XXHighEnd

Mani - I agree.

Peter - Not just for NOS1 owners, but also for those us whom don't have an NOS1 (like me at this time).   I've been trying HQPlayer under minimized OS thanks to xxHE and love Miska's asymFIR filter at NS9 noise shaping at 4x for 16/44.1 material.  It sounds so very analog and pure that I am often preferring that sound over even highRes.   For this to work with my OS DAC, I set the filter rolloff to slow, opposite of defaults, and makes this DAC act more like a filterless DAC.   It is very open like this, almost filterless.  My DAC is the W4S DAC2, the original.   Of course I'd also like a minimum phase fiter as well per Mani (by this I mean a min phase reconstruction filter).   

Miska's (HQPLayer) asymFIR filter rings, but the preringing is less than usual and the post ringing is more than usual and so this gives a more analog sound while sounding lush (because of the reconstruction part of this type of filter).  There is some phase shift at some point, but less so than with a minimum phase filter. 
     If I encounter any hardness from a CD, which is rare, I switch to poly-sinc-shrt-mp or poly-sinc-mp (both minimum phase).   And if hardness remains, I downshift to polynomial1 (interpolation) or minringFIR.

Please please please????

Also, can we have some noise shaping with that?   (BTW - I read elsewhere on this forum that you tried noise shaping but could not measure it.   I ask, measure it?   Please.   It is very apparent if you listen with your ears and put away the distortion meter.   And also,  forget about loosing one bit.  Apparently it is better than round-off or truncation because those end up as music dependent error, which according to some is worse than noise.)

Hey Charlie,

You have always been special with your findings and good/clear reporting. Ok, in my view that is. :yes: But maybe also there's something special going on between us. I mean how big are the odds that I start working on this on day 2555 of XXHighEnd's life (which was yesterday) while only 16 or so hours later you ask for it with a couple of "please"'s.
Of course, that post in your topic about 48->176.4 could have been a hint, but still not because you found a dozen of these kind of bugs before. Still, when I started this little project I was thinking of you ...

So I guess something of this will show up in the next version.


And that too. ;)
But let's say I have been ignorant till now because I don't see how any truncated bits which do so in the system noise itself can work out for the better. So please notice I measure reality (DAC output) and not theory (which can show noise at -200dB and better). And as I say so often, as long as things like Q5 etc. at 1 makes such radical differences, well, no way any dither is going to do *that*. So I'm merely in the leage of first things first, but to be honest there is no other first thing that I can see at the moment, so that is why this new project.

Regards,
Peter

PS: I think I will move this to a separate topic.

Wow!  That's bizarre.  I think this might be the second time in about 8 months.  Maybe we have a telepathic connection. 


Thank you, thank you.  That will probably be freakishly good for us OS delta-sigma users, especially unattended.  And I predict that those who can choose a slower rolloff in our DAC settings will see bigger improvements.   And this will work in your favor because it validates the NOS1 / xxHighEnd as a concept because most the work is being done by the software filter in the PC,  and not by the filter in the OS delta-sigma DAC.  That is, your filter running under xxHE takes over and becomes dominant by far.

Regarding noise shaping,  who knows why each one sounds so different.  The only thing I can tell you is that it's audible even if it theoretically is too low in level to bother with.   Miska's NS9 sounds more airy (higher in frequency) than NS5 or Gauss1 at 4x.   Gauss1 is more evenly spread and in some ways sounds more natural.

It we can't hear the bit data influence,  then it must be the computational influence on the jitter.  Who knows.   Bottom line: it's audible. 


I agree.  It is a different topic and also was thinking about quoting Mani and starting a new thread.  It was just easier to just hit reply and start typing.

Well Charlie, I always hate it when I can't explain something, so here's an attempt undoubtedly full with idiocy :

As you may hace read elsewhere I measured HQPlayer (mind you at 1KHz only). And, it measured worse for all the filter settings (THD I mean). Not by miles, but still. Also, all but one leave more HF beyond the audio band (22049 Hz) than Arc Prediction does. Remember, this is measurement only and I never listened to it. Now :

What if that dither or noise shaping just matters for that poorer THD to begin with; I don't recall I tried with and without dither types, but maybe that is not important. So, if dither makes the THD better, it should also be audible; All what measures better is also audibly better (to my ears), which does NOT mean that when it measures the same there are no audible differences. But this is for another many days. ;)

So what's really relevant is whether I can improve by means of dither. And here another "watch out" : I never listened to my own attempts on this either - only measured (and saw nothing change - obviously as explained in my previous post).

And so we'll just provide the options and then we can all see for ourselves.

Something else :
I engaded that multi threading for AI the other day and had the idea it worked now. But while I never listened to it, this morning (which is 8 hours ago) I started measuring and noticed that the signal dropped after 15 or so seconds. So it took me that 8 hours to find out that this was caused by the parallel processing and things not being ready yet while some piece of the software did not take that into account. So yep, that's software, and all day I was looking for yesterday's changes which had to have caused it.

And then another thing you won't know about because it was prior to you being around :
Ever back AI worked with sneaky stuff so it could start playing while the remainder of the conversions (even of the same track) were still going on (quite normal for the other players because they convert real time). This caused so many troubles that eventually I disengaged it. But I will reactivate it, so we don't need to wait for ages before music starts. And hopefully make that decent now of course.

Regards and thanks,
Peter

Ok, I have decided to let this topic be a sort of "progress made" topic which can be useful for myself already, but which also could be informative to those interested.

First progress made :
a. The "AI" filtering means can now (Redbook) upsample to 24/705.6 so I can compare with Arc Prediction like apples and apples;
b. It sounds like sh*t.

:swoon:

Of course this is only a first attempt but it can't make me positive. Actually it is so bad that after two hours of listening to it I was quite dead;

The last time I really tried must have been with the predecessor of the NOS1-USB, when I developed AI to my best knowledge. Now, this is long ago and since then I can be called spoiled with so much better sound since then, but this really looks way beyond "capabilities". So with nothing much more changed than the upsampling to 24/176.4 as we know it (but thus now to 24/705.6), this is what I hear :

1. The very worse : all sounds the same, and if there is one thing I can't stand it is that.
2. I thus hear the filter (and filters do make the sound).
3. All grey mush.
4. There is now life in anything and this is because
5. All sounds totally congested. Compressed if you want. Flat. Processed.

Before it is misunderstood, please keep in mind that the NOS1 I listen through does "nothing" to the sound that I know of, nor it has been reported by anyone that it does. It is just totally neutral, and if more neutral than that can exist then I don't know of that.
But therefore it is obviously the perfect machine to observe the filters through.

Also good to know : All the processing done is through a "pre-process", so what comes from that are giant 24/705.6 files (4 minutes or so is already 1GB) and they are played by the very same means as I do otherwise. Yes, I had to increase the RAMDisk size to 12GB to be able to play a few tracks, but this doesn't matter. That all is apples and apples does.

For this standard configuration (as said, as we know it from upsampling to 24/176.4) it is also nice to observe that the more lower frequencies measure a tad worse while the higher frequencies measure better. This latter is all logic because that's what such a filter it for. However, measuring (THD) is about periodic signals (sine waves) and this by far is not music. So the transient response is nowhere and I guess this is the complete smear I hear and which makes all sound the same and grey. Btw, this is linear phase and thus pre-rings as much as it post rings, and with a fairly (or very) steep filter (roll off starts at 20Khz and is done at 22.05Khz).

Next thing I am going to do is provide a nice set of settings for a few major response changes, so I can already use them myself to easily change things and listen. But please be warned : I see no single really helping solution; the revelation was just too big after changing back to Arc Prediction.

Regards,
Peter

Hi Peter

Thanks for trying.  Though I think you might be missing the point, or the points.

A brick-wall linear-phase filter (AI) is itself part of the problem, not the solution.  As such, it is not surprising to me that it doesn't sound good.

Some are saying that pre ringing is not found anywhere in nature, nor in traditional analog circuits, and therefore does not sound right to the human ear.  It shouldn't, so this line of explanation "rings true." (sorry)

That is why I'm trying HQPlayer, to try out some filters that address pre-ringing.  I can tell you that, over here, all of the filters that address or reduce pre-ringing sound more analog, more natural, less hard or harsh, less digital, sound smoother, and less distorted; and now some downsides, sound less live, less real, less detailed, move the soundstage back, sound more like a good stereo system,  blah blah blah.  Of these, the ones that ring more can sound more lush but can become riskier in that they can introduce some dirt on other tracks;  conversely,  the ones that ring less (we are talking mostly or entirely post ringing here) sound more dry / boring but are cleaner with less risk.

Some (most?) XXHighEnd users (me) are using OS DACs that only go to 192.  So why not compare at 192 or 384 with an OS DAC?  Oh, and if you do, open the DAC up as much as possible so that the DAC still filters Fs and as little below it.  This makes the OS DAC very fast and it acts more like an NOS1 will, except that it filters Fs.  (Remove the brick wall.  For this experiment, I set my DAC Rolloff to SLOW, which makes the DAC fast.)

Really, the original point was for requesting minimum phase filters and other filters that address pre ringing and AI is not it.  Over here, the best and most consistent one at 192 is not minimum phase but instead an asymmetrical filter with less pre-ringing and more post ringing.  This has full ringing but the pre ringing is shortened / addressed, and this results in a phase response that is better than that of a minimum-phase filter.  Izotope and others have these types of asymmetrical filters as options.  I encourage you to try one of these asymmetrical filters, as well as minimum phase, with both short and normal ringing.  ...and with dither (which I can confirm is usually better, but can be also be that None works best sometimes.)

