The only thing I know is that IIRC 64 always worked for me at 24/96 but I always set it to 96 for safety. At this moment I don't know, because I don't have the FF connected to the music PC anymore.
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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« Reply #28 on: October 07, 2009, 11:17:05 pm »

Sill using arc prediction... I don't think I ever have tried just straight 44.1 with 9y-4, maybe I should, but I'm still blown away by what I'm hearing and spending way more time listening than I should.  Holy smokes the system can play so loud now with no strain.

Just for fun (for everyone) ... try to focus on S'ses. How loud do they sound now ? Compare with pronouncing one yourself at the listening level.

They sound quite loud now, right ? So, much louder than before, and no harshness. In fact, quite similar to the one you pronounce yourself ...

yes
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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« Reply #29 on: October 07, 2009, 11:41:49 pm »

Si Si señor!! good good good
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« Reply #30 on: October 07, 2009, 11:53:36 pm »

Just curious: Bitperfect, first 16 bits?

Does Arc calculate the extra bits from 16 to 24/32 or are they padded with zeroes.
Because only then it makes sense that double can do so much...
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« Reply #31 on: October 08, 2009, 06:33:25 am »

Quote
happy because from what I've heard, lower sample rate

Uhm ?

Furthermore, I don't know. The only thing I know is that IIRC 64 always worked for me at 24/96 but I always set it to 96 for safety. At this moment I don't know, because I don't have the FF connected to the music PC anymore.

Safety?! yikes  Anyway, I said lower was better because when I've left it at 128 sample (after listening to 24/96 and forgetting to set it back) and listening to regular 44.1 in the past it did not sound as good as when set back to 48 samples ... so, just figured lower sample rate was better.  Anyway, running it at 64 samples seems to work well and FF settings says it's a 88.2 signal it's sending to dac... so, alls well so far, nothing falling apart.

Quote
If I've understood peter right, the Qs interact with the dac you have ... so everyone may be on their own with Q settings?

...I think (think ! people preferring Q2/3/4/5 = 0 (not Q1 !) have the ability to sort out distortion better, opposed to people who like the non-zero settings...

Did you just accuse me of preferring to listen to music with distortion over cleaner SQ?  Thems fighting words. hehe Anyway I've stuck with 4 26 26 0 0 all day today and I hear no reason to switch; to me it's sounding very realistic, and I use the term realism because the changes I'm hearing with 9y-4 are just so vast and different from other versions.
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0.9z-8-3a WAV/CUE files on HDDs via MB FW400>; Win7 pro ttp://www.phasure.com/index.php?topic=352.msg4021#msg4021); [XXHighEnd player  Qs 7, 0, 0, 0, 0; eng 4; adaptive; scheme#3; player priority low; thread priority realtime; clock res 5ms: SFS 420 Wink dac is 24/192 w/32bits; Play Unattended; Stop Services ticked; Wallpaper & Show Back ticked - Mirror Image unticked; Start Engine unticked;garbage collect ticked; copy files to XX-drive; *quad arc prediction upsampling*: straight contiguous:>PCI FW800 card>Fireface 800 DAC [latency 2048 samples for 176.4]; usb/ethernet/mb audio shut off @ MB
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« Reply #32 on: October 08, 2009, 07:42:24 am »

[oh, found this this morning. Forgot to post it yesterday. Sorry.]

If Fidelio stays alone (with his wife of course Happy), he should try to listen to this with an NOS DAC.

My wife loves listening, she plays bass in a nice punk rock band so she is cool with loud music, and sensitive to SQ.

What DAC are we talking here? Can you give me an example of a NOS DAC that is nice?

For this Arc Upsampling it shouldn't matter much, but to do really good it should be 24/192. 24/384 preferrable. swoon

The DAC Mani has may be in the leage, and otherwise I don't really know.
To do really well, these are the prerequisites :

- NOS
- Filterless
- 24/192

May you not know, my own DAC does comply (and this is in the end where I made the Arc Upsampling for), and it will be commercially available.
I don't want this to sound like a commercial though, and of course I made the Arc Upsampling available to you all in the first place. XXHighEnd is one, and Phasure NOS1 is two. Them together is what I just listened to for a couple of hours, and it may be just that other step into what we may judge as normal in a couple of years.

But what is normal these days ?
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« Reply #33 on: October 08, 2009, 11:26:38 am »

Woot indeed !

My immediate neighbours must really hate each new release as I'm sure they must get sick of listening to the collection of reference tracks I play in the evaluation process .. haha.

I think I was the second to download this but refrained from posting until I had a couple of days to fully comprehend what I am now hearing. This extended listening period also negates the effects from Mr. Placebo Happy

There have been a number of milestone releases in the development of XXHE during the last two years .... this is one of them!

I'm not sure I still fully understand what has altered, but it's noticeably for the better, and has reset, yet again, my level of acceptance for music reproduction.

There is a level of transient and dynamics apparent now that, upon reflection, simply wasn't there before or was masked. I was listening to some Flim & The BB's albums yesterday, at levels I rarely use, and when "Jazz Patrol" started up, from their "Further Adventures" album, I nearly jumped out of my skin when a sudden crash of cymbals sounded. Amazing stuff. Further listening just reinforced this new found clarity, and I found myself jumping from favourite album to favourite album, relishing in it all.

If I had an intravenous drip inserted I would still be in my chair listening, but I suppose one has to eat and drink sometimes .. hehe

I have XXHE now configued for Arc Prediction using Doubleing, and this is due to a technical restriction of my pre-processor ( Integra DHC 9.9 ). If I feed it a 176.4 signal the pre-pro disables the onboard parametric equalizer which I use for taming room modes. The DAC's in the pre-pro are Burr Brown PCM1796's , and are Oversampling in nature ( http://focus.ti.com/docs/prod/folders/print/pcm1796.html )

Even though the DAC's are OS, the signals being fed by XXHE are of significantly better character in nature, and that translates to an audible benefit that's quite apparent. I can only but wonder what something like the  NOS box that Peter has created would reveal.

Peter ... is this Arc Prediction your proprietary idea or something you found in a white paper somewhere and have adapted, as I've found no reference on the net about it at all ? Does the technique lend itself to specifying a set output characteristic such as 24bit/96Khz to fully exploit the capabilities of our DAC's, rather than, what I gather currently, is a integer double or quadrupling of the sample frequency ?

Back to my listening chair for now.

Well done Peter .. again Happy

Cheers,

Russ

BTW .. I'll get onto the bug report later Grin
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« Reply #34 on: October 08, 2009, 12:07:49 pm »

Just curious: Bitperfect, first 16 bits?

Does Arc calculate the extra bits from 16 to 24/32 or are they padded with zeroes.
Because only then it makes sense that double can do so much...

No, not bit perfect, because that can't exist with Upsampling. However, to stay in the "bit perfect" realm, the original samples are still there (which does not happen with OS/Filtering), and only the additionally needed samples (for the new sample rate) are injected.

As many bits are really used as the DAC permits. So if it's 20, then 20. If it's 24, then 24. However, calculation is done in "32 bit float", and after that decimated to the 24 bits (etc.), BUT which is done by the DAC itself, unless it is a 24 bit-only device, then it happens in the software.

Quote
Because only then it makes sense that double can do so much...

Not really. The bit depth matters for the accuracy, but it is really the sample rate and my means I perform the Upsampling, that makes this happen.

I hope it is clear a little (bit Happy).
Peter
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« Reply #35 on: October 08, 2009, 01:05:29 pm »

It is a little clearer now, I think.
So in theory it would also be possible without upsampling, just recalculating the extra bits so the shape of the wave gets more 'natural'? (leaving out the peaks between two samples of course).
Makes me think of making a picture by connecting the dots on the paper. 4 dots can become a rectangle, but if you get more dots it appears it is supposed to be a circle. You are then calculating the perfect circle.
No idea what sinx will do with that!
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« Reply #36 on: October 08, 2009, 01:34:06 pm »

The DAC Mani has may be in the leage, and otherwise I don't really know.

?
Quote
To do really well, these are the prerequisites :

- NOS
- Filterless
- 24/192

But no commercial DAC I have found boasts these parametres. Or am I wrong?
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« Reply #37 on: October 08, 2009, 02:28:51 pm »

But no commercial DAC I have found boasts these parametres. Or am I wrong?

Hence why so many of us wait with baited breath for the commercial release of the NOS1 DAC... me included... even though I might already have a DAC that is 'in the league'.

These are very exciting times!

Mani.
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« Reply #38 on: October 08, 2009, 02:48:56 pm »

Did you just accuse me of preferring to listen to music with distortion over cleaner SQ?  Thems fighting words. hehe Anyway I've stuck with 4 26 26 0 0 all day today and I hear no reason to switch; to me it's sounding very realistic, and I use the term realism because the changes I'm hearing with 9y-4 are just so vast and different from other versions.

No, but I did try to accuse you of using "sample rate" (the 64/96/128 subject) where it should be "buffer length in samples". I failed on that I guess.

On the distortion matter, I think this subject is more interesting and 5am deep so to say, than what's normally accepted. I mean, if you'd for example claim that NOS sounds better, any techie on the area would accuse you of liking distortion. He would even mean well, but would be sure you just like distortion (just not saying that you are completely deaf and can't recognize distortion). This is at the most rough level, and IMHO alredy completely untrue because where added distortion is one thing on the negative side, there's the all so wrong filtering on the other side which brings another type - and even more distortion (see longer post), but *that* our techie won't see, because his maths are correct and proven by the analyser (equipment).

What we talk about here (the Q2/Q3) is at such a super micro level that it can't be measured by known means (but my own software), and theoretically it should be so that 0/0/0/0 is the best. However, when we think about e.g. harshness, we all know how it can be solved : adding more distortion (read : filter it, smear it). Thus, liking something like 30/30/0/0 better, may indicate that something else is to be filtered. This is no crime, and when it works out for the better, why not. However, it is never to forget that when indeed it is so that a higher setting adds distortion (all still being bit perfect !), there might be a cause somewhere else, and of course it is always good to find the source of it.
All 'n all, it still can be so that when someone hears the added distortion as an added something, that may disturb so much by itself, it better not be there, and leave you with that other (base) distortion.

I could blahblah further about how people's brains and stomache etc. work upon/with audio, but I think in the end it is just so.

Lastly, it is not said at all that a higher Q2 etc. setting adds distortion. This is only what I expect from theories. It just as well can cancel out distortion, referring to jitter. And let's not forget, I made it like it works for a reason, and it is all based on everybody's experiences, and my perception of what kind of things may work out for the better. Also, there's xx% distortion in digital audio anyway, so why not attempt to cancel it out.

I hope that by now at least it is clear that Q2/3/4/5 are no explicit means to create "a" sound, nor that having them all on 0 is the absolute best. Nevertheless if I must judge something for real merits, I *have* to set them all to 0 (not Q1) because I know they otherwise influence, while without influence seems more honest.

Don't make a big deal of it, and just set them to something which is good for you !
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« Reply #39 on: October 08, 2009, 03:21:27 pm »

Peter ... is this Arc Prediction your proprietary idea or something you found in a white paper somewhere and have adapted, as I've found no reference on the net about it at all ? Does the technique lend itself to specifying a set output characteristic such as 24bit/96Khz to fully exploit the capabilities of our DAC's, rather than, what I gather currently, is a integer double or quadrupling of the sample frequency ?

It is completely my own idea, and the kind of explicit necessity to outbetter what a DAC normally does, while the DAC under construction just has left out all this "fuzz". Happy
So, at knowing a little what SRC's, digital and analogue filters and all do, I *had* to outbetter it, because otherwise the whole DAC project would have been without sense.

In brief, the technique is a combination of all what is needed to actually reconstruct the analogue wave, and which comprise of very basic elements in signal processing. The only thing I left out are the solutions from the schoolbooks, which had become objectives of their own (like : "let it ring, and the wave will reconstruct automatically"). There's two general parts in it :

1. Reconstruct the wave as good as possible, which is different per (upsampled) rate;
2. Reason out why something is 100% good, while it still looks wrong.

Mind you, I have been staring to simulation graphs for two weeks on how to do it (think of some 80 hours of staring, which is no joke), next I have been trying for two weeks to achieve something I could see as possible for further improvement, next - when that at last worked visually (in the simulation) - it sounded as sh*t, and next I needed two days to reason out how the in my eyes not yet 100% solution was 100% afterall.
So, as I said earlier, the waves cannot be resonstructed 100% looking at the graphs, but they sure just are afterall after applying some good math. In the end it comes down to, say, two different waves being able to output the exact same net result. One of those is the real analogue wave how it ever was, and the other is the Arc Upsampled representative of it. As with more things, I leave it to other experts as a challenge how this ever can be so. But, they may never get to the beginning of it all, because all begins with the prediction of the samples itself which can exist in one form only : the good one. yes

In order to answer your question more directly, the higher the upsampling, the better it gets. However, this does not say that the higher the upsampling the better you will perceive it. This is because the base (or objective) of it all is removing harmonic distortion of the stepped digital wave, while at some point other forces are there which prevent from making it better, or be able to measure it. Still the theory is so that the more upsampling the better, but (thus) with the foot note that you can just as well stop at some point at doing it.
Going from 44.1 to 88.2 clearly is not enough, because the HD is fairly above the capabilities of my DAC (but here you go already ... *my* DAC ...).
Going from 44.1 to 176.4 is as good as the dreaded ringing filtering does it, which may be a measure for some. And, I don't have another measure because a. I can't measure beyond 96000 for the "half of nyquist" (equipment limitation) and b. I have no means to output 352.8 from the PC (half of Nyquist would be 176.4 then), which is necessary in the case I would try to go beyond Quad upsampling (Octral ?).

All 'n all Arc Prediction puts the sample exactly where they would have been at a higher (A/D) sample rate, but it uses the representative (wave) model for it.
And this is what you hear.

Now, as said earlier, I must sit back and stare again, in order to reason out how it can be that for OS/Filtering DACs it works too (or at least for some as it seems). But I really can't imagine it works as good. This seems "mathematically" impossible. But we'll see.

Peter
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A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
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« Reply #40 on: October 08, 2009, 03:30:04 pm »

It is a little clearer now, I think.
So in theory it would also be possible without upsampling, just recalculating the extra bits so the shape of the wave gets more 'natural'? (leaving out the peaks between two samples of course).
Makes me think of making a picture by connecting the dots on the paper. 4 dots can become a rectangle, but if you get more dots it appears it is supposed to be a circle. You are then calculating the perfect circle.
No idea what sinx will do with that!

Haha, no, it is just the other way around. The extra bits do (about) nothing, because they are only about the amplitudes of the waves (frequencies). So, this is accuracy in the amplitude domain (also quite important of course). It is only possible with upsampling, because only then you have the "opportunity" to capture the shape (but think in peaks only for your conveniency) better. With extra bit depth only, you can only let rise or drop existing peaks and this can't be done on its own (unless all goes in an even fashion, but then there wouldn't be any change accept general volume change).

Your imagination of the picture and 4 dots is okay. But this is not the actual problem. Draw that rectangle, and give it four corners. Imagine those corners are the sample points. Note : In order to really do this, you'd need infinit rise time which does not exist. Now shift the time domain a little, meaning that your A/D converter just starts a fraction later, which is 4mm on you paper to the right. What do you get for the shape of your original rectangle ? Keep in mind, your rectangle stays where it is (the singer started singing), only the A/D starts a litter later.
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« Reply #41 on: October 08, 2009, 04:27:09 pm »

Quote
There is more going on than what I currently can reason out 100%;
This is related to hires material not sounding as good as this upsampled 16/44.1; I won't say that this is caused by all hires being upsampled (by the same wrong means) from 16/44.1, but something is going on with "native" e.g. 24/192 material which makes it hide those on stage sounds and everything. Personally I don't think we can end up at "but Arc Prediction is wrong afterall", just because it does so right for such a long time (5 weeks in my case). Notice the difference with "faked high detail", which is very easy to create by just removing some (lower) mid frequency output. Result : "Ah ! you can hear her spitting in the microphone now !!". Yeah, right. But this won't last for long, because it is "created" by removing something else first. It won't last for 5 weeks.
So, it intrigues me that it is just sheer high resolution I perceive from this Arc Upsampling, while native 24/192 (or 24/96) already shows explicit lower resolution. So, "as good" would already be strange (but according to my own applied theories it can be done), but how "worse" falls into place ? ...


This still remains odd, yes, don't like the HiRez SQ either.

Strange!
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« Reply #42 on: October 09, 2009, 10:40:47 am »

LIVE & ALIVE, that's the only way I can describe 0.9y-4. Fantastic SQ, Peter you have done a remarkable job.
This is without having an NOS/Filterless DAC, as yet. Borrowing one in a few days to hear the difference.

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« Reply #43 on: October 09, 2009, 11:07:22 am »

Thank you Arvind. I was hoping for that, especially for you (being especially dissatisfied with the later version(s)).

I'm already very interested in the results from your comparison to come.
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« Reply #44 on: October 09, 2009, 12:44:34 pm »

hello P
is there any upside soundwize in upgrading from 9y3a to 4 if I cannot turn off the upsamling?
best
Leif
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« Reply #45 on: October 09, 2009, 01:01:09 pm »

Hi Leif,

Yes, there sure is according to what -thus far without exception- everbody tells just without applying upsampling in XXHE. However, thus far with only 1 exception, everybody just applying the upsampling onto OS DACs is even more satisfied.

So I was actually awaitening your response from the ESS Sabre !
Go for it ! (and let me know please)

Thanks,
Peter
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« Reply #46 on: October 09, 2009, 01:07:37 pm »

hello P
will try this weekend
be back tom or sun
until then: have a nice weekend
best
Leif
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« Reply #47 on: October 09, 2009, 11:59:08 pm »

... thus far with only 1 exception, everybody just applying the upsampling onto OS DACs is even more satisfied.

I'd be very interested in hearing people's thoughts on the following:

1. engage double or quad upsampling
2. select 'No upsampling' - take a listen
3. select 'Arc Prediction' - take a listen
4. what are the differences between 2 and 3?

It would be interesting to see if there is ever the case where (with certain DACs) 2 could sound better than 3.

Mani.
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« Reply #48 on: October 10, 2009, 10:51:22 am »

... thus far with only 1 exception, everybody just applying the upsampling onto OS DACs is even more satisfied.

I'd be very interested in hearing people's thoughts on the following:

1. engage double or quad upsampling
2. select 'No upsampling' - take a listen
3. select 'Arc Prediction' - take a listen
4. what are the differences between 2 and 3?

It would be interesting to see if there is ever the case where (with certain DACs) 2 could sound better than 3.

In my system, upsampling causes worse timing and flow. Both alt. 2 and 3 sounds less like music than vanilla, I am not sure if upsampling with or without Arc Pred is the worst, but quad clearly sounds worse than double. Mind you, "worse" is relative here, it still sounds good, but I am not willing to trade an ounce of PRaT against more detail and space.

The DAC on my card is a Wolfson WM8740 I think.

And let me just add that with the below settings, y-4 sounds sodding fantastic here.
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« Reply #49 on: October 10, 2009, 12:04:12 pm »

hello P
I´ve just been reminded why I haven´t thrown this software and replaced it by Itunes! Happy
The y4 works great!
1st impressions
right "out of the box" same settings as y3a, it sounds better resolved in the mids and up.(slightly) improved midrange and up dynamics
funny but the soundstage is denser in the centre with better depth,maybe a little narrower but the channels hangs better together. my ss width is maybe narrowed from 6,5 to 5,5 meters but with better centrefill.
bestr with arc pred and quad ? may that be correct?
q 2 and q3 tested down to 0 but too dull and reset to 15 on both
IMO
well done
will be back L8R when lived a little with these settings
best
Leif
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« Reply #50 on: October 10, 2009, 12:27:19 pm »

Hi there Leif,

Quote
maybe narrowed from 6,5 to 5,5 meters but with better centrefill.

I must admit that with the earlier y versions I received remarks from people about what I had done to the sound stage in my own room  because it had gotten so wide (this was not negative by itself). So I guess it is back to more normal now perhaps ?

Quote
bestr with arc pred and quad ? may that be correct?

For me sure it is, and for all except Fidelio too (unless they can't do Quad, but then Double is the best).

You may have read though, that I can't reason out yet why this would be (the best) with OS DACs. So if you (later) have comments about what you perceive from it, it may be useful for me to further improve on this (ha, right, I didn't say I was ready with it !).
Of course I now the ESS Sabre (24 bit version) so you can talk in "Sabre" language and I will understand I think. I don't have it connected anymore, so I can't (easily) check myself.
The more people this works for, the better it is of course.

Peter
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A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
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« Reply #51 on: October 10, 2009, 12:59:42 pm »

hello again
short update
general improvement even more obvious on high rez material except of course I had to set the xx to no oversampling!
better resolution gives better sight into the ss
best
Leif
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« Reply #52 on: October 10, 2009, 04:57:46 pm »

Geeze, I'm not sure what's better anymore. In my system I basically have two 24/96 dacs. The first, an M-Audio Audiophile usb, I use to convert my usb connection to spdif (it sucks as a dac). From there it goes to an older MSB Gold link III. I have no idea how good (jitter free?) the conversion from usb to spdif is. The MSB has a toggle switch for upsampling which allows me to upsample to either 96 or 132K. To date, with every version of xx I have used, the upsampling to 96k is turned on; this setting very much reduces what I perceive as digital harshness. However, with all types of music, I have to leave the Q1 setting of xx at 14 or I reintroduce what feels like digital harshness back into the sound (lately Ive been at 30,30,0,0 for the other Q settings). To my very uneducated ears this setup has blown away an elite series pioneer cdp and a camelot merlin transport I have used in the past. One could say I have become very used to, perhaps even addicted, to this (distorted?)sound.