I hope you don't give up!

Haha, maybe you missed some essences of my post ?
The most important one is that I didn't even start ... BUT must create an apples and apples situation.

And since today I have all running well for fast enough loading etc., so only now the job can start.

Btw asking me to listen to an OS DAC for its best sound regarding "a" filter is asking the impossible. If it sounds not on (my) par then it doesn't. So it will be all up to yourself. I'll just provide the parameters.

Best regards,
Peter

Oh,  okay.  I did miss the point.  Thanks.

Things were not making sense to me because of all people you would know that a brick-wall linear-phase filter is the problem, to which the NOS1 and xxHE are a solution.  D'oh!  Sorry.   :fool:

He he, no problem !
Maybe to clarify a bit more about the history ...

At first there was only Linear Interpolation. But this was only the very eary attempt to make something out of "upsampling". So, back in 2007 (2006 for myself actually). It never worked with the notice that I first had a "normal" oversampling DAC, which later turned into a NOS DAC, but it just did not work. People (including me) tried, but never liked it. Btw, same with the Rabbit etc. means from Foobar.

Then the NOS1 was due and I promised myself to make a god filtering means for it. That became Arc Prediction; it was part of XXHighEnd ad people were structly advised not to use it (nobody had any "NOS1" anyway). But, people tried it, and automatically everybody started using it. So, something in general must be at play, and at the time I could not reason it out much.

Then the NOS1 came about and because "upsampling" now 100% sure worked for the very better, I tried to implement a normal filtering means for *that*. And well, this is actually the AI means of still today. It just did not work out but I left it in (just like Linear is still in).

Then came the next stage of people going about with software + DAC like me - Miska. And along with that all kind of Mac means like Audirvana+ and iZotope. And so, while I was not doing anything about it ... that dis not work as well. Notice though that for HQPlayer I am dependent on other's judgements because I never tried it (only measured - as said).

Now anno 2014 it is about time that I try something myself and let's say that this topic is about that.
Of course all has to be in the "XXHighEnd realm" like no real time conversions etc. because that deteriorates. This is how "AI" (now between quotes) is to be fast enough first, like your remarks about the multi threading and such. So this has been step 1 and that now is sufficiently OK. Still slow. but way better than it was. Easy calculation : with a 12 core processor it is 12 times faster for 12 tracks to load, and the time it takes is the time the longest track takes.

Next thing indeed is the post-ringing only (so bye to minimum linear phase (but maybe not)) or anything more in the middle if the post ringing becomes to much to bear.

But I have quite some more options in mind like a way more slow rolling off filter that thus has to start way more early. And thus less ringing.

So we will see, never mind I will keep on liking Arc Prediction better BUT that is for the NOS1. Meanwhile I am hoping myself to improve for the NOS1 at the same time. And it is only that I am not much positive about that. We-will-see !

Regards,
Peter

If I'm reading this right (may not be), you're saying filtering is hardware-dependent.  So the new filter may be good for other hardware, but you have doubts whether it can be good for the NOS1.

Can you explain a little more about this hardware dependency, if it is not just a matter of computing power?

And also, a pet topic of mine, considering minimum vs. linear phase filters:

- Once you say "bye to linear phase," is there any way to recover phase behavior before the signal hits the speakers?  I think I'm fairly sensitive to phase, so this interests me.

- Why are people always onto minimum phase as a means to stop pre-ringing?  I dislike the post-ringing as well, which minimum phase only makes worse (while it's messing up the phase).  What makes a solution that minimizes *all* ringing (a less steep filter) inadvisable?

Hi there Jud,

I think this will require more than one post for some answers, and while having only a bit of time now, here is a first :


I knew it would be confusing ...  ;)

From previous posts in this topic it might now suddenly look I adhere ringing filters or something. But the anwer to your question is almost too daft to put up :
The only advice *I* personally can have is : filters should not ring at all - period. And that is Arc Prediction.

Well, you actuallly knew it (that kind of response).

So that's out.

When we talk hardware, it will only be about "DAC's hardware". So of course, computer power may play a role, but this is not what I have been hinting at. Unless of course you saw that any filtering needs to be fast enough to not disturb (because it is a preprocess as done in XXHighEnd).

Otherwise ... I am not much saying that it is DAc dependent, but some people think it might be just because Arc Prediction is not sounding as the better one.
But I never judged that and all I can say is that A.P. was not made for any other DAC.

Ok, this is all too fast typing. Must go now ...
More later !
Peter

Hey Peter, I don't want to interrupt any thinking you have on these questions, because you're off to a very interesting start.  I only wanted to give a brief mention that when I referred to "computing" power, I didn't mean computer power.  Instead, I was thinking of something involving the DAC chip(s) or FPGA or whatever in the DAC, and wondering if that was the reason you were saying AI might be better for other DACs but not the NOS1, where Arc Prediction was superior.  Or perhaps it is the delta-sigma modulation in other DACs?  Just part of my question about hardware dependency, specifically referring to the DAC.

To others : Don't let yourself overwhelm by out of context technicalities with the explicit notice that I hardly ever talk(ed) this kind of way in my own forum. But elswhere I am more or less able to do that, and Jud knows it.


Hi Jud,

No, that can not be my reason because the NOS1 is just that and it can not contain any processing for filtering (or it should be done in the current FPGA which is not very likely). However, we could see the subject a kind of upside down :

Any processing performed in-DAC is very similar if not worse than processing done in-PC with in the middle of that the processing needed by the interface;
All needs spikey current and therewith all influence (generally let increase) jitter at least.

If we indeed skip a 1000 other posts about this (me involved one way or the other) than by now it has gotten so much more complex that we better stop attempting to reason out what results will really be, but let's just have one example :

First of all we must acknowledge that something like Q3,4,5=1 of 1.186 makes a hole world of difference for SQ. So, those who can't agree with that (or don't perceive any difference from XXHighEnd's settings in general) better stop reading. But the Q3,4,5 is just a best example of how from a very long distance hence very indirectly the performance of the DAC is influenced. So for those who agree (and understand the concept of this to some extend) we continue ...

If this very indirect means already does its intended job, what about the far more direct and heavily impacting processes in the DAC itself ? You can call that devistating. So, a bit outside the subject, this is how the NOS1 does totally nothing or at least nothing which could be avoided. But still a lot is going on (it has to be) but anyway this is audible (that it does nothing).
Again on a side note, this is also how the heavy processing of AI has to be a pre-process, because since anything what we do in the PC matters, this should be out of the way once we press Play.
Thus, all consistent thinking and consistent behavior for the results of it (if you'd ask me of course).

Still there ?
Now, when Charlie says (and keep in mind it is just an example of how complicated things had grown) that he likes a slow roll off better, he will already be correct (on audible results) because that requires less processing in his DAC. So, a slower roll off means less order filters and any "order" can be seen as an additional processing step (which is already more than linear - thus progressively more). And thus simple : imply less processing and your DAC sounds better.

Well ... nobody can be really sure about that because what we do is influence jitter and nobody really says that any kind of more processing implies more jitter; it can easily be the other way around because the more processing the more "even" noise will be implied (depending on the speed / frequency of the current peaks) and the more even the noise the more "random" the implied jitter will be.
Fact is (for me) : The way the filter works influences jitter and since this is of vast importance, choosing another filter already changes SQ for that reason alone.

Fun ? maybe not, because this is totally out of control.

Meanwhile of course, the changed (thus other) filter heavily impacts just the same. This too makes it not so funny because apparently we *have* to change two variables at the same time while functionally we only (want to) change one. It just is so.

Not very much related but still mentioning it because it could clarify better my thinking :
Still thinking upside down somewhat - if Arc Prediction would be implemented in-DAC now *that* coincidentally would mean the leanest way possible of applying a filter. So/and, people may not realize it, but A.P. surely *is* a real time filter, though in PC software. And how much CPU usage you see ? well, around zero. When moved to the DAC it would mean the lowest current spikes (lowest in Amperage) and its spread would also be very even (which any normal filter can not be ("even")). So lowest jitter and most random like (or white like if you want).

When the latter is reflected back on daily practice, you see that this lowest means of noise implication is now also at the distance (in-PC) and theoretically still less immpacting than when it would happen in-DAC. But :
What also follows from all is that when this would be a pre-process (so all arranged for before playback starts) it's even less noise implying and using less CPU. Maybe.

Maybe, because - and this is the interesting part of it for me - ... because now the files are huuuuge (all 24/705.6) and now all the processing goes into the reading of the files which has to be 16 times higher for the higher sample rate alone, but ... For AI this *has* to happen, or better, happens by now. So, if I can do it for that, I can do it for AP as well. Sort of. Anyway it would be an interesting comparison if I'd allow AP to be a preprocess as well (for choice).
(but I don't think I will do that because it's a sort of gadget and can't have any priority really)

Maybe more in a next post.
Peter

Yes I this is possible (whether for me is another matter) but I am not so sure this is realy needed; go look at the output of your speaker, assumed you have some filtering means in there. This is also minimum phase. And, it can already well be that what happens there can cancel out what we do in our "software filters" (in-DAC included). Can, because it would be quite conincidental.
Anyway what I wanted to say is that most of us are completely used to IIR (speaker-passive) filters and only when you are able to explicitly complain about that it would be useful to make that linear. But I'm afraid we only know the difference once the speaker puts it out linearly and we can compare.