Along comes 9y-4 with arc prediction upsampling which I am able to use at the double setting. At this setting, I am now able to turn off the upsampling of the MSB and I am, for the first time, able to change my Q1 setting from 14 to 4 and the sound seems to just get better (not sure if this will work for classical music). With this set up the sound is  different but I'm just not sure if its always better. When I have the volume at divorce inducing levels (as my wife will tell you, the system is in the living room) 9y-4 is better--I can go significantly louder before the sound begins to seriously distort. However, at quieter levels my 9y-3a setup, MSB doing the upsampling, seems more "present" or maybe slightly louder is the best way to describe it. Another way to describe the difference is that the 9y-4 settup sounds more refined but subdued, more like a studio recording, and when the MSB does the upsampling the sound feels less dry and more "in the room", or more like a live recording. Does this make sense?

At this point I'm on the fence. Am going to just listen to 9y-4 for a while and see if I'm moved one way or the other. (Am sort of suspicious that the usb to spdif conversion is probably compromising the 9y-4 setup.)

Oh, I haven't compared 9y-4, without arc prediction upsampling, to 9y-3a.
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« Reply #53 on: October 10, 2009, 05:55:26 pm »


Anyway, I said lower was better because when I've left it at 128 sample (after listening to 24/96 and forgetting to set it back) and listening to regular 44.1 in the past it did not sound as good as when set back to 48 samples ... so, just figured lower sample rate was better.  Anyway, running it at 64 samples seems to work well and FF settings says it's a 88.2 signal it's sending to dac... so, alls well so far, nothing falling apart.

I don't want to muddy up the forum, but I want to address a change I've made with the Fireface 800 in case others have it and are trying settings... Right now I'm going back and forth between FF800 settings of 64 samples and 128 samples and for some reason, with this upsampling arc prediction, I think 128 samples sounds better.  This is kinda baffling to me because I thought if it plays at a lower buffer sample rate that that would sound better, like it always has with 44.1 material in the past. oh well.
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« Reply #54 on: October 10, 2009, 07:12:47 pm »

Oh, I haven't compared 9y-4, without arc prediction upsampling, to 9y-3a.

You should try, y-4 without AP and oversampling is the best sounding XXHE so far - by far (in my system, that is). Especially in the groove department. Tried out a few arias from Orlando Finto Pazzo (Vivaldi's third opera) today, and they rocked just as much as Queens of The Stone Age at full volume.
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« Reply #55 on: October 10, 2009, 08:06:08 pm »

Keep in mind guys ... this whole things has been created especially and explicitly for NOS/Filterless, and if it works out wrongly there - or I did something wrong, or there's something wrong with you system. But :

With OS/Filtering it "should not" work, although hard to proove for me at this moment, and the evidence is merely against my own thinking.
Apart from your evidence Fidelio.

So keep in mind these completely different two leages; anyone saying that Arc Predection works out for the better on OS is *not* stupid, and anyone with OS sayng that without Arc Prediction works out for the better should be right. But all may heavily depend on the kind of filtering with OS. I don't know yet. So, everbody is right, which btw always is the case in here. Nevertheless I really really like to hear from more and what they perceive.

As always, thanks.
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« Reply #56 on: October 10, 2009, 08:17:25 pm »

Peter,

Maybe a bit like musterd after diner  Grin

I have been able to play for a few day's now and my findings are: Pure and crystal clear. The bass is unbelievable and very well placed. Had to lower it a bit though. (Never done that....)

Indeed the best so far and as you know just 16/44.....  Happy

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« Reply #57 on: October 10, 2009, 08:38:40 pm »

Apart from your evidence Fidelio.

Being in a massive minority, I have been trying out AP vs no AP some more. My clear opinion is still that AP kills the "natural" aspect of the music, making it sound, while very detailed and neuanced, a bit hard and dull. Makes me wanna change songs.

Had my wife switch between AP and vanilla this evening (omg she thought I was a nut) without me knowing what played, and it was extremely easy to say if AP was on every time, especially with rock. AP makes rock shouty, vanilla makes rock rock. AP also makes sibilants more aggressive.

Mind you, my speakers are a bit particular, very dry and detailed, and very fast, which makes me sensitive to timing. I don't really give a fuck about soundstage and 3d and space and what not, only the groove. In light of this, I have probably chosen the wrong speakers, since they throw a massive soundstage and have a bag full of traditional round earth hi-fi qualities. However, when one gets it right with the other gear, they are probably the best flat earth speakers ever with tons of boogie.

My point is that, with these preferences, you can see what I think is important in a system and the reason I don't like AP.

I propose a test. Find a track you KNOW makes you want to move your body. Turn the volume loud, but not so loud that it's unpleasant. You should feel the bass slightly (no headphones here use speakers), but it should still be comfortable volume-wise.

Now, turn on AP, preferably with Quad but Double works too. Play the track for a minute. Play it again for a minute. Now turn off AP and any upsampling. Play the track again. Does it make you wanna move more?
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« Reply #58 on: October 10, 2009, 11:36:09 pm »

Sorry Fidelio, but I think you're missing the point here.

Others won't be able to replicate your experience, unless they have the same gear that you do. For example, your experience is totally the opposite of mine. But I'm not surprised, because we don't have the same gear (apart from my WB Chimeras).

But this is OK... and actually to be expected, I think, from what Peter has said.

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« Reply #59 on: October 10, 2009, 11:49:00 pm »

Sorry Fidelio, but I think you're missing the point here.

Others won't be able to replicate your experience, unless they have the same gear that you do. For example, your experience is totally the opposite of mine. But I'm not surprised, because we don't have the same gear (apart from my WB Chimeras).

But this is OK... and actually to be expected, I think, from what Peter has said.

I agree; however, I wasn't trying to get anyone to agree with me, I was more wondering what parametres people judge their SQ by. AP sounds more detailed and airy here than non-AP, but that does not mean I like it better than the non-AP sound. The two settings are distinctly different in my setup, and I was wondering if others are of the same opinion, and if so, how they differ.

btw, great sodding speakers you have there m8 drool What do you use to drive them?
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« Reply #60 on: October 11, 2009, 12:14:24 am »

Fidelio,

I tried your experiment and also hear more livelyness without oversampling/Arc, BUT I don't like it!
To me it sounds like digital the way it has been for always, while OS/Arc gives a more relaxed/natural sound the
way good vinyl can sound. Maybe it's just a wrong taste but this way I enjoy listening for longer periods (with higher volumes).
But I'm sure it depends on the system and a lot of people will have to reconcider their systems (not their taste) because the new XX sounds really different!
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« Reply #61 on: October 11, 2009, 12:44:05 am »

Fidelio,

I tried your experiment and also hear more livelyness without oversampling/Arc, BUT I don't like it!
To me it sounds like digital the way it has been for always, while OS/Arc gives a more relaxed/natural sound the
way good vinyl can sound. Maybe it's just a wrong taste but this way I enjoy listening for longer periods (with higher volumes).
But I'm sure it depends on the system and a lot of people will have to reconcider their systems (not their taste) because the new XX sounds really different!

Thanks man smile

I was principally interested in whether or not others experienced big differences between AP and no AP. What one likes best is subjective and there is never "wrong" taste. Computer audio is so confusing at times that I may have preferred AP by merely changing mainboard in my PC, one never knows.

It's weird though, your description of AP is just like how I experience no AP. Goes to show that diversity in gear gives great diversity in results when it comes to PC audio.
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« Reply #62 on: October 11, 2009, 12:45:21 am »

btw, great sodding speakers you have there m8 drool What do you use to drive them?

Ah, a bit complicated... and probably too off-topic to discuss here.

What I will say is that my DAC (it's actually an AD/DA) seems to 'like' quad upsampling with AP. This is by far the best upsampling I have ever heard. I've pretty much been dead against upsampling until now, feeling that the onboard over-sampling does a better job. But if you can switch this over-sampling off (and also any upsampling that might be going on) then quad AP definitely seems to be the way to go.

But if you can't switch these things off, I can't see how quad AP can do it's magic... which I think is what you're experiencing.

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« Reply #63 on: October 11, 2009, 06:45:51 am »

Hi all.  No extensive back and forth here.  I've liked double arc prediction (can't do quad) with 9y-4 so much it was hard to start doing the audiophilic back and forth dance... but I did it a little.  I used a Chesky recording The Coryells and rock, Pinback and Nada Surf (normal phase for his one).  With the very well recorded The Coryells, I could go either way and be happy plain vanilla or arc prediction... but I still preferred arc prediction quite a bit more in comparison.  For me, it just takes a big step toward sounding more live... maybe thickening up a bit too much but much better than the thinner tinnier sounding plain vanilla.  For rock it was no contest.  Double AP was slamming articulate bass, burried vocals come out better... ssssssssssibilance is much Much better ... and that's a big one.  The designer of the Stello DAC really wanted people to like his upsampling to 192... I never have, and have always selected 'bypass'... but maybe with Peter's AP and whatever Stello did to this OS DAC in the analog section to maybe make it work with his upsampling,,, maybe Peter's AP is doing well with it in bypass mode... still, I can hear now where I'm really chomping at the bit for Peter's DAC,,, if I can afford it :/ hehe.  With double AP things aren't perfect with this Stello dac.  Sometimes I can hear upper midrange vocals jump out a bit (who knows, could be other equipment, recording... ).  I also might try turning the bass down a bit,,, but, otoh, it's the best bass I've ever heard and sometimes live bass can be very ripe... keep in mind this bass is very articulate, but I might, might I say, turn it down a smidgen... dunno.  That's about all I can think to 'fault' at the moment.  I'll try going back and forth later, but tonight I just want to listen to some music and double arc prediction for me is just working well.  I surely can understand how some might like plain vanilla, it does sound good,,, remember, at this point we're all overstating things... XXHE has really taken my system to a new level that I have to learn to listen to for awhile to be able to describe better... all ways tried with 9y-4 are better than they've ever been in my system.  This is a good family to be in guys.
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« Reply #64 on: October 11, 2009, 02:11:10 pm »

Now I feel like my DAC is the weirdo DAC that does the opposite of everyone else's. SeVeReD's description of vanilla is just like my description of AP. Well.