Speakers without filtering means (wide band drivers) wouldn't show any anomaly, but now please attach your turntables (again) because otherwise (indeed !) your DAC will molest. Or use a NOS1 which outputs all unchanged, or be ahead of your DAC doing things by letting AP doing it ahead of it so the DAC doesn't do a thing any more ... if so.

Dizzy ?

If you disregard my previous answer to this question, then the answer would be : because it implies a worse THD for the higher frequencies. Or IOW, because it will violate the frequency domain. So, any reconstructing filter which is not a genuine interpolator (like AP is that) will "ring" because it integrates previous and/or upcoming samples. So, part of the energy of those other samples end up in the currently output sample which makes the current output sample actually a false one for level, but also and merely for its position in the time domain. This is a but touhg to explain in one post and without the other 1000, but actually and for result it can be seen as that the more the frequency domain is better reconstructed, the more the time domain will be violated. And the other way around : the more the time domain is respected, the more the frequency domain will be violated.

While I suggest that we can choose some middle - this is not so as far as I have seen so far. Thus, it is not so that if we let a normal (sinc) filter ring only a little, the time domain is almost still perfect. So no, this is destroyed right away and this is just because of the means these filters have to work. However, when looking closely at the outputs of the filters, there is an interesting thing to observe, and this is in a quite new direction for me also. I'll try to explain without plots, realizing that this is impossible to begin with :

When we are careless about the filter's behavior and only look at say a balance between pre and post ringing without looking at the steepness really, and THUS allow for steepness needed just thinking normally ... which would be a roll off beyond 20KHz and at 22.05 all has to be dead) then the filter will behave quite nasty. It's not only that the transients are killed (which is the unavoidable anyway as in my view), but it is not consistently behaving. It is about how the one sample sequence runs too fast into the next one which needs different treatment (lower frequency actully not subject to reconstruction) and which is because too many adjacent samples are invloved to form the current one. So, say that a filter is set to use 128 asjacent samples then it just does that regardless. But, that 128 emerges from the steepness; make that slow and maybe 12 are needed only. Filtering is as good (read : reconstruction is as good), but if Mr ol-ear Jud please can ditch some frequency in advance.

Side note : If Charlie says he likes a slow roll off, he must realize that this works to the LEFT of the frequency spectrum and not to the right. So, the response should be dead at 22.05KHz anyway, and the slower roll off means that we can not start at 20KHz, but e.g. at 16KHz. Just saying ...

Now, what I managed so far (and again I should show a plot of it) is that thinking like this (meaning : 16Khz is way than enough for me) I managed to have zero pre-ringing and only 4 post ringing samples. This is not quite true because I look at the really impacting samples as far as my imagination goes. So, there are more samples ringing but they are so minor that I can count them out (visibly). If I balance out *those* minorly ringing samples, *then* I have zero pre and 4 post. This, while I otherwise can easily have zero pre and 20 post.

The big fun (and new to me) is that I can see that such a filter indeed still kills the transients (makes sines of dirac pulses) BUT it is a totally flat filter without ripple when music is playing (which is what my special Dirac pulse train can show). So it is very well behaving.
Now, when I compare this to a normal minimum phase filter with normal roll off, it is a total mess and I envision that I see the phase behavior just in the impulse responses.

So think I already achieved something major, but I did not listen to it yet.

Peter

Peter, thanks for your last three posts. I find all this stuff really fascinating. And Jud, thanks for bringing it up in the first place.

Maybe slightly off-topic in this thread, but there's something that's been on my mind for quite a while now that I felt Peter alluded to here (highlight mine):


How does increasing the data rate from 1.4Mbps to 33.9Mbps affect the noise in the [USB] interface? And if this does indeed lead to a big increase in interface noise, is this not a strong argument for having the filtering done in-DAC?

I say all this from having had to use a number of 'standard' DACs in my office system whilst my NOS1 is with Paul (Scroobius). And for all the DACs I've used, I prefer to use no filtering whatsoever in XX. As soon as I do, the sound becomes grey and unlistenable to me. I can only explain this by extra noise in the USB line due to higher data rates.

(Incidentally, before I received my first NOS1, I used a Pacific Microsonics Model Two as my main DAC. And here, I did prefer filtering in XX before sending the signal to the DAC. But I was using a PCI or firewire interface, and not a USB interface. And also, the DAC was non-oversampling at 4fs rates, which made it prime for accepting higher data rates.)

Any thoughts???

Mani.

Hi Mani,


I think it is clear already, but by guessing; I suppose you meant to say "as soon as I start using any XX filtering ..." correct ?

Peter

Yes exactly. As soon as I use any XX filtering. Actually, not just XX filtering. I've tried with HQPlayer, and again I prefer no filtering in software. The sound is best with straight 16/44.1 going into the DAC.

Of course, I'm talking about 'standard' DACs with oversampling filters built into the chips, and not the NOS1.

Mani.

First off, I don't think this is so off-topic because if someone perceives sound for the worse while actually it should be for the better, something is going on. But remember, it is all very hard to judge because we'd first need to be sure what the DAC does with the now higher sampling rate signal; if all is right "nothing much any more" (like your PMII example Mani), but undoubtedly not all is right (or the designer had another view, or the upsampling chip's manufacturer (from often ages ago) had another - or not much view at all ... or ...

But that not assumed, then the next subject is how robust that interface is against this noise. Or put differently somewhat : how much any means of, say, "filtering" is able to let influence less that USB noise, or how much USB cables matter and, well, endless stories.

Actually it can't be put into words what all can happen and to me it almost seems best to refer to Nick who was witness of a small part of it just by means of looking at this "interface noise" from a sort of your own NOS1 which is now at Paul's.
Let's say in advance : nothing much can be made out of it hence no real conclusions can be drawn or "what to do with what you see" except ... that you see something the most clearly. So let's say all kinds of sh*t all over at just measuring the supply to the oscillator.
Mind you, the normal solution for the NOS1 is not free of such noise either, but at least that is under control and explainable. What you see from that Dexa thing is not explainable at all because it only adds sh*t to the explainable noise, and all you can say is that it is quite worthless.

Did I drift off ?
No.

This whole clocking is just part of the interface just because that too needs "supply" and while the supply itself is (thus) more poor, all will be more poor of it.
So what I actually tried to say is that we can't apply general rules to common DACs which can easily be proven by looking at our own NOS1 and when we try to change a few things; right away nothing works any more *IF* we are allowed to look at the noise which now is in there all over and which in the end will end up at our oscillator(s). So, without measuring the effect of that on jitter, it seems clear to me that all sorts of effects will be there, and most probably not for the better (though it could by accidence).

Same with the higher rate, but which merely comes down to a faster change of bits going on and off. Bit = say 1.8V voltage needed for a bit going on, and close to 0V when it is off. So have more of that varying because of the higher data rate and you will have more noise. But per time unit more "flat" (think about one sample per minute and once per minute you will have a peak which really will be noticable).

Being on my stool of "proceed by inventing" we can say that it needs these kind of questions/answers for be to dig deeper in what's going on. I mean, it won't happen by itself and it really needs this typing and thus thinking somewhat in order to not write total rubbish (and as said earlier today, most are not much used to this, although they might think I wrote  total rubbish always anyway :)). So :

When we think back about Nick's measurements on the i2s data and how the bits changing themselves already imply "music" just because of the noise it implies, this can be brought to another level if you think of this I found :
(and of which I think it is totally new)

---

The louder you play the more jitter.
:yes::yes::yes:

---

Same story; you could say more bits change á la the i2s findings, but you can also (more easily) say the higher the output level of the D/A needed, the more voltage needed hence the more current needed ... the more noise implied. Ehm, the more jitter shows.

Again, I need plots here, and though I have the prepared I don't have access to them; they were waiting for nice times and nice topics. So later.
But it is just true ... the more loud the digital level is, the more jitter shows as varying jitter in between the peak-peak jitter which is a given (for at least the NOS1 situation).

What I did not do so far because I just never thought about it, is measuring the difference in jitter because of a different data rate towards the DAC (thus change output sampling rates from the player). But this is exactly why this "sparring" is needed and how finally all sould end up for the better again.
So straight on topic :

While for Arc Prediction the higher upsampling rate sure improves for various reasons but mainly in the reconstruction area, I am doubtful about any more normal filter regarding this, because no extra high sampling rate is needed for the reconstruction (just 2x is enough for that). The difference though is in the aliasing beyond the audio band, and when 2x is applied only, the energy will be at -40dB at 2x the sampling rate (thus 88200) plus/min the frequency of the signal. So, this is rather high, but whether harmful or audible is again that other matter. BUT, if this normal filtering means can bring down the jitter just because of less on/off rate going on in the interface (and further) ... then suddenly it is a good reason to not upsample so deep at all.

Ok, post is long enough already so I better stop for a while.
Peter

Hey Peter, are you talking about the analogue output stage here? So the more 'work' it's doing, the higher the noise it injects back into its supply, which finds its way into the digital supply? Or have I totally misunderstood what you're saying?

Mani.

Vandersteen speakers are supposedly designed to be time and phase accurate, including use of first order crossovers.  See, e.g., http://www.soundstage.com/interviews/int07.htm

Jud,

At being at the bottom of the first page I thought "OMG, another page to read !". And then I ended with page 2 and saw a third. There I gave up. :wacko:

It was only at the second page where I saw that this was from 1998 which explained a few things for me. I mean, I don't think this would have survived long when brought out today. However, many things told are recognizable (for me) and at least very interesting to read.