I have decided to get a new DAC, not because of XXHE, but it has been a long time coming. Please feel free to chip in the thread I have started if you have suggestions. The choice makes me wacko

http://www.phasure.com/index.php?topic=941
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« Reply #65 on: October 11, 2009, 08:32:54 pm »

Peter,

Can It be so that the sound is sometimes, to much center-focused instead of more wider.
too much pronounced in the center, can sometimes be overwhelming
You have the album by now, I guess, haha
track 8 is a good example, but again it can be just some tracks.

I think it has been said before, by someone.

thats all, still very happy with latest version.
Dry and tighdy basses, and the hights I can only say, WOW, this was never been there, this was missing from cd-player day 1.

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« Reply #66 on: October 11, 2009, 10:56:13 pm »

Roy,

Before I am going to listen to you example, I think I may have an explanation in advance ...

First thing is : nobodoy knows how wide the sound stage musy be, right ? That is, not with random recordings you don't know about much. On this matter, it always occurred to me that MP3 (via XXHighEnd) always had a wider sound stage. So, should that be right ? I guess not (this is just an idea, not science).

Now think of the term "coherence". It is not difficult to feel this is much better with this Arc Upsampling. If one thing, everybody would agree upon that (I think). But now let's look at the coherence between channels ...

We all know - and wil agree - that when the two channels exhibit the same information the sound will come right from the middle. Right ? right.
Now think of the smearing I talked about, either from harmonic distortion or the explicit means of it by oversampling. Now, what you may not realize much, is that this goes on a per channel base. Thus, the left channel is dealt with independently of the right channel. Now try to see the implication of this ...

If there is any instrument more in one of each channels, this will influence this "right from the middle" sound, because that sound being there equally in left and right is influenced differently. Thus, that sound is actually "drawn" to the channel with that instrument more pronounced in that channel (the lower frequencies "draw" (at least for oversampling and the filtering in there, it will). So, that sound, originally being there equally in left and right is now more pronounced in one of the channels. This makes it "wider".

This is just my thinking, but it is connected to MP3 being wider. There too happens similar I'd say, because the higher frequencies are removed, and that too won't go equally per channel just because the higher frequency data is not equal.

From all this should follow that smearing in general, smears over left/right, that widening the sound stage. Is that good ? well, you could say that you like it, but if it originally wasn't so, it is just wrong.
In the end it should be about the size of instruments and voices. They should represent themselves for size. This may be very hard to recognize, because your room seems to influence it. I don't think this is so, and when it seems to be so afterall, you are fooled by something else being wrong. The room does influence of course, but just imagine your wife singing there somewhere in between the speakers. No way you will ever think her mouth is too big. Well, not for this subject. scratching This is because your room just belongs there, and your brain will interpret it correctly. But when it's coming wrongly from the speakers, you brain can't cope, and if anything the picture becomes too wide.


Completely besides the above, I like to add a small anectotical story;

I know many of you in here know Bert Doppenberg and his speakers (www.bd-design.nl). When I met Bert at first I got intrigued by his "capabilites" of letting any way bass sound to spring from a point in mid air. This was with his "before Orphean" horn designs, which quite some of you use, I know. This is timing, time alignment, in-phase stuff. This is where I started to to dive into the technical merits of sound. I just wanted to know how it was done, never mind I thought it was wrong. I even wanted to have it in my own room.
So, this is an example of things really being able to be too narrow just the same, but it is rare.
But was it really too narrow, or will you be able to "see" that pin pointed live bass just the same when it would be playing in the middle in between your speakers ? I think you will, but you are just not used to it, for so many years perceiving everything differently.

For now, I will leave it up to you whether a smaller sound stage is right or wrong. But my theories say it is correct, or otherwise I first have to know how it should be in reality. Mind you, this has been an intrigueing thing by itself for me for a longer time now. What is real in this ?

Peter
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« Reply #67 on: October 11, 2009, 11:43:24 pm »

Quote
for so many years perceiving everything differently.


Thnx for your explaination.
This is the case for me indeed, and for many people more.
From now on it will be completely NEW.
How to compare, with what, with reality, I guess.

Still trying a way to find why ARC should be wrong, but I can´t.
I´am not even sure about my DAC specs, but its just plain good
It is straight to the point, its me that should ajust, thats for sure.
I have to set my own merits right, I guess.
And I am not the only one.

How do instruments sound, and how have we perceived them for years out of our system.
This keeps me occupied for weeks, but this is very interesting.
Maybe try some classical music, next week.
Its not my thing, really, but now is the time  to give it a try.

Maybe some advise on certain tracks we do know,

and maybe even make an audiophile Top 20 test track list.
To determine some merits.

What to look for...................
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« Reply #68 on: October 11, 2009, 11:54:14 pm »

Anyone tried pushing their mono buttons (if you have one) and compare AP and vanilla?
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« Reply #69 on: October 12, 2009, 12:02:15 am »

Holy shamoly, Batman! Today was a wet day where I live, so I finally learned how to install and use hotkeys and upgraded to version 0.9y-4.  Using a Wavelength Brick (16/44 NOS), I cannot take advantage of the upsampling, but just out of the box this sounds better than any previous version.  Just played a little Kip Hanrahan  I have played easily over 100 times and was floored by the detail I was hearing and the vocals are to die for.  Trained the wheel button on my mouse to do ALT +X so I can come back from unattended without the keyboard at my chair (the mouse and screen are by my chair).  Thanks Peter.
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« Reply #70 on: October 12, 2009, 12:35:55 am »

rofl Thanks Glynn. And good you made it to the HotKeys ! But I really was lauging out loud at reading your post. Happy
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« Reply #71 on: October 12, 2009, 12:40:06 am »

Ok, here is one for going back to vanilla from my side ...

I just played a 24/96 album. Okay, let's say nothing wrong with that;
Then I tried something AUDIODIDAKT pointed out to me to test this "narrow" thing (see a few posts back). My impression : bweh, what a tame, lame mess this is. However should I judge that !

Went out, lit a cigar, thinking how to respond. But before the cigar was even lit I knew I forgot to set the Arc Upsampling. Bwahahaha, stupid me. But this was the first time I went back to vanilla, always thinking "maybe tomorrow" because of the joy as it is.

Well, what a change ! The freshness, the clarity a distortion / feeback guitar can have.
On that track 8 I see a person actually out of reach of the microphone - stepping towards it and back - in a quite large room / hall singing in between his own guitar melodies (ah, a distortion guitar with melodies, right).

But wait a minute, track 9 ... oh, this is a goose bumps track. It reminds me of Let me Roll it from mcCartney and Wings from back then. Oh yes, I see that darkened room with pillows on the floor still in front of me. love
Sh*t, 5 minutes and already done.

Anyway, this is Dave Holy (The Plumber), and while track 11 just started I'm under the impression that this is a long distance mike recording (listen to the hall at the hit of the snare). But also listen to how this makes you think this is just live, while I doubt it is. So I guess some processing is added to make it look like live ? (or everything sounds live with this upsampling, which is a very general impression I have), but it doesn't sound 100% natural to me.

All together, I receive anything but things being narrow, but this is because of the hall in general. Listen to track 12, there it gets crazy.
Btw, that drummer must be standing behind a number of floor toms. But hitting them with some angled (bended plastic) sticks (so the surface hitting is larger).

Veery nice album.
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« Reply #72 on: October 12, 2009, 01:56:38 am »

Holy shamoly, Batman! Today was a wet day where I live, so I finally learned how to install and use hotkeys and upgraded to version 0.9y-4.  Using a Wavelength Brick (16/44 NOS), I cannot take advantage of the upsampling, but just out of the box this sounds better than any previous version.  Just played a little Kip Hanrahan  I have played easily over 100 times and was floored by the detail I was hearing and the vocals are to die for.  Trained the wheel button on my mouse to do ALT +X so I can come back from unattended without the keyboard at my chair (the mouse and screen are by my chair).  Thanks Peter.

Yea for you; hot keys isn't too hard eh?  Sure is worth figuring out for the SQ unattended brings.  Are you going to have Gordon upgrade the Brick to 24/96?  Since it is NOS it would be interesting to hear how it does with double arc prediction.
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« Reply #73 on: October 12, 2009, 04:40:53 am »

I didn't even know the Brick could be upgraded to 24/96.  Since all my files are 16/44 this has not been important for me, but the new XXHE features may make it so.  As a confirmed Bottlehead, I am also waiting for their DAC kit which should do 24/96 (at least).  Either way, today I am just elated with the sound at 16/44.  This evening I spent several hours hearing my music sounding better than ever - actually got emotional about it a few times.  Today I am more impressed with XXHE than at any time since I started with it.  This new sound is that big an improvement.
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« Reply #74 on: October 12, 2009, 02:25:08 pm »

hello P
just an update:
having a day off today and now I´m listening to live at blues alley with eva cassidy and have tried back and forth with or w/o upsampling and in my system it´s no doubt: quad and AP gives the reproduction more detailed ambience and realistic sense of space
the voice and the applause is also presented more free standing from the acoustic venue
nobody´s frying bacon here Happy the applause have substance and body
bass is also more nuanced and easier to follow
and then I don´t really give a rats ass what works better in other systems;
like the true ego I am;i only care about myself and my system
well done
best
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« Reply #75 on: October 12, 2009, 05:39:15 pm »

Hi Peter,

Got hold of an NOS\Filter adjustable DAC & compared with my EMM Labs. The NOS beats the EMM hands down in every way. I tried setting XXHE with no upsampling & surprisingly there is no significant difference between double/Arc Prediction.

Another thing it seems there is a bug in the Cover Art area. Loading 2 albums or more in the playlist in a random manner, while playing the cover art changes with the track being played but the track  name (on the cover art) remains the previous one (sometimes).

Loving the sound of 0.9y-4.

Arvind
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« Reply #76 on: October 12, 2009, 06:40:37 pm »

Thank you Arvind for both mentioning that bug (which I will look into) and you report about the DAC.

Quote
Got hold of an NOS\Filter adjustable DAC

But can you also switch off the filtering completely ? And if not, is that just an analogue filter ?

A very general remark : As you will have noticed, the character between OS and NOS is quite different; give yourself some time to get used to NOS; If it is really filterless it should bear loads of harmonic distortion (but remember, NOS lovers don't care about that). Double/Arc removes quite some of that, but officially sure not enough. It may now depend on the music how much the HD still "disturbs".
With Quad the HD is sufficiently removed (but I guess the DAC you now use can't do that).

Anyway, thank you very much for letting know your experiences.
Peter
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« Reply #77 on: October 13, 2009, 06:22:50 am »

Hi Peter,

Unfortunately it is not an Analogue Filter but it reduces the filtering quite significantly. Though the DAC is 24/192 the USB limits it to 24/96 hence I cannot go to Quad. I am continuing with Double/Arc Prediction for the present.

Arvind
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« Reply #78 on: October 13, 2009, 08:47:49 am »

Hi Arvind,

Quote
Unfortunately it is not an Analogue Filter

In that case it becomes "logic" that you don't perceive much difference between Arc Upsampling and vanilla. This will work like "the worst will prevail".