Otherwise I don't think this is about the "linear phase" as we talk about in this topic. Or not about "linear phase" vs. "not linear phase". I can't be 100% sure but I think this is about proper time alignment only and how the phase of the different drivers won't be different compared to each other;
But that is quite a different subject.
And if vdS feels (hears) that any passive (!) 1st order filter is phase linear, then I personally think he is wrong. It will be "way more" linear compared to more order filters, but that is all.
And otherwise I don't understand. Also OK.

Regards,
Peter

Mani, you are inherently correct of course. But things are not as simple as that. For now (and this is the more easy explanation) think that current spikes not necessarily imply noise as such (though all could be called like that) but that it dips the supply up to eternity ... which is virtual of course. So supposed we have a digital supply and a supply for analogue, these both will still end up in the transformer. And if we have separate transformers, both still end up in the mains. And if we have a separate mains ring then this still ...
And don't underestimate this, since some people claim that power regenerators help (usually in the opposite direction when "surge" would be a subject).

Of course there's a 100 little subjects more in between this, like how capacitors should buffer which they do not and thus much more.
How 1 mile thick copper planes could help out and such.

But Mani, still :


With the emphasis on this "say" (and my attempts to put out some English) it was my idea of making more easily clear what happens. Of course someone diving into the matters really, comes up with a remark like you did.  If I had to vote I'd be voting for the i2s merely being the culprit but with the notice that although this would be solely in the digital domain, it is not all that easy to see (for me) how the "volume" of the signal relates all the way. This doubt springs from what's really happening in there and which is "2's complement" related and where the lowest frequency determines the number of times zero is crossed and this is unrelated to the level of it. Actually, the lower the level the more crossings will take place because now the higher frequencies will do it too.

Of course I can try to make all posts to be as complicated and not understand as much as possible, so sometimes I take the easy road.

I can also say this :
I can guarantee you that the D/A conversion itself does not create the jitter I see. The why/how is another story again for later.
For myself though I now must try to reason out how the level of the signal (the due analogue signal) can incur for the jitter.
Ok, let's add some data :

A nice test signal shows less jitter than music. Peak-peak is more distinct for the test signal (like 1KHz), but the more random between the peaks is less there. Music makes the peak-peak less disdinct but fills up the area in between (read : the jitter varies more).

Time for pictures ? yes. But again not at hand and when at hand I am doing other things. :swoon:

So this is "data jitter" ? well, I tried to sort that out, and to my firm belief it is not.
What would be true though is that the zero crossing would be a very discrete thing for a test signal (always exactly the same pattern). For music it is not at all.
So i2s.
And NOW it becomes mighty difficult to combine that with my being certain about the D/A is not doing that, hence how I now that; this does not combine. But after sitting back for a while, I can now think of i2s being present in two locations :
At the DAC (this was clear) but also where it is generated (in the interface).

Ok, must stop. But I think that part is set now.
So Mani, thank you for this sparring because I wasn't as far as that.

Peter (who by now forgot how ever this subject came into this topic but alas)

Ah!!  "Someone finally understands me."   :)

For technical readers:
Sounds a little like the Heizenburg principle (can only know position or momentum with precision, but cannot know both with precision.)

And so for those who are proponents of "44.1 carrying all of the information and reproducing the signal perfectly" like they teach in schools to students who are unable to challenge the teachers,  let's just say that if ADCs sampled 20K music at 192K, we could simply interpolate and have BOTH transients and frequency response.   We would not have to choose one over the other.  Such interpolation would also not introduce phase errors.   We'd have it all with just AP.   We'd also not have distortion after 16K.  Simple.

That is true, of course, for Fs of 44.1.

However, as it turns out, the slow filter being -3db at .45xFs is not of any consequence with the settings I prefer.   I prefer the SLOW rolloff filter, but upsampled 192 or 176.4.   So the -3db point is actually around 90 KHz ,  and the lowest images start 20Khz below 176 or 192.  (= inconsequential)

It it helps,  the best sound right now for me is with 4x with no filter selected (xxHE repeats samples four times).  This slow filter with these xxHE settings is unreal sound-wise.  So far best by far with OS DAC.   The filter is very open, reaching -100db not until 0.8 x Fs.   It is almost no filter at all and is probably meant for upsampling with some sort of apodizing filter.

Hi Charlie,


About the inconsequential : I don't think so. You can make the sampling rate for 20KHz material higher what you want, but when not filtered right away you will see the images and aliases all over the place. May sound nice to you (like harmonically rich) so no comments on that. But good it can not be.
Or I did not get what you were saying. :)

Regards,
Peter

I agree.  That's correct. 

Things should be filtered.  And when filtered, there would be little from 22.05 KHz out to 158 KHz or 174 KHz.   And in which case, with 4x upsampling and a filter, the -3db around 90 KHz would be inconsequential.  And so AI or AP should provide this filtering and should sound best.

But actually, no filter selected (=repeat samples four times) sounds best right now.  And, yes, the images would be all over the place and I'm not sure where.   Have a picture of this mess?   If you do, I'd be interested in seeing it because my amps have workable bandwidth out to 100 KHz or so.

And to back up a little, I was talking about the SLOW filter on my DAC, which I think is a function of Fs, the sampling rate within the DAC.  If your new filter will filter to -3db at 20 KHz regardless of the multiplier (Fs again), 2x, 4x, etc., then we are talking about two very different filters.

Wow!  I also think you achieved something major.  You are the man.  :yahoo:  How soon can I listen to this filter?  Should sound great, and very analog.

To be honest Charlie, I want to put it up for maybe 4-5 days already because it sounds so good. Don't tell !
But I have quite major stability issues with the playback itself for various reasons including unknown (think freezing PC's). But since yesterday they seem to be solved, but still it is way difficult. On a side note, the whole weekend I have been working on a non-preprocessed method which should be very fast, and that is now in a "prepared" stadium.

But otherwise envision that for external reasons at this moment I can use 16x only and thus is any file of a few minutes 1GB right away for 4 minutes or so. So I now allow 11GB for RAMDisk so I can play sort of a whole album and the RAMDisk is needed because the reading of the 24/705600 is too demanding for everything - but what everything is I need to slowly find out (hence relearn what XXHE settings (buffers etc.) will work.

Anyway I may be more eager to put up a beta of this than you are ...

Regards,
Peter

Non preprocess would be great because of load times.   Quick like AP or no filter.

For us in the 4x world (192 DACs), perhaps it might be a good idea to give us a choice to preprocess AP, AI, and this new filter.   We won't have the huge files of double octo.  I say this because  AI is coming in ahead of AP right now with DAC almost wide open (slow filter, unlike the setting in my sig).   The order, best first is:

no filter @4x,
AI @4x,
AI @ 2x, and finally
AP @4x.

But AP should win overall or at least any AI.   AI is a preprocess.  So I can't help think that if AP were also a preprocess (optionally chosen in settings) that it would win out.   

A few updates ...

On of them is that only after a longer while I found that this all even can't work with tracks larger than around 8 minutes becaue that implies more than 2GB of file while WAV officially can be 2GB only. That all goes quite unnoticed because nothing complains really but some sort of random noise exhibits once the 2GB limit is passed during playback.

WAV can be larger with some tricks, but in the end this is not the way to go; I got it all working quite well and without lag when playback is started and by now a 12 core (or more) processor really is convienent; loading of an album of 12 tracks or less takes maybe 1 minute but in the end as long as the longest track. However, would the album consist of 1 track of 60 minutes, it theoretically takes 12 minutes.
Playback starts right away anyway, but it degrades the SQ performance hence is against all what XXHighEnd is for.

So, not the way to go.

And so I finally sat back, read into a few basics and started to build my own filters right from the ground up. Or at least one type (many exist, each with numerous options).

Small side note : I never approached it this way because it was my thinking that only Arc Prediction would be lean enough to perform it as a real time process (and then with parallel processing harmless to SQ). But since "back then" quite some things happened and one of them was learning how a processor speed of 430MHz would still imply a virtual 0% CPU usage, also because of improvements over time. And of course now encouraged by the supidly super large files otherwise (pre-process) I just had to give this a go.

Well, I should have done it earlier. Or at least I just as well could have;

Of course all starts out with selecting a filter type which doesn't take ages for its necessary processing but which is to combine with the desired result. So, that already implies quite some "reading into", also (by now) knowing that it takes several days of programming before something is working to begin with, to only see then the result. Thus that is no way to travel the paths and the theories must be well known in advance. Takes additional time, but is gained back right away because the filter should work out as intended if only no programming mistakes have been made. And I can tell you, this is tough already. Example :

I ever back started out my programming life with creating a chess program. This was cool, and fairly quickly I could not win of it myself. Small problem : I could see that it would let trade her queen for some minor piece and I could never see why. I lost from the program all right, but there had to be a major flaw in it anyway.
Now, such a thing can not be debugged; what such a program does is "trying" all the moves as a first step and look what the opponent (me) would be able to do for each of such a move. But for each move of the opponent again it analyzed what the program will be able to do. This (as I recall) 3 steps deep from each side, or otherwise there wasn't enough (processing) time. Uhm, back in 1976 or so.
Anyway - and I forgot the exact numbers - when you're in turn you will be able to make, say, 200 moves. The opponent the same for each of your possibilities. This 6 times gives 200x200x200x200x200x200 = 64,000,000,000,000 settings which are analyzed. In at least one of all these settings a queen is offered for the better, but since this is about the stacked result of all there is no way of debugging this. It is too much to begin with.

The chess example would still lead to discrete analysable results per setting. Not so with audio (data). You can't even judge the input, let alone what the filtered result of the output has to be. It just has to be 100% working OK without a single bug.
And then to think that I saw example code (lectures actually) which in practice lead to something like SQRT(-1). Well, try that. It can, it actually should for this kind of filtering which is about real numbers and imaginary parts, but if the lecture code already fails ...
And you know what ? I found all kinds of real programs which worked with that lectured code ...

Right. Didn't I tell that existing filtering means measure worse than Arc Prediction for the lower frequencies ? I did. But now I made a first one my own, and that measures BETTER.
Aha ...
Well, marginally, just as the others measure marginally worse, but at least it gives me the idea that I possibly did better (somehow).

Not unimportant : When I run this on top of Arc Prediction it is not even noticable (for the time it takes). Or not yet, because I now firstly use a most simple form (though that's not for complexity or speed reasons but for not being a steep filter - maybe the whole subject of this topic).

Now my (or "our") major advantage is that I can look at Arc Prediction and know what's to be improved on it and especially what's to remain. The latter of course is about the transients, plus AP shows how phase behavior should look like. So all I need to do is look at the latter two while tweaking the fomer to improve. Actually quite easy ...

And so I now listen for the past few days to a first theoretically set filter which looks good and it sounds better.
Aha.

"Sounds better" could be subjective because sound does change and what the brain needs to do is whether changes for the better do not get overwhelmed with things for the worse. So, to me and for now it seems true that after listening to this filter for a couple of ours and then switch back to AP, it sounds more accurate again but then the other way around shows more "music" right away. So let's say a more analitic sound changed into a better coherent music sound.
But one super major thing : I can't recognize a flavor of the filter up till now, let alone that I get crazy of the zooming etc. like commercial filters do (e.g. iZotope) no matter the settings.

Notice : In a previous post I talked about the hardly visible pre ringing and 4 samples of post ringing. That (preprocessed) filter worked quite well too, but that was still too obvious for the smearing; I envisioned myself making it up but with *that* filter I already thought "oh well, if some people like it for the better they go ahead with it". Not so with this real time filter; I think I really like this one for the better myself. But then this one is phase linear and meanwhile it doesn't ring either. Well it does, but to the same extend that other filter pre-rings (hence hardly visible).

So far for now.
Peter

Thanks  Peter.   That's exactly what I think I need in my setup  --that choice.


I don't know what is meant by "zooming."

LOL. And then to think I use that word weekly for years before someone tells me it doesn't exist.

Of course it is totally obvious to me now that this is in-existent. We Dutch ... We say zoemen (pronounce oe as oo).

Buzzzzzz

But buzz with oscillation like a fly sounds when flying.

Then of course people from the northern part of Holland should be careful because what normal Dutch call a fly, they call a mosquito. And what "we" call a mosquito they call a fly.
I don't know how *that* happened.

:)

How does a fly sound when NOT flying?

 :teasing:

Stanley

Hi Peter

Gee...  How much longer are we going to have to wait for these new filters?
I've been checking daily...  it's driving me crazy a little.

Charlie

Haha, that's your own fault, because you don't have a NOS1. :swoon:

Well, sort of and not really of course, and the problem is merely this :

Without first making *that* we have no provision of seeing what the result of the filter will be. So this is in another leage (if emulating it with plots will show a fair picture at all and I do NOT think so), and this is what has held me back somewhat. Ok, apart from the time it takes while clearly some other things were (and are) going on in parallel. But think of it : you won't be able to "see" what the filter causes, nor will I myself. And this latter I can at least do for the NOS1(a).

Anyway, see beta-beta download below;
You will see quite some improvements for the AI filter which even to my ears is listenable but which I gave up upon because of it's (still) akward-ish usage, plus no real solution in the end. Nice for the Frequency Domain, but that is really all and it will never serve the Time Domain.

So (and already talked about I think), in there there's also a "Custom" filter, and *that*'s what the sh*t is all about. Next small problem (no, same problem) : although that works perfectly all right for the NOS1(a) it can not work the very best for 176.4/192 only (44.1/48 upsampled). This is just too low to avoid images in the audio band (and the Anti Image filter does not).
Still I made one for you (or anyone else with 192 max capabilities) so you can try. And it should at least work way better than normal Arc Prediction. It still does not ring a bit, and "utilizes" the frequencies we won't perceive anyway, assumed we are old enough to begin with. Just as you asked for (the new AI filter in its default settings does that too but keeps on ringing, but less now).

For NOS1(a) users ... they too can try the AI filter of course, but they for sure should try the "Preferred" 705600 Custom filter (you will understand when using it). I am using that myself for maybe 6 weeks now, and I don't think I will go back. Improving it yes, but I never spent a second on it after this first attempt.

People should notice that the Custom filter's name is exactly that : ready for you to set in all directions. But back to the beginning of this post - you won't be able to see what you do, and it's really "no task" to start dialling without analyser (or well done emulation of which I think it is not possible).
So what I'm saying is that he "customizing" will be for myself for now, and Pre-selections will emerge from it. Just like how it is now (with actually one setting in there at this moment).

The settings are envoked by means of the [ F ] button you see below (see mouse arrow) and the Custom Filter is engaged by means of the "Custom" button above "ArcPredict" (top left section). Don't forget to click the Help link in the settngs screen.

The download at the bottom contains 4 files. Unzip them over your current 1.186 or higher folder and nothing else (!). Before doing so, kill XXEngine3, XXHighEnd and AutoHotkey from memory.

Important : This version will not allow you to go back to the ever existing version of AI filtering; for that you'd need to revert to your current XXHighEnd.exe and XXEngine3.exe. So set those aside when you think you want to compare.

The 1.186-g you will see is just the latest version but no Release Notes. All I can say is that it may be quite useful for (tablet) remote operation (since I started to use that exclusively myself again, haha).

For all who try this for the filter, let me know your findings !
Peter

EDIT : At this moment it appears so that the Custom Filter does NOT work for "DAC Needs" = 24 (bits).
This is not sorted out yet, but take it this implies static (loud noise). See posts below.

Edit June 24 : Removed the download because of a few issues. A new download (1.186-h) will emerge in this topic soon.

Thanks Peter,

The custom filter does seem to be a noticeable improvement on my non NOS 1 system. But a word of caution,  I was using AP upsampled to 192 and left that setting when I switched to the custom filter and received  deafening static. 
All works well without upsampling and this may just be my system but others might want to pay attention to the upsampling setting.

Regards,

David

David, did you ever enter that [ F ] screen first ?

Are you 100% sure you removed XXEngine3.exe from memory ?

What did you select in that [ F ] screen ?

All IOW, something is not right and nothing of the sort should have happened.
Similarly you can't even use "no upsampling" at all ... (it will give you an error message that Filter 5 could not start).

So I'm a bit clueless.
But sorry that this happened to you. Now let's (seriously) work out how.

Peter

But for a next one trying ...

It could be crucial to first go into that [ F ] screen before really playing; that should fill all the data, especially when nothing has been selected yet from the "pre-selections". Now, it is so that I obtained a provision for that, but I never tested that really.
So David, if you can confirm that you just pressed the Custom button and never went into the [ F ] screen before that, this probably would explain it. But otherwise ...

??

Peter

Hi Peter,

Yes,  first went to the F  screen and chanced the setting to the 176/192 setting before starting the engine.   I am using an Audiophileo 2 with purepower at 24 bits fed into a marantz dac.
The static that is created when upsampled has a "pulse"  to it or a regular rhythm.  The original  music is not heard at all. 
I do get the warning about the filter sampling mismatch if I start from 48000 and go to 96000 but it I bump it up to 192000 then I have static with no warning.

If there  is something else you would like me to try,  let me know.

Best,

David

Hmm ... Are you saying that you have "DAC Needs" set to 24 (bits) ?

Thanks,
Peter

PS:
This "message" I actually don't understand then. Okay, maybe you are *not* saying "works for the better !" but that is what I got from it, at least at first.

Yes,   my dac needs 24 bit in order to play.  Now,  I may be a bit ignorant about some of this but does the custom filter make any difference without upsampling?  With the quick listen this morning,  I could have sworn there was a little bit of an improvement in the sound stage.  Am I wrong? 
I've tried a couple frequency's and have trouble with upsampling any with the with the custom filter.  Some will give me an error and will play normally once I close the window about filter 5 not starting,  but others will just be the static.

Regard, 

David

David,

When the output sampling rate is lower than that of the filter chosen, the error appears (should appear) and no Custom Filter will be engaged. I think that Arc Prediction will be engaged instead but I'm not even sure about that now (actually it should not play at all, but beta-beta ...).

So it should be about the 24 bits, but to be honest I never even thought about that while I also can't even see how it would be related at this moment. But let's take it it is.

Sorry David,
Peter

PS: I put a message about this in the "OP".

Hey Peter, the Custom filter is totally phenomenal. Only now can articulate what I've thought about AP for so long. There's a certain 'nervousness' in the extreme HF that gives an edge to the music. Initially, it might even sound appealing. But after switching to Custom, you realise how much more natural the sound can become.

Thanks so much for listening to 'us' and pursuing this for us. I'm sure you've put it quite some hours on this... Well, let it be known that your hard work is much appreciated.

Mani.

 :soundsgood: :xx: :veryhappy: :thankyou: :dancing: :clapping: :good:

Fantastic and very musical filters.  Thanks. You can tell I'm very happy from my previous post.  I'll have a hard time picking a favorite between custom-176/192 and the new AI.  Over here it is majorly different from before, for the better, and I'm running 4x only with an "open dac" meaning using the simple filter that has slow rolloff.  So slow upon slow.