Ok, Thanks.
Peter
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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #79 on: October 13, 2009, 12:33:13 pm »

After reading the possitive news about the latest version with upsampling I yesterday installed 09-y4.
My dac is a DDDAC1543. After some tying and looking  I was able to play with double or quad option. Best sound with engine2 and double
Sounding far better than any version I installed before.  Grin
I have a few questions though:
Nowere I find the option Arc Prediction Upsampling. Where should I look
In the settings I must leave the DAC on 16/44,1 any other value gives the message that y DAC can not handle the given value
How can I use the double or quad option is the DAC is on 16/44,1 ??  scratching
Anyhow, even with the settings as mentioned above the sound comes far more close to that of my analogue inout than ever before

Werner


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« Reply #80 on: October 13, 2009, 01:29:35 pm »

Hey Werner,

I am very sorry, but with a 16 bits DAC it can't be done. However, before anything you have the problem of not being able to use Engine#3, and it only can be done with that (meaning Vista is necessary).

I now actually wonder whether you (outside of Vista) ever see this combobox (see below), because I now realize I never tried it for ... uhm ... Windows2000. Haha.
... which reminds me ... No, I still didn't try Windows2000, although it must have been 2 years ago now that I promised. But I can't imagine it will be on par ever anymore (same for XP which is even worse (we think Happy)). And, since we will be cleaning the room this week where all obsolete audio stuff went to the past 5 years or so, I guess it's never going to happen anymore. swoon
Garage sale !!


Outside of everything I wonder how your DDDac managed to produce a sample rate of 88.2/176.4, assumed it really does it with your tweaks. I know the 1543 can do it, but can the DDDac do it as a consistent whole ?

Kind regards,
Peter


* UpsamplingCombo01.png (30.06 KB, 302x221 - viewed 1077 times.)
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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« Reply #81 on: October 13, 2009, 11:02:10 pm »

Peter,

I have to put some things right here.
By accident I did not download version 0.9y-4 but 0.9y1c. That will explain why there is no checkbox for upsampling
I do run vista and are able to use engine 3
With the bitrate setting 16/44,1 engine 3 XXhigh-end lets me choose double or quad. Also in the 0.9y-4 version I did now install
trying to choose Arc Prediction I receive the message that this will not work with 16 bits dac

I sent a mail to ask Doede Douma wether his DDDAC1543 can do something with upsampled material. He replied that this dac will stop at 16 bit 48K.

To find out how the new options of XXhigh-end sound I switched to the REALTEC prodigy 7.1 soundcard in my computer that has lots op options for upsampling. I prommised Doede to let him know if I prefer DDDAC1543 in 16 bits or the 'cheap' soundcard with high-end upsample options. If you like I will describe that findings here as well

Werner

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« Reply #82 on: October 14, 2009, 12:33:45 am »

Quote
To find out how the new options of XXhigh-end sound I switched to the REALTEC prodigy 7.1 soundcard in my computer that has lots op options for upsampling. I prommised Doede to let him know if I prefer DDDAC1543 in 16 bits or the 'cheap' soundcard with high-end upsample options.  
I have made the switch from DDDAC1543 to the Audiotrak HD2 soundcard with another opamps, to test all options in XX. I don't miss the battery charges for the DDDAC 1543.

http://www.phasure.com/index.php?topic=801.msg5936#msg5936
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XXHE Ver: 209/MSI Z87M, i7 4770K Hyperthreading On (8 cores) @ 1,2 GHz, 32GB, Windows 10-14393.0 from RAM / Engine#4 Adaptive Mode / Q1/-/3/4/5 = *30*/-/1/1/1 Q1Factor = *40* / Dev.Buffer = 4096 / ClockRes = 15 / Memory = Straight Contiguous / Include Garbage Collect / SFS = *20.19* (max 120) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = *Core3-5* / Not Switch Processors during Playback / UnAttended (Just Start) / All Services Off / WallPaper On/ No OSD / No Running Time / Minimize OS / XTweaks : *Use Settings* - *43,1, 1, 1, 1* / Time Performance Index = Optimal / Time Stability = Stable/ Octo Arc Prediction Upsampling / -> USB3 -> Clairixa-Itona-Clairixa ->24/768 PHASURE NOS1a-75B-G3 , Driver v1.0.4b (*16ms*) -> Blaxius BNC interconnects ->Audion Silver Knight 300B-> Speakers CornScala
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« Reply #83 on: October 17, 2009, 07:22:39 am »

still blown away with my 44.1 material
wow, just wow double arc sampling
listening to a new album for me old album; stanley clark, jean luc ponty, al diemeola ... what fun live sounding music.
also an old 1976 Lee Ritenour album First Course ... poppy funky jazz stuff that just bounces.  yea!
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0.9z-8-3a WAV/CUE files on HDDs via MB FW400>; Win7 pro ttp://www.phasure.com/index.php?topic=352.msg4021#msg4021); [XXHighEnd player  Qs 7, 0, 0, 0, 0; eng 4; adaptive; scheme#3; player priority low; thread priority realtime; clock res 5ms: SFS 420 Wink dac is 24/192 w/32bits; Play Unattended; Stop Services ticked; Wallpaper & Show Back ticked - Mirror Image unticked; Start Engine unticked;garbage collect ticked; copy files to XX-drive; *quad arc prediction upsampling*: straight contiguous:>PCI FW800 card>Fireface 800 DAC [latency 2048 samples for 176.4]; usb/ethernet/mb audio shut off @ MB
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« Reply #84 on: October 17, 2009, 10:35:05 am »

Good.

Is there anyone out there who wants to come back on his initial impressions ? I mean, I don't for over 6 weeks now, but this should count for everyone. If it is good it is good, and I can continue on it. I never perceived a single downside anywhere, but if anyone of you do, please let me know and try to describe it carefully.

Thank you all.
Peter
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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« Reply #85 on: October 17, 2009, 05:24:54 pm »

The downside is that I am now spending more time listening and less on my chores Happy
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« Reply #86 on: October 20, 2009, 01:00:39 am »

Hi Peter,

Latest version sounds wonderful! Don't have the ability to use the Arc Prediction since I can only do 16-bit but still getting amazing sound with the other upsampling settings. There is more depth, better clarity in the upper spectrum without being harsh, vocals are almost friggin' scary real! Better mid-bass which I had issues with in my system and room acoustics, music seams a bit more dynamic, soundstage is better and I find now I can use the Q settings at 4/0/0/0/0 to get the most balanced sound where before those settings would sound a bit weak in the bass and little bright in the upper mid-range. I have had thoughts of getting another dac to take advantage of the higher sampling rates and maybe sometime in the near future I will but the current set-up is sounding amazing with your latest version.

Nice Work!

Tony
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Vista Basic, Usb ->Musiland Monitor 02 -> Parasound Amp ->2.5 way Vifa/Scanspeak.
Ver. 0.9z-2, Playerprio-Low, Threadprio-Realtime, Buffer 256, 2x Upsampling, Adaptive Mode, Arc Prediction, Scheme-3, Q1- 1/10/10/0/0, SFS 70, Unattended, Services Off.
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« Reply #87 on: October 20, 2009, 09:41:03 am »

Thank you for sharing this Tony. And nice to hear of course !
Peter
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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« Reply #88 on: October 28, 2009, 12:02:06 am »

Quote
Is there anyone out there who wants to come back on his initial impressions ? I mean, I don't for over 6 weeks now, but this should count for everyone. If it is good it is good, and I can continue on it. I never perceived .......


Hi Peter,
......here some remarks about 09y-4. I'm listening via fronthorns. The sound is very direct and a lot of detaills. From the start of using 09y4 with the Quad Arc Prediction Upsampling I have the impression some tones/instruments are to much in front of the ears. Sometimes i have the impression that i hear a sort of resonance in instruments like trompet, vibraphone. When I switch off the quad Arc Pre the sound improves in a more relaxed sound. Less in front and no resonating sound in some instruments. I can't remember that I had the same experience in older versions of XX. I listen today to Mike Mainieri's (vibraphonist) An American Diary (great work!).

I do hear the qualities of the latest version of XX, detaills, a lot of space, quality of instruments........but I get a preference the last days to switch off double/quad Arc Prediction option. 

Johan
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XXHE Ver: 209/MSI Z87M, i7 4770K Hyperthreading On (8 cores) @ 1,2 GHz, 32GB, Windows 10-14393.0 from RAM / Engine#4 Adaptive Mode / Q1/-/3/4/5 = *30*/-/1/1/1 Q1Factor = *40* / Dev.Buffer = 4096 / ClockRes = 15 / Memory = Straight Contiguous / Include Garbage Collect / SFS = *20.19* (max 120) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = *Core3-5* / Not Switch Processors during Playback / UnAttended (Just Start) / All Services Off / WallPaper On/ No OSD / No Running Time / Minimize OS / XTweaks : *Use Settings* - *43,1, 1, 1, 1* / Time Performance Index = Optimal / Time Stability = Stable/ Octo Arc Prediction Upsampling / -> USB3 -> Clairixa-Itona-Clairixa ->24/768 PHASURE NOS1a-75B-G3 , Driver v1.0.4b (*16ms*) -> Blaxius BNC interconnects ->Audion Silver Knight 300B-> Speakers CornScala
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« Reply #89 on: October 28, 2009, 09:28:23 am »

Hey Johan, great to let this know. And btw, I don't recall you ever expressed anything about the sound quality. I really like you just did now.

I certainly won't dispute what you hear, but I have some remarks that may interest you;

You may have read in my "DAC topic" about the Firewire connection I achieved, and the real alive "jump on your chair" sound. I liked it very much. What I never said is that it didn't last long because me and the family got crazy of it. Too tyering. Very interesting, but tyering. So, I "quit" that connection. But, this was not "the connection" I wanted to achieve anyway, because this was with SPDIF output, while I head for i2s. Now :

My front loaded top horns are loosely "fitted" (read : not fitted at all, but layed upon a holding structure) on top of the base unit, and they have a small downwards angle. What I noticed is that during that period they had come forward 5 cm, just because of vibrations. At first I thought it happened over 6 months or so time without me noticing, but within a few days it had happened again. Now :

What I know of my own horns is that the horns themselves are able to over express when things are not quite right, and trumpets really start to sound like trumpets, but saxes also get a turmpet like sound. The sound actually is too "squary", and the resonances coming from that really resonate inside the horns.
This phenomenon is quite easy to "achieve" when working on a DAC (or really the output stage of it), and while different "sounds" coming from differently setup output stages, the front loaded horns are easy measurement devices because they start to over express the wrong stuff when indeed it is wrong;

Now, at leaving the output stage as it is, there's also the means of connection, and in the setup as I have it now, I just can't use SPDIF when it is compared to i2s. So, SPDIF is way more rough (and so much more it is hard to imagine this would be jitter impeeded only, but I guess it is), and when I use SPDIF I don't want horns. I think Roy may confirm this from an earlier visit, although the DAC was setup completely different back then. But anyway, it is a difference of day and night.