One problem I must report about AI:  AI is not multitasking as I expected from reading your previous posts.  I only see one yellow CPU-core light and albums load slowly as before.  Example,  one Spiro Gyra album has 8 tracks with all tracks nearly 5 minutes long, and the longest track is 5:49.  Well AI prodded through these 8 tracks one by one and I timed it with a stop watch.  This album took 2:09  :sleeping: to start playing. 

You can see my PC specs below.  This is a fast xxHEPC and I'm running it at 2.5GHz; I don't use playback drive. 

One thing that could be causing this is my AI settings.  I have the pre-post ringing set to 17% and I use dither set to "noise shaping."

Why is the new AI so slow at loading?

One other problem with AI:  hitting alt-N during unattended playback causes "DAC can't play... check settings.." (I think it's that one).  Hitting alt-P does not resolve the problem.  I need to hit alt-X and hit play button for it to play again.

The shortest path to getting to the next track is to hit, while a track is playing in unattended mode, hit alt-X, alt-S, select next track and click Play.  Well, at least that works without ever getting error.

So hitting alt-X appears to be the key.

Hi Charlie - Thank you for all the feedback.


But it sure is, unless I eliminated it all because of other issues I ran into. So I will re-check !
But 2:09 for a whole album ? seems quite impossible to me (if track per track), although I don't really have the reference on 176/192.

Please notice that I regard AI a dead end. It is in there because I started out with improving it to later come to the conclusion that it is and remains akward (and slow). So representatives of it will be built onder Custom. But, takes time of course.

I will check the multi threading.
Peter

Hi Peter,

SQ of the new Custom filter is more than excellent, even with my NOS1 non a (yet).

I am however experiencing ticks in the right channel on 705.6KHz samplerate. These are related to the level of the music playing.

The music sounds significantly more subtile on 705.6 than on 352.8KHz. Much more like my preferred v8e setup.
FWIW:with the latter sample rate there are no ticks.

Is this still work in progress or should I adjust some specific setting?

regards, Coen

Raising SFS helped a bit.

 :)

I myself have the hunch that it is a good idea to kill the sound engine before playback is attempted with a new setting.

I heard ticks too, but when I try to incur for it (so I can solve it) it never happens.

About the volume :
I would not play at -0dBFS with this. So maybe people don't do that anyway, but at this moment the filter is not 100% protected against "clipping".
-6dBFS at least should be OK.

Peter

Thanks Gerard, but no success here. Maybe I have been to conservative  ;).

regards, Coen

Hi Peter,


This does not seem to cure it right away. Maybe I have a still to low SFS like Gerard suggested. I am reluctant with this because some of the magic disappears when raising the SFS here. I will try values above 12MB. Do I recall correctly that the Q1 and Q1x also have to be raised?

I have to note that I heard the ticks right after the 'fresh install' of the filter upgrade package.


General advice taken, though I deliberately never play louder than -6dBFS.

regards, Coen

Hi Peter,
made a test with the „Custom“ filter, and what amazing cymbals, detailed with the “clicks” etc.
But overall this filter emphasizes the highs too much on my system.

Joachim

Joachim, did you try 16x AP? I think you need to in order for the filter to work properly.

Mani.

I wish I knew myself what is going on really. This is a text from someone who doesn't frequent the forum :


  I've managed to play without hearing any more ticks. I've got the SFS down to 1 now, each time hitting the Blue LED as I adjust the SFS lower.

  Now the sound quality is fantastic. More pin point focus, recording ambience is clearly captured, and the highs are simply gorgeous.  It is really hard to imagine what will it be when this new filter is combined with NOS1a.

Hi Mani,

yes, the filter works in 16X AP only. The sound (characteristic) of my speakers are from the "bright side". So, bad luck for me.

Joachim

Joachim,

If you prefer 8x in general you can also select the 176400 filter (and use 8x).

Regards,
Peter

Forget about this. This was a mistake (I was told).
Sorry.

Hi Peter,

got it. This setting is more pleasing for the speakers and has an impact of the location of the instruments in the recording studio too.

With 16x the piano was in the back of the studio, and moved now (8x) forward, which is more realistic. (Stanley Clarke Trio)

Joachim

More data on this problem:

The same error happens when I do:
- play unattended using AI set to 4x
- hit alt-E to pause.
- hit alt-P to play   (error happens here).

When I come back to alt-x, engine 3 is already unloaded.

Curiously,  the following doesn't result in the same error:
- play unattended using AI set to 4x
- hit alt-X and wait for xx to come up,
- click on the pause button
- click on the play button.

Plays just fine and xxHE goes into unattended.

New Data Point on this 2:09 situation with AI.

I have "found the problem."  Recall that I set my upsample rate to 4x for my 192 DAC...

-AI multitasks every time and just fine when the source is 88/24 or 96/24 HD material.  Conversely, 
-AI fails to multitask when the source material is 44.1/16.

I'm still not sure if the problem ties to 88 vs. 44 --OR-- 24 vs. 16, could be either. 

That might be enough info? 

BTW, I see/hear no problems with AI and multitasking with 88 or 96.  So probably no reason to disable it.

[Edit: I found one album 48/24.  It multitasked.]

Thank you Charlie.
Peter

Hi,

After some tests with the different filters (rev. “e” & “g” ) the revision “e” (sent with the NOS1a) is with 8xAP the preferred one on my system.
Rev. “e” with 8x AP brings a seamless acoustic pattern, nothing protrudes. The soundstage is in a perfect balance.

Best SQ I ever had.

Joachim

I found some 44/24.  It also multitasks.

This confirms that the lack of multitasking on AI is a 16 bit vs. 24 bit issue.

Can't this be the difference between WAV (not) and FLAC (does) ?

I found the ticking issue (and solved it). Now for the DAC Needs = 24 noise ...

A new 1.186-h will emerge hopefully soon (1.186-g download has been removed).

Peter

But not here. BUT :
While writing this, I suddenly recall that it works at the album level. So, load a Playlist with tracks from several albums and then it won't work (and never will - just saying).
So that will be it.

Peter

Uh oh...  I think it could be.  I will confirm tonight. 

But I hope not because most of my CD collection is ripped to AIF using dbPowerAmp, AIF being like WAV.

No no, WAV and FLAC both work. AIF(F) will too (not that I tested that).
On another note I used that AI version myself for a couple of weeks and if it would have not used multithreading I would have known. This with the notice that I used it with upsampling ti 705600 and a 430MHz set processor ...

See New Filter request(s) - XXHighEnd 1.186-g (http://www.phasure.com/index.php?topic=2944.msg31590#msg31590) for the original text coming along with the Filter Beta.

Unzip the 4 files in the download (at the bottom) over your current 1.186 or higher folder after removing XXEngine3, XXHighEnd and AutoHotkey from memory.

This solves the ticks.
With thanks to those who noticed it was in the right channel only (now it took me minutes only to find the culprit).

24 bit "DAC Needs" noise takes more time to solve (so later) but it won't play now with the Custom Filter engaged.
Once more apologies that this could happen (but I guess that's beta-beta).

Peter

Edit : Download once again removed. :sorry:

Actually, here is the data.  See a pattern?  This took much of the night to find these.

Album	Rate/Bits	Format	Multitasks?
==========	========	=======	=======
Hothouse Flowers	44/16	Flac	Yes
Mumford and Sons	44/24	Wav	No
Other Aiff (many)	44/16	Aiff	No
Stravinski (DG label)	96/24	Aiff	No
Tedeschi Trucks Live	48/24	Flac	Yes
Many other HD	96/24 or 88/24	Flac	Yes

Certainly not 96 vs. 44.  Certainly not 16 vs 24.

Re-chance (see my previous post in this topic);

What I wanted to say but completely forgot :
The 1.186-h version (but the 1.186-i still) contains "restored" code from what has been tested to work a few weeks ago but which code was eliminated because whatever it does, it does in the NOS1 driver now. However, nobody can receive this driver yet. But because for me "it" is already arranged for in the driver I use, it can't go wrong in the restored code so I can not test it. But for you it can go wrong, and so it did for 1.186-h which has been available for a short period of time.

Thanks to one user who reported this to me, and tested the next iteration of the code, not all that many suffered.

Still it can be so that some "GPIO" error message occurs. I think this can happen :

- Right after the NOS1 had been switched off and on again (killing the sound engine should solve this (blue led) and retry playback);
- When the NOS1 Control Panel has been removed from memory (just cross it away when you had it up and do not chose "File - Remove from memory").

In any event meant error messages can only occur for NOS1(a) users.

Remember, paste the files from the zip over your current XX 1.186 (or higher) folder; nothing else. First kill XXEngine3.exe, AutoHotkey.exe and quit XXHighEnd.

Peter

But certainly also no WAV vs FLAC (because WAV here works).

There is one thing where WAV is special (and where AIF is not - just saying) and this is in the possible explicit copy actions. So, FLAC has input from wherever it is and the output goes to e.g. the XX folder (e.g. because when not denoted otherwise) while WAV stays where it is in the same situation (no input-output operation involved);

I for sure worked with a Playback Drive exclusively, which means that WAV is now copied too (or activate Always copy to XX Folder and this would also imply copying of the WAV).

Because all (multithreading) is related to input and outputs for inputs (and e.g. HDCD implying another part of such chain) I can imagine that a certain combination just is too difficult (to even "see happening" for me).