What I try to say is : it is all not one element only, but everything playing together and interacting. This includes the horns themselves, which are nasty beasts to begin with, but beware when you have them right (as a single element) !
I recall Jan Garbarek as a "testing device" you may have read about in the past, and how I could tweak him sounding OK through horns, while at first he would hurt your ears (if he's not doing that in the first place and on purpose Happy). This was the time when Q1 emerged !!

So ... all I want to say is : I think what you perceive, may be perceived by everybody with horns (to a certain extend), but it will be SPDIF (or maybe USB, I don't know) that pronounces the "accuracy" in a too sharp way.. Also remember what my ever first findings about the i2s connection were, and which up till today never changed : delicacy (up to fragile).

In your case I can imagine that playing the Q1 slider may bring you a new better position opposed to before (without the Arc Prediction). But, I am not sure whether you have to set it lower or higher. Lower will remove micro detail resonances (try Garbarek and you know what I mean), while the higher position smoothens. To be honest I never noticed that Q1 could help for "too much trumpet", but I do recall that during the "jumping on the chair" period I had Q1 at 14 (while normally I use 4 and Garbarek may need 0).

Keep in mind : I am not disputing what you observe, but maybe this can help you ?
Peter


PS: You may, yesterday, have read about Eddie and his reorganized website. In there he talks about an interesting meeting at Bert's. It was exactly that day where Garbarek teared the room down and everybody was sure this was unsolveable for horns ...
haha
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« Reply #90 on: November 04, 2009, 04:05:16 pm »


Hey,

I want, belatedly, to echo Fidelio's first comment in this chain!

Initial feelings only, since I only started listening about an hour ago, but this is a BIG, BIG step up from the previous version I had (y-2).

Immediately, bass has more 'slam', to the extent I was about to go and tweek with the subwoofer settings.

Much more foot-tappy.  Literally.

More depth in the image, forward and back.

The best front-end I've ever heard.  Period.

Tim
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« Reply #91 on: November 10, 2009, 11:58:07 pm »

Ugh,

Latest XX keeps me really interested and more and more drown to the speakers.
Kind of feeding just that kind of music, were latest xx makes that big difference.
Percussion, percussion and percussion.

It doesn't matter what kind, pauk, wood, drums, bongo's but also the metallic ones like bells, triangel and such.
XX really makes me drool, its sounds most honest ever heard, totally unheard of since the digital music age started.
But again, how to judge, when never heard before.

Really sofisticated, every album becomes a gem again.

Peter, let me know your experiences (with psu on dacs, I mean) about how gitars will do distorted or acoustic.
Try some hardrock albums or try something difficult and complex like Zappa.(can your mind cope with that ?)
This is the final frontier, IMO

How do you perceive complex music, with lots of music samples going one at once.
Thus multi-instrument parts of music were speed becomes really important

Especially with those PSU's on DAC's, you lucky bastard

Nice movie quote from The Italian Job:

I get myself,.....,............Burr Bown Dac's, and speakes SO loud, they can blow a woman's clothes of.


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« Reply #92 on: November 11, 2009, 12:22:40 am »

And turn the "Buzzing" in the music to "Sparkling", get yourself some decent wiring for your reference system, hehe, really!
This will open another dimension.
NOS1 deserves, it. (instead of using those mediocre cables, not your first choice, I know)
Also amps can be closer to DAC and make longer speakercables instead of very long interlinks (not being shielded)
It becomes rather important, at this final stage.
Don't hold your horses on this one.
I can provide you with those, you know that.

 Happy


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« Reply #93 on: November 20, 2009, 03:51:47 pm »

A couple of days ago, I had the chance to listen to 0.9y-4 and compare Quad Arc Prediction upsampling with the standard playback of 16/44.1 files on my DAC (i.e. using its own oversampling, the precise details of which I know very little about - I think it's 4x for 44.1).

Let me just start by stating that my DAC is a very capable DAC for 16/44.1 playback. The sound is very fluid, with an incredible bottom end. The dynamics are phenomenal - to the extent that it makes ALL other DACs that I've listened to (quite a few!) sound flat, lifeless and boring. (These include top-end DACs from dCS and Esoteric.)

But my God, QAP still manages to improve things further. It shows up a hint of 'bloom' that my DAC produces with its own oversampling engaged. The sound really is smeared without QAP (but still in a very nice and listenable way).

With QAP, everything becomes more focused. Details and harmonics that were smeared out before can now be heard - it's quite incredible.

There is only one downside that I hear (or at least I think it's a downside) and that is that the bottom end doesn't sound so 'full' with QAP. This could well be that QAP is actually reducing smearing down there. But in any event, it somehow doesn't sound so satisfying without that  bottom-end bloom.

On a final note, and without meaning to upset Peter, there is one thing that I'd like to mention. As good as QAP upsampled 16/44.1 files are, they are absolutley no match for the RR 176.4 HRx files... on my DAC at least.

Mani.
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« Reply #94 on: November 20, 2009, 04:00:14 pm »

Oh, there's one thing I forgot to mention. My DAC (actually an AD/DA converter) has some very accurate peak level meters built in to the front panel display. On virtually every 16/44.1 file I play with QAP engaged, the max level warning lights come on. This does not happen with the same files played without QAP.

QAP really is adding something somewhere...

Mani.
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« Reply #95 on: November 20, 2009, 05:22:59 pm »

Mani, do I remember correctly that you have a Pacific Microsonics Model 1 or 2? 
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« Reply #96 on: November 20, 2009, 05:29:20 pm »

It's the latter... as per my signature  Happy

Mani.
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« Reply #97 on: November 20, 2009, 05:53:40 pm »

Quote
QAP really is adding something somewhere...

DAP/QAP ... why not Happy

But that is true ! This is - theoretically - about cut waves because the samples (sample rate) didn't allow the proper peak, and that peak is now added (restored). However, when the track has been mastered at (or near) peak level already, that level is extended a little. But, it can't extend, so what happens instead is that the already cut peak gets widened a bit, and while your meters may not trip on a few samples of maximum value, it may on a few samples more.

Notice that things could really improve for SQ again if I first drop the volume a bit and *then* apply QAP. I'm working in these areas anyway, so I could apply a checkbox for that, so you (we all) can test that.
Notice that I always use attenuated digital volume in the first place, and it is really done in a good/harmless way.

Anyway, many thanks you mentioned it. If there's anything else, just shoot !
Peter
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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #98 on: November 20, 2009, 07:21:03 pm »

I notice when turning QAP on that the volume goes up versus having it off.  Why is that?
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« Reply #99 on: November 20, 2009, 07:35:21 pm »

You mean it sounds louder ?

If that is what you mean, yes that can be so, because normally after just under 5KHz there's a roll off, which doesn't exhibit here.
Notice that there is more to explain than this simple sentence, because any good filter doesn't roll off either. However, coming from the opposite direction so to say, a normal filter also rolls of (from above 22050 Hz (120dB attenuated) into the audio band, and hopefully it stops somewhere around 20KHz which it mostly does not because of other anomalies otherwise (the steeper the attenuation, the worse).

But then I am not sure whether it is that you perceive ...
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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #100 on: November 20, 2009, 07:50:29 pm »

Unfortunately, I'm not currently in the UK to test whether QAP sounds louder, but I really don't remember thinking that it did. If it weren't for my peak level meters, I'm not sure I would have noticed any difference in output level whatsoever.

On a totally different note, I'd like to share my experience comparing Quad-with-no-upsampling and QAP. Initially, QNU sounds 'clearer'. Everything seems to 'shimmer'. But this gets grating very quickly... almost painful! On many occasions, as I was switching between No-Quad and Quad, I would forget to tick the AP checkbox [EDIT - sorry, I meant to say "select AP from the drop down list"]. After a while, I could tell within a few seconds that AP was not on.

Mani.
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« Reply #101 on: November 20, 2009, 09:05:11 pm »

Interesting....perhaps it is just perception.  I'll get my trusty portable frequency analyzer and SPL meter out and run a trial. 
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« Reply #102 on: November 20, 2009, 10:52:27 pm »

Or maybe not. Happy
Let me know !
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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #103 on: November 21, 2009, 05:02:24 am »

I ran a test using an SPL meter.  Going QAP into my DAC increases sound pressure by 5-6 db versus going in with no upsampling.  This occurs independent of whatever filter setting I use on the DAC.
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« Reply #104 on: November 21, 2009, 08:54:49 am »

Ok, I will check it myself too, but for various frequencies (test tones).
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« Reply #105 on: November 21, 2009, 03:37:21 pm »

I spoke with the guys at Berkeley (formerly Pacific Microsonics) and did some more experimentation with my Berkeley Alpha DAC (BADA). 

Mani, as I mentioned before the BADA has 4 filters (actually 8 as there are 16 and 24 bit versions of each of the 4 filters).  Filter 2 is the same filter as used in your Model 2.  Filter 1 is a more sophisticated filter which required computing power that was not available when your Model 2 was designed/manufactured.  AFAIK, the BADA will not upsample an incoming 176 signal, but will upsample incoming 44.1, 88.2 streams.

Coming out of XXHE with no upsampling (NOS) and no interpolation/filter and using filter 1 I get amazing SQ.  Basically, filter 1 is a form of QAP - i.e. it is an upsampling and interpolation algorithm.

However, while the differences are subtle, I continue to prefer using QAP into the BADA - using either filter 1 or 2 gets me excellent results.  To my ears, with QAP the sound is more "filled in", there is more detail and three dimensionality and that bit of glare at the high frequencies is reduced/eliminated.  BTW, to my ears, the bottom end is more resolute with QAP which some might hear as less powerful.  I think this is because with QAP there is a reduction in time smearing which is what makes the bass sound louder (like a loudness control).

Based on my input and their other customers, they are going to set up XXHE on Win7 and experiment with it.   They told me that while upsamping is obviously optional, every DAC has to have some form of digital filter.  Peter, I got the impression that your DAC was going to be devoid of any digital filter?  Is that right?  They asked me what type of minimal digital filter would you want to see used with QAP?
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« Reply #106 on: November 21, 2009, 04:39:26 pm »

David, about your question :

My DAC doesn't contain any filtering whatsoever, and besides that it has been explicitly made to eliminate all natural filtering which could come from analogue steps (like my first I/V contained over a 1000 parts (not that the signal went through all of them), while right now the count is, say, zero.