Only guessing possible reasons ...

and so Aiff  gets copied just like Flac no matter what?

and it is only wave that sits and stays where it is?

Correct.

Okay,  then I will check tonight by forcing to playback drive or forcing the xx folder and see.  But since most of my collection of CDs is Aiff, and Aiff must get copied, I don't see why that would work.  But as you said, this I/O stuff and multitasking is complex.  Anything could be.

Haha, AIF indeed does not do it.

So that's solved so to speak. Now WAV ... (your turn for now)

specifying a playback Drive did not solve the problem.

Aiff 16 bit 44 kilohertz did not multitask...  only one track at a time.

Yes but I (thus) know that by now ...


But WAV should work.

Had an hour last night so I gave 186-i a try on my standard NOS-1. As there are so many alternatives and I didn't have much time these are only first impressions. First, as great as the soundstage has been, it became even better. Single Instruments as well as orchestral sections are even more defined and ""right there" within the sound stage. Second the venue whether recording studio or live on stage is much more easily heard even on cr*ppy Eva Cassidy recordings.  And lastly, it seems that its no longer possible to get the HF to distort, which is both pleasing and a little weird. I was playing female vocals at 90-92 db and they sounded great except it was too loud. My normal levels around 85-88 sounded good too but at 16x upsampling they were a bit too smooth sounding. At 8x and even 4x they seemed to have more weight and punch. When I was testing the upsampling settings I was using the 176 filter which allowed me to play 4x. 8x, and 16x without changing the filter. I thought 16x sounded much more like 8x using the 176 filter so that filter was definitely preferred for 16x. The 705 filter just looses to much punch on my system (91db speaker sensitity).

Am hoping a 384 filter is added to the mix just to give that a go too! I tried the AI filter using a wave file but it took forever to process. After two minutes of waiting for a standard 16/44, 4 minute, wave track I gave up. Though I'm saving my pennies for the NOS-1a upgrade, am really wondering if its necessary after what I heard last night!

Brian, engage "Start playback during conversion".

Side note :
Officially this is not the best of course because now the conversion will be commencing when playback occurs. But this is how so many things come together with the NOS1a and the irrelevance of this all then ...

And thank you for sharing.
Will make a 352.8 filter when I have some time. But we can already see (from your post) the permutations occurring ...
:swoon:
Peter

Will try your AI suggestion sometime this weekend. I've started a new position at work (life in prison without the possibility of parole cases instead of death penalty cases) and until they hire my replacement am doing two jobs. Time is precious........

OK, don't forget to pass your new prison address for future upgrades.
Hope you have some space for speakers in there.

:whistle:

Maybe I misunderstood, but I thought that without the new NOS1a driver we couldn't select the Custom checkbox. I just did that, selected Custom,  and got a very different sound from the arc prediction checkbox. I think I like Custom more than arc prediction but am surprised it played. Is there something wrong that I can play music selecting the Cutom checkbox without the new driver?

The only thing wrong would be that you now listen from a different space - prison.

But no, this is not related to the NOS1(a) driver at all. :)

Peter

Thanks, the sound is phenomenal.  :)

Thank you Brian.

I guess 24 bit output versions of the Custom filter are also on the to-once-do list?

regards, Coen

Indeed, it does, well, sort of.  By that I mean that the first time I tried a wav file, it did not work. (huh?)  So I stopped the loading, clicked on [c], killed engine3,   and tried it again.  Then it worked!   Well, again sort of.   By that I mean that for that particular album, only 7 out of 12 cores (hyperthreaded) lit up.  Huh?   why only 7.

So I found the few other wav albums that I have and they had no problems lighting up all 12 cores.   What's up with that?

And playback drive, or not, made no difference.

So, okay, new datapoints.

(been busy and I could swear that I have already posted about this but I don't see it.   maybe it was just a dream.)

Sound impressions of new Custom and AI filters.

Firstly,  I should preface this  by saying that I'm running my Sabre32-chip OS DAC  (Wyred 4 Sound DAC2) with the PCM filter set to SLOW, which except for the sampling frequency +-20% is like having no filter at all almost everywhere.  Meaning: I'm relying on these XX filters almost entirely for the sound.   But also, running the DAC like this tells me much about the filters themselves.  And also, it is closer to an unfiltered NOS DAC which shall remain nameless.

The sound.  These filters sound fantastic.  So far I've only used the Custom 176 filter and AI using only one setting which is with the % pre-post ringing set to 17% (which I imagine is like 34% toward linear phase from 0), plus noise shaping dither.  To various extents, they bring to XX what was missing, which over here was an overly clean sound which sometimes sounded dry.  Not that I ever heard a problem.  It was more that I knew something was missing.  For example, I have some tracks by female vocalists and I know, because I've heard them before, that I should be melting in my listening chair.   But there was no melting.   Bit perfect would solve this but was too rough.  Once you hear the smoothness of AP it is hard to go back to bit perfect.

So I found myself using AP for rock, instrumentals, and male vocalists.   But for female vocalists, flutes, violins, violas and such, I had to turn to AI or no filter, or HQPlayer, and for all of those the melting happened / happens.   But all of these alternatives had their down sides as well.  For HQPlayer and to a less extent the old AI, the soundstage and imaging was nothing like AP.  Big losses there.

Then came these new filters in 186a and i.  We are already well past the melting.  The harmonic richness, which is what was missing, along with a correctness are now extremely satisfying for both Custom and the new AI.   And as expected, to varying degrees.

Custom still sounds a little like AP, and now only rarely exhibits this "something is missing" harmonically situation, but retains the amazing soundstage of AP.   And I'm glad to hear Boleary write that it exceeds AP.   I've not noticed that it exceeds but I will listen for that.  Certainly AP continues to be amazing in this regard and Custom is close, at least. 

The new AI is also a bit of a miracle.  Certainly is sounds better than any HQPlayer filter right now.  That's amazing when you consider the refinement of HQPlayer's filters.   But like HQPlayer, the new AI (@17%) looses a lot on the soundstage and imaging..   But what it looses in imaging it compensates in harmonic richness.     Where Custom can sometime sound a little lacking harmonically,  the new AI never lacks, and perhaps gives a little too much (GOOD for an alternative).  Have a dry sounding CD?   No problem:  just throw the new AI 17% at it.   Also amazingly,  the new AI never trips.  It never stumbles,  and everything seems to sound great.  Oh, except for the lack of soundstage in comparison to the amazing AP and Custom.

Hint hint...  I'd love to see something Custom that was asymmetrical in between this current custom 176 and the new AI set to 17%, more or less.  And okay,  if too much soundstage is lost,  then closer to current custom 176. 

Very very happy to have these filter choices.  Thank you Peter.  You are either very lucky or you are a genius.  It's always hard to tell between those two.

So for months now I've preferred 8x to 16x oversampling. 16x was just too refined and flat sounding. However the new Custom filter checkbox with the "Preferred" 705 setting has not only put the punch back into 16x but the tonal quality is in another league from anything I've heard from a digital system. Entirely natural and balanced. Nora Jones like never before. Thanks Peter!

Now what about hi res stuff? Previously I'd turn off all upsampling for hi res. If I do that now I guess I have to turn off all filters? Haven't tried yet and am just wondering.

Brian, your up times in prison are quite strange. But I don't care of course. Hahaha.

But no, these filters are not much suitable for hires, although you can try it. Well, "suitable" is not the correct word, but at least the roll off will be unnecessary "fast" (should be more upwards the frequency band).
Otoh, it shouldn't be audible if my theories are somewhat correct ...

Peter

Hey Charlie,

You provided there a very good description which will allow me to proceed in certain directions. All spot-on what it's about (in my view).

So thanks a great bunch,
Peter

So I tried to revert back to 186.a last night and could not overcome repeated preset errors. Unfortunately, when I removed the 186.a XX.exe and engine3 files from my XX folder it had presets active. Moving them back into the XX folder all I get now are preset errors and going to a "previous preset " setting doesn't work. Do I have to reinstall 186.a from the beginning or is there a particular preset file I could delete to get my original 186.a files to work?

Brian,

In your XXData folder for each day a backup is made of the active preset file of the previous day. So use one of which you know it still worked for the version concerned.

Let me know whether this helps you.
Peter

Yes, but only after deleting the PresetLoader.XXSI file. Then and only then was I given the option of selecting a previous preset loader file. Hope this makes sense.

Before deleting the XXSI, if I just tried to swap the engine 3 and .exe files, I got the following two errors:

1. Error at initializing conversion from string "to type "Decimal" is not a valid setting: 63, and after clicking OKAY on the error dialogue box,

2.Error opening settings: Preset file

From there I was not given a choice to load a different preset file until I rebooted. But that "choice" happened while rebooting. After the reboot I'd get the same series of errors. As stated above, this was only solved by deleting the PresetLoader XXSI file.

Thank you for the feedback Brian.

Let's say that the whol eprocedure anticipates on not even being able to start XXHighEnd any more. So, that would be a manual job anyway.
Otherwise, but depending on the situation, indeed there will be the auto-question about wanting to load back a backup, but possibly this does not lead to a normalized situation again. What I recall though is that when this happens - after selecting the backup file, XXHighEnd quits and all should be fine. If it did not quit at all then nothing will be fine.

Lastly, I'd say that explicitly using the [...] button next to the Load button (or somewhere there) should replace the file as well. But after that similar anomalies might happen, again depending on how bad the situation is.