And so, this is exactly what the Arc Prediction has been made for : to supply the DAC with easy to test and control custom filtering, that normally being in-DAC. It will obviously work on any DAC which really is 100% filterless, BUT that DAC should be 24 bits, or otherwise it can't work.
That it also works out for the better on OS DACs (or NOS filtering DACs if you want) mist be because the algorithms normally used (sincx) do not get the opportuniy to destroy. Like : the work already has been done, and the very same algorithm can deal with that (something like : the required result = 1.5 while the base is 3 *thus* devide by 2, versus, the required result is 1.5 while the base is 1.5, *thus* do nothing).
This is just my estimation afterwards, but I think the math concerned allows for just that.

Peter
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« Reply #107 on: November 22, 2009, 05:52:42 pm »

I ran a test using an SPL meter.  Going QAP into my DAC increases sound pressure by 5-6 db versus going in with no upsampling.  This occurs independent of whatever filter setting I use on the DAC.

Ok, just spent a few hours on this, but there is no way I see what you see. Not by SPL measuring test tones, not by SPL measuring music, and not by looking at test tones at the scope. But maybe to keep in mind : I can only compare with the other means of upsampling, or no upsampling, and, my DAC doesn't filter either way.

It should -said softly- be the conclusion that when your DAC is allowed to filter effectively itself (read : it hasn't been pre-done so now that math in there goes along with it (as how I estimate it works !)) it "filters" everything down with some 6dB ? hard to believe of course, but theoretically possible.

I can imagine it depends on the music, and for testing SPL with music I used the Yellow Submarine track from the original Yellow Submarine album. Just the first 20 seconds or so. Maybe you should try that one as well, just in case.
If you don't see it happening there, you might mention to me what you used to see it happening. Then I can check that too.

Peter
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« Reply #108 on: November 22, 2009, 10:55:45 pm »

Mani, as I mentioned before the BADA has 4 filters (actually 8 as there are 16 and 24 bit versions of each of the 4 filters).  Filter 2 is the same filter as used in your Model 2.

Earflappin,

The following has been posted on the Goodsoundclub forum: "The Berkeley Audio DAC reportedly uses the Delta-Sigma Analog Devices’ DAC AD1955A." Could you pass this by the Berkeley guys and find out for sure? If this is indeed the case, then I can't see how the filters can be the same - the Model Two uses custom, discrete full-ladder converters.

BTW, a 5-6dB increase in volume with QAP is huge! I wonder what's going on...

Mani.
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« Reply #109 on: November 22, 2009, 11:53:13 pm »

Mani, since my post I spoke to a friend who owns both the BADA and the Pacific Microsonics Model 2.  He told me the Model 2 is more resolving to his ears and regards it as being superior. 

The Berkeley guys decline to confirm what DAC they use....but I'm pretty sure it does use the Delta-Sigma. 

This comes right out of the BADA manual: "Filter 2 is the same filter used in the Pacific Microsonics PMD-100 and PMD-200 HDCD decoder IC’s."  And they were clear in saying they regard Filter 1 as being superior to Filter 2 and that neither the Model 1 or 2 had the computing power to run it.

Have you had any experience with the Merging Pyramix Mykerinos software/hardware offering? 
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« Reply #110 on: November 23, 2009, 11:20:03 am »

I think things might be a little more complicated. Loads of CD players and DACs use the PMD-100 and PMD-200 filters. Indeed, anything with HDCD replay must use these filters (other than the BADA that is, which I think uses an equivalent - superior? - filter).

For 16/44.1, I can well believe that the BADA's Filter 2 is the same as the filter that the Model Two uses... at these SRs.

But the Model Two uses true multi-bit converters, as opposed to delta-sigma, and I think does not oversample at rates >=176.4. But as you've found, it can be hard getting any definite answers from anyone.

To be honest, I'm still stumped as to why QAP still sounds 'good' with delta-sigma DACs, which by definition, need to use heavy oversampling to work. But many people are reporting that it simply does. At some point, I'll try to find some time to experiment with my FF800...

(Your friend has a BADA and a Model Two? Lucky guy...)

Mani
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« Reply #111 on: November 23, 2009, 12:54:35 pm »

Have you had any experience with the Merging Pyramix Mykerinos software/hardware offering?

No I haven't... but I am intrigued.

One thing that I've wondered for a while is how the interface can still affect the sound if it's being slaved to the DAC. It seems likely that different power supplies, etc,  could certainly affect the situation if the clock resides in the interface... but when the clock doesn't sit in the interface but in the DAC instead and the interface is bit-perfect, what else is there that can affect the sound of the interface?

I've been threatening to compare the 3 'interfaces' that I have (AFI1, FF800 and MOTU) in slave mode for a while now. Hopefully, I'll get a chance to do this in the next few weeks.

Mani
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« Reply #112 on: November 23, 2009, 01:22:19 pm »

Actually, perhaps getting the clock from the DAC to the interface 'intact' is not as trivial as it seems. Certainly, I can't see how the breakout cables that the Lynx card uses could maintain the required 75 Ohm impedance for receiving the clock 'intact'...

Mani.
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« Reply #113 on: December 09, 2009, 01:55:58 pm »

Well, it took a while but slow learners do learn..... About three weeks ago I reconfigured my system: got the dac next to the laptop and replaced an 8 ft USB and a 6 foot S/PDIF cables with a .5 meter kimber USB and S/PDIF cables, respectively. sh*t. 9y-4 with double arc prediction makes my delta-sigma sing. I had no idea just how smeared the highs were until now.....

Right now it sounds so good that replacing the laptop with a desktop just has to be on hold for a while longer.... Happy
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« Reply #114 on: December 09, 2009, 02:25:23 pm »

Hahaha, all great experiences !!

But ...

Quote
and replaced an 8 ft USB and a 6 foot S/PDIF cables with a .5 meter kimber USB and S/PDIF cables

Do note that the big trick here is the conversion from feet to meters. XXHE outputs in meters you know, so this avoids a conversion !
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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

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*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #115 on: December 20, 2009, 06:48:50 am »

the explanation is quite easy, I am seeing the same on my BADA and it's by design; to support HDCD decoding, 24bit is 6db louder than 16bit:

"encoding may use reversible limiting called “Peak E xtend” that can reproduce peaks
4 4 .1kH z is the only sampling rate that uses H DCD amplitude encoding. Amplitude
6 dB higher than normal 16 bit data. To reproduce those peaks, the output level to
the DAC must be decreased by 6 dB, or 1 bit, to allow for peak reconstruction if an
H DCD encoded source is played."

I ran a test using an SPL meter.  Going QAP into my DAC increases sound pressure by 5-6 db versus going in with no upsampling.  This occurs independent of whatever filter setting I use on the DAC.

Ok, just spent a few hours on this, but there is no way I see what you see. Not by SPL measuring test tones, not by SPL measuring music, and not by looking at test tones at the scope. But maybe to keep in mind : I can only compare with the other means of upsampling, or no upsampling, and, my DAC doesn't filter either way.

It should -said softly- be the conclusion that when your DAC is allowed to filter effectively itself (read : it hasn't been pre-done so now that math in there goes along with it (as how I estimate it works !)) it "filters" everything down with some 6dB ? hard to believe of course, but theoretically possible.

I can imagine it depends on the music, and for testing SPL with music I used the Yellow Submarine track from the original Yellow Submarine album. Just the first 20 seconds or so. Maybe you should try that one as well, just in case.
If you don't see it happening there, you might mention to me what you used to see it happening. Then I can check that too.

Peter
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« Reply #116 on: December 20, 2009, 07:03:12 am »

This may be a hard thing to understand by some, and even I am a kind of astonished by an outlay like this.
It would be true though ...

"Peak extension" is exactly what happens, and for the same reason you may get "crack detect" trips because the check for that doesn't anticipate on that yet in 0.9y-4.

Moritz ... great again.
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« Reply #117 on: December 21, 2009, 05:29:37 pm »

the explanation is quite easy, I am seeing the same on my BADA and it's by design; to support HDCD decoding, 24bit is 6db louder than 16bit:

No, 24bit is not louder than 16bit.

However, 24bit is louder than HDCD-encoded 16bit... on an HDCD-capable DAC/player... because of the 'Peak Extend' feature.

It now all makes sense to me now. Earflappin must have been using HDCD-encoded 16/44.1 material when evaluating Arc Prediction using the BADA. With no AP, the BADA would have engaged the HDCD processing and reduced the output by 6dB, in line with the HDCD specs. However, engaging AP would have changed the LSB, and the BADA would have switched the HDCD processing off, increasing the output by 6dB.

This is all down to earflappin using HDCD-encoded files. If he had used 'normal' 16/44.1 files, this should not have happened.

[EDIT: I prefer playing 16bit HDCD-encoded files back without any upsampling, to take full advantage of the HDCD 'Peak Extension' and 'HDCD Limiting' processes.]

Mani.
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« Reply #118 on: December 21, 2009, 10:27:48 pm »

Of course, my comments in the previous post might well not apply to the BADA. It could well be that the designers decided to reduce the output for all 16bit material by 6dB... in the event that some of it might be HDCD-encoded, thus requiring this attenuation for the HDCD 'Peak Extension' process to work effectively.

The only rationale that I can see for doing would be to normalise the output between HDCD and non-HDCD material from the BADA's outputs. But this then imposes 6dB of digital attenuation (still a 'no-no' in many people's eyes) on >99.99% of the 16bit material that just doesn't need it...

Of course, this 'imposed' 6dB attenuation is then lifted in the BADA once XXHE's upsampling increases 16bit material to 24bits.

My sincere apologies to earflappin if I came across as arrogant in my previous post.

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« Reply #119 on: December 22, 2009, 03:06:39 am »

But this then imposes 6dB of digital attenuation (still a 'no-no' in many people's eyes) on >99.99% of the 16bit material that just doesn't need it...

Of course, this 'imposed' 6dB attenuation is then lifted in the BADA once XXHE's upsampling increases 16bit material to 24bits.

Hmm ... Going from 16 bits to 24 bits is not much related to a higher volume ... only dynamic range and volume resolution. yes

Also, if 6dB would be needed to extend the peaks, I don't know of any album that has that headroom, although they might exist. Take for example Crime of the Century (Supertramp) which is an album carying much headroom (and is very soft for that reason), which still can bear some 2dB only.

Don't ask me how much a soft-clipped album needs attenuated first in order to get to the full peaks again, which btw gives me the idea doing just that.
I think I told elsewhere that the Arc Prediction still can be improved, and this is one of the subjects. Another is dealing with the hard clipping which actually requires a similar solution, though even more attenuation. This would be a good thing, because currently AP *causes* hard clipping where soft clipping is just under the hard clipping point.

I dare say that 50% of albums reach the maximum level at more than one sample, and therefore hard clip.