Summarized, think of this as a manual emergency situaton. So the replacing how you did it should always work. Thus outside of starting XXHighEnd.

Lastly, I guess that "backwards compatibility" could be difficult (so going back to older version with newer Preset file). I always take it into account (so should be compatible) but probably it is not in this case (of upgrade).

Again, thanks,
Peter

Two things with 186i: sometimes when changing volume during unattended playback the track will "skip", like a needle bouncing on a record, as the volume is changing. One time I had to stop the track because it wouldn't stop skipping. This is very inconsistent as I can often change the volume with no skipping at all. Secondly, I played a 24/96 track yesterday and it sure sounded good! I unticked "Custom," did not tick arc Prediction, and turned off all upsampling. Very nice.

Okay, three things: I'm really surprised that more NOS1 owners, standard and upgraded, haven't posted 186i impressions. The sound on my system is the best ever, again.  :)

Hi Brian,

Change your SFS to attack that skipping (to higher I'd say). If that doesn't help change Q1.

I never noticed such a thing; just looked ... SFS=2, Q1=14, (x), ClockRes=1ms (Core-3-5). Q3,4,5 = 0,0,3.
Of course I needn't worry about SQ (NOS1a) but you can try these settings.
Oh, PC still runs at 430MHz.
And played from RAMDisk (XXHE and Playback).

Regards,
Peter

I can't find where to download it (must be blind).  I suppose I could send Peter an email.

Anthony

Hi Anthony,

I will find it here :)
http://www.phasure.com/index.php?topic=2944.msg31692#msg31692

Alain

Thanks Alain, Peter hasn't "officially" released 186i so it's only been posted in this thread as a "Beta" for those who are particularly impatient.....thanks Stanley! :)

Thank you Alain.  Much appreciated.

Anthony

Hi,
in general it is difficult to get the same result/opinion of  a “software/hardware” test due to the different systems/rooms we are listening with/in and personal preferences.

1)   test of 1.186-i – Custom at 16x AP 
                              
replay of acoustic music/instruments such as

a)   first track on the Brian Bromberg CD
http://www.amazon.com/Wood-Brian-Bromberg/dp/B000FQJPBO/ref=sr_1_5?s=music&ie=UTF8&qid=1404651943&sr=1-5&keywords=brian+bromberg
and
b)   first track also on the Al DiMeola CD
http://www.amazon.co.uk/All-Your-Life-Tribute-Beatles/dp/B00E1SM5UA/ref=sr_1_3?s=music&ie=UTF8&qid=1404652122&sr=1-3&keywords=al+dimeola

There is a very detailed replay with an amazing resolution. A superb metallic sound of DiMeolas guitar strings.
You can hear the picking of the bass strings very, very clear. It seems that the recording microphone is appr. 0,5m away from the instrument. (Brian Bromberg)
I hear less reverberations/resonances of the instruments and 3D image as used too.

2)   test of 1.186-i – 16x AP       
The recording microphone moved away from the instruments (Brian Bromberg), so less resolution (different characteristic of the filter), better 3D image and more reverberations/resonances of the instruments.

3)   test of 1.186-i – 8x AP       
Best overall performance with a seamless soundstage, nice 3D image, instruments (acoustic bass and guitar) have body and a perfect integrated bass.

Joachim

P.S. We have an open kitchen/living room area of appr. 55m², and our listening SPL is around 70dB(A) only.

Thank you for your extensive reporting Joachim.

With your post as a guide I tried 8x normal AP against 16x AP-Custom against 16x normal Ap myself yesterday. Just press Pause in the middle of a track and switch a couple of times.

Maybe I can hear what you mean. But this is my own "verdict" :
(let's keep in mind, I listen to 16x AP Custom now for many weeks (over 8 I think) so this has become my automated reference)

16x AP (normal) brings quite similar overtones in cymbals everywhere which seem right but can not be because too much the same throughout. This looks like *additional* harmonics but in comparison with the Custom filter at least I now perceive it as distortion.
This "distortion" is now more profound with the NOS1a because of the way lower distortion the DAC itself now produces.
Also good to know : this distortion in the really higher frequencies (think about 12KHz) can easily be measured. No worries because this is just normal for a filter serving the time domain almost exclusively.

8x AP (normal) did not bring much of a difference. Notice that I played it in the sequence as you read it here and maybe it was not wise to first try 16x AP normal. So how it works : 16x AP normal brings me not the best (see above) and thus a next setting is to improve on that specifically. But it can't work because nothing was done to the distortion mentioned (it will even have gotten somewhat worse).
So ... there might have been a difference, but it did not want to go into my head because overwhelmed by things more important (as how my brain interprets it).

Now notice that I in fact started out with 16x AP Custom because I was just playing that all the time and started "testing" in the middle of some longer track.

Back to 16x AP Custom;
When it is performed in this sequence, but I think my 8 weeks of using this anyway, this becomes uper apparent :
Richness. So what lacked in both normal AP session -and what I did NOT notice - now is super apparent when going back.

Of course this is the same harmonic richness Charlie talked about and - just saying - it is also 100% logic. There just *is* more HF output to begin with (for level I mean) but is is also "distorted" at levels which theoretically should be OK. Or at least far better OK (I am talking about measuring it all now).
This richness can be put to words better if I describe it as "more vibrant" and then envisioning this more vibrant as the whole left/right experience, in the end making the sound clearly more full. Like a warm bath thrown at you (gently).

For you Joachim, it could be worth while to explicitly watch for this. On the other hand, I did NOT watch explicitly for your remarks because the track playing coincidentally was not right for that. And I actually had reasons for comparing (but with your post in mind).
So here we go again with one of my "sh*tty" albums :

Made in Japan (Deep Purple) - original version (watch out for that super bad remaster);

This is always a test album for me because it is about fairly much smashing on cymbals of which you can hear that there's more in it than shown (through your system). Ever back this started for me with "all grey" because, well, I played the LP so many times it just was grey.
Through CD this is different, because through CD (or computer playback) this is about how much of cymbals come to the surface.
The fun (MY fun) : The cymbals seem to be "over" now. Seem to be, because how to judge well with a 1971 rock album. But point is : since NOS1a this now happens for the first time and where now all the smashing is heard very well it suddenly becomes a matter of "and how to represent those well". This thus while previously it was a matter of "how to bring them to the surface". So really a big difference.

Because the filter adds to the highs I started to A-B. Just the first track (Highway Star).
The 16x AP brought the overtones at always the same frequencies as described and if there's one thing I hate it is "not neutral". The 8x AP did not change a thing to that. The 16x Custom made it disappear again. All what lacks that is just too much of it (in my subjective view of course and it will depend a lot on the speaker). Plus all got so nicely rich and full.

Not much important for this post (say against your post Joachim) but when Child in Time came up ... man, that now shows playing on the ride cymbal I never perceived. So, a ride cymbal (used in ride fashion) always shows the best of cymbals (say 40 years again also) but this time I could see how the stick was hitting the cymbal from various angles (you must have seen drummers doing that in the first place to get that into your bain from listening only). But not only that, smaller cymbals are used just the same for "ride".
Never ever I got a glimpse of this in this track, and possibly I played it 500 times in my life.
OK, I just wanted to share this because it is a typical example of how a certain album or track can show huge differences while others stay close to the same. But remember what I said : I often use this album for testing, just because I can "hear" that there's more in there than shown through my system. And now that finally happened.
(I so often give Sgt. Pepper as THE example of things being in there which don't come out - and today this whole album sounds totally gorgious (believe it or not) ... most Beatles do now).

Anyway, watch for that richness / fullness. I even played a double album of the Beastie Boys throughout. This did not work ever before because too lean. Now it works (but head for a fight with your wife ;)).

Regards,
Peter

Hi Peter,

I don't know if it is because I have been listening to iPod audio at my friends place for the past 3 days or whether the new 1.186-i Custom filters are revolutionary, but since installing it when I got home from a short holiday this afternoon the sound is truly holographic.

I will compare tomorrow with 1.186-d and original AP that I was running before my holiday, but 1.186-i does sound very nice.  

One question though.  If I just select AP in 1.186-i is this the standard AP that I have been running for the past year or so or have you altered it?  In other words, do I need to run 1.186-d again to get a decent comparison or can I just flick between AP and "Custom"?

Regards,

Anthony

Hey Anthony,

Arc Prediction (without Custom) in 1.186-i is just the ever old one.

Thanks,
Peter

Hi Peter,
this/yours is a very good description of the (Beta) “Custom filter”, which is the best with new details “raised” from digital music ever.

Made some additional comparisions between “Custom” & 16xAP, and raised my listening volume from -33 dB to -28,5 dB.

The only thing I’m missing is ambience. ( http://www.stereophile.com/content/sounds-audio-glossary-glossary )

A good sample is the CD “Naoko Live” from Naoko Terai. (3rd track “Black Market”)
http://www.amazon.co.uk/Naoko-Live-Terai/dp/B00005J4AO/ref=sr_1_7?ie=UTF8&qid=1404737519&sr=8-7&keywords=naoko+terai

With the filter “Custom” I’m sitting on the stage and with 16xAP the seat is appr. 25m away.

Joachim

And you like the third row of course. :)
(joking !)

NO, in that row a neck stiffness ................. .

Since most concerts are amplified today the best distance for a good listening position in concert halls is around 20m away.

At present I'm going up the ladder and stranded at 16xAP after a couple of weeks again (due to the rev. a).
Since the Ron Carter concert my priority for listening is ambience. (we do have to live with compromises)

Joachim

Some