Ha, but wait a minute ...
You guys shouldn't have allowed me to apply some "processing" by liking Arc Prediction. Now the bear is loose;
With some nice Jamaican inspiration I just thought of a way to uncompress compressed albums ... whistlewhistle
Yep, should work.
The only reason I see causing it not to workout to how the (dynamic range of the) recording ever was, is that I expect not all compression to work linearly. And if not, how to know how the relations are (between low volume compression and higher volume compression). I think this can be "seen" though. So once I manage to indeed analyse those relations ... watch out for a next version of Arc Prediction ... yahoo

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« Reply #120 on: December 22, 2009, 10:26:16 am »

With some nice Jamaican inspiration I just thought of a way to uncompress compressed albums ...
Well, there's nothing like Jamaican inspiration Wink Considering the prevalence and level of compression nowadays, this would be really Cool

... if 6dB would be needed to extend the peaks, I don't know of any album that has that headroom
No, to understand why they apply 6dB attenuation to 16-bit material in the BADA, you need to understand the 3 HDCD 16-bit amplitude encoding options. To avaoid going too OT, I will give a brief overview in a new thread.

In any event, none of this has anything to do with Arc Predicition, other than the fact that using AP on 16-bit material will switch off the 6dB attenuation in the BADA.

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« Reply #121 on: December 22, 2009, 08:04:15 pm »

Quote
In any event, none of this has anything to do with Arc Predicition

It has though ...
This needs the "knowledge" of how the 16 bits domain operates, and there is no way peaks can extend 6dB without attenuating the whole lot 6dB first. Thus, from your other topic (thanks !) :

Quote
allows peak levels up to 6 dB above standard full scale level (+6 dBfs) on HDCD 16-bit recordings without generating “overs

can not be done at all, unless you believe in ghosts. Just for the 16 bit domain (mind that !) this would mean : drop the least significant bit hence attenuate with 6dB, and then allow the soft limited peaks (on purpose with HDCD) to extend to -0dBfs.
+6dBfs does *not* exist. It can't.

This all sure is related to A.P. because the limited peaks have to be extended (and will automatically !), while it first needs the headroom in order to be able to that (which latter currently does *not* happen).

What it comes down to is that A.P. does not need HDCD (knowledge) in order to work so anyway. The limited peaks are sufficient knowledgde to extend them (which currently *does* happen, until -0dBfs is reached).

I don't have a HDCD decoder, but you will see in the next version (and if I understand all correctly scratching) that you won't need HDCD decoding in order to "decode" it anyway. But, you will also see that non-HDCD encoded matertial will be worked out as good as HDCD, if only soft (or hard) limiting has been applied. And this is true for most of the albums (see my before post before I'm repeating everything of it). One thing : it will be *better* than HDCD decoding, because HDCD will loose one bit at least, while I think it is two. A.P. will not loose enything, if it's only *non* HDCD material.

Notice that this only counts for the Peak Extension, and not for HDCD limiting (worthless IMO) and certainly not low level extension (which just destroys the original sound).

We'll see. dancing
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« Reply #122 on: December 22, 2009, 08:19:45 pm »

Quote
In any event, none of this has anything to do with Arc Predicition

It has though ...

Sorry, I didn't make myself very clear, did I? I was referring to the 6dB increase in volume that BADA users are experiencing. This isn't really down to AP reconstructing lost peaks. As you said, this would be pretty incredible...

Mani.
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« Reply #123 on: December 22, 2009, 08:59:00 pm »

... the samples (sample rate) didn't allow the proper peak, and that peak is now added (restored). However, when the track has been mastered at (or near) peak level already, that level is extended a little. But, it can't extend, so what happens instead is that the already cut peak gets widened a bit, and while your meters may not trip on a few samples of maximum value, it may on a few samples more.

Notice that things could really improve for SQ again if I first drop the volume a bit and *then* apply QAP.

I totally get this. And if it beats HDCD's Peak Extension, then great. By how much are you hoping to drop the volume before allowing AP to kick in?

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« Reply #124 on: December 22, 2009, 11:30:44 pm »

As how I think of it now, I will do it the other way around : just calculated the peak values (no matter they will become + dBfs) and then lower the volume accordingly.

But as you will notice once 0.9y-5 is there, much much more is going on. Happy
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« Reply #125 on: January 01, 2010, 02:20:22 pm »

Firstly... Happy New Year to everyone!

Now, I had some time this morning and I really wanted to verify Peter's comment below:

I don't have a HDCD decoder, but you will see in the next version (and if I understand all correctly scratching) that you won't need HDCD decoding in order to "decode" it anyway.

I chose an HDCD track which I know has been mastered with the HDCD options engaged (from the Jacques Loussier Trio playing Satie).

With no upsampling and vol at 0 on XXHE, the HDCD indicator on my DAC lit up, as expected. The sound was very warm and lush... listenable, but not very exciting or foot-tapping.

I then set the vol in XXHE to -6dB, keeping everything else the same. My DAC automatically disengaged its HDCD processing (the LSB had changed after all) and increased its output by 6dB (all as expected). I expected to hear a decrease in SQ... BUT this just wasn't the case. The sound was actually clearer!!! More tuneful. Everything started sounding more real.

Then on to QAP with vol still at -6dB in XXHE (but making all the necessary changes elsewhere for my DAC to accept 24/176.4K). And there was yet another step up in SQ!

But the killer was when I then increased the vol in XXHE to -0dB (and decreased my analogue volume control by as close to the same amount as I could). Now, this sounded very, very good... substantially better than when I started out.

The only thing I can really conclude from all this is that how attenuation is applied is absolutely critical.

Also, don’t worry about buying HDCD material and not being able to decode it properly – it sounds wonderful played on 0.9y-4, with or without upsampling.

Mani.
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« Reply #126 on: January 01, 2010, 05:21:45 pm »

Quote
But the killer was when I then increased the vol in XXHE to -0dB (and decreased my analogue volume control by as close to the same amount as I could). Now, this sounded very, very good... substantially better than when I started out.

Assuming this has been the only change compared to the last described situation before, the least I can say is that analogue volume works differently from digital volume. So, it hardly can be said that the digital volume degrades in this situation (-6dB), but the additional analogue volume may just do the trick. It can work the other way around just the same and notice this depends largely on the amount of attenuation needed in the first place.

Also to keep in mind (for those trying) : something like a TVC is quite unlinear (regarding freq. response), so in that case it is easy to find a setting that sounds quite different from another.

Thanks for sharing Mani. We keep on learning (and improving from there !).
Peter


PS: I only now realize that where I said yesterday "I love the sound from Kernel Streaming" (W7) it is also the (QAP) "peak extension" engaged now which may do some tricks. I didn't A-B, but I applied it together with hopping over to W7 and completely forgot about it because of W7 itself (but at first I thought the peak extension didn't workout at all because of the WASAPI sound of W7 (again, can be personal)).
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« Reply #127 on: January 02, 2010, 01:40:59 am »

Hi all,

my first post here  Happy

I hope I won´t confuse more as I can contribute by now.

Thinking about my overall very positive experiences with the y-4 (which is at the same time my 1st XXHE version) I thought first that I should collect at least some 8 ... 10 listening sessions to make me familiar with XXHE. But after approx. 50% of those I have the feeling that I should share with you following:

1. my current favourite setting is NO OS in XXHE at all
2. followed by the 2x AP
3. and finally the 4x AP

This ranking seems to remain independent from the other settings.


- The #1 above delivers in my case (my individual taste & setup with a Weiss DAC2) at most the feeling of an analogue/real acoustic atmosphere.
- #2 seems to be more dynamic/accurate, with tighter impulses in bass, but seems to loose transients and esp. the feeling for the space in comparison to #1. It seems to drift in the direction of "digital" - impressive but virtual
- #3 finally goes along the path of #2, but significant further down the road and I can enjoy it only with an inverted phase - why ever!?

Nevertheless: XXHE, despite the lack of comfort in comparison with esp. MediaMonkey (with iTouch), Jriver and Foobar - is for me currently in terms of SQ the most enjoyable way to listen music reproductions; in addition to the good old analogue way.

It represents for me at that moment also the hope, that crossing the stream to the "Mac side" is not an obligation, when you´re searching for the SQ optimum.

Thank you Peter, so far!
Please proceed that way and don´t forget the "good upsampler DACs" - even if my impressions might change later on!?

Cheers

Josko


PS: My OS is Vista on a dedicated music notebook via Firewire to the DAC.
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« Reply #128 on: January 02, 2010, 10:06:19 am »

Hi Josko,

Thanks for sharing your experiences. Actually, your order of preference is exactly what I would expect from someone using an oversampling DAC (which your Weiss is, along with the 99.99% of all other DACs on the planet).

What you're experiencing with AP (I think) is your DAC doing unnecessary oversampling, once this has already been done in XXHE. The point here is that with 4 x AP (I like to call it QAP), no more oversampling really needs to be performed - everything 'nasty' is well outside the audio band. However, delta-sigma DACs have to continue oversampling in order to work. Many multibit DACs also continue oversampling with 4fs material... BUT my contention is that they don't have to.

Although I've never heard it, 8 x AP should be even better... played on a NOS DAC... that can handle 24/352.8K.

In any event, I think the one thing we all agree on is that XXHE is a great player... Welcome to the club!

Mani.
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« Reply #129 on: January 03, 2010, 09:53:28 pm »

Thank you Mani,

considering my "theoretical expectations" I am/was not really surprised and even less disappointed (with the DAC2  Happy ) + share your opinions so far as I can follow ...

Finally XXHE is an additional attractive ball in the audio game and the playground has never been so broad as these days.

Apart of the good old analogue world: obviously the period has aready started where software will dominate more and more the setups and the related reachable results.

Josko

btw: Chapeau! You took a legendary name Wink
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« Reply #130 on: January 04, 2010, 12:09:53 am »

btw: Chapeau! You took a legendary name Wink

For the record, I want all our German friends to know that I do not drive an Opel Manta.

Mani (not short for Manfred)
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« Reply #131 on: January 04, 2010, 12:56:00 am »

... even I am not a German - sorry!

To become serious again: during the comparison btw. XXHE and the known Win alernatives (JRiver, Foobar, MM) I have got the impression that XXHE depends less on formats (in my case FLAC<=>AiFF). With other words: by switching from FLAC to AIFF I observed with the classical software greater improvements in SQ (or less rel. difference in SQ) as with XXHE; but don´t have any clue, why.

Josko



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« Reply #132 on: January 04, 2010, 05:31:54 am »

....by switching from FLAC to AIFF I observed with the classical software greater improvements in SQ (or less rel. difference in SQ) as with XXHE; but don´t have any clue, why.

As far as I know XXHE convert everything to WAV before playing it. That will explain your observation.

Per
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« Reply #133 on: January 04, 2010, 08:53:14 pm »

Sounds reasonable. If I remember well, I have seen something written about that.

thank you Per!
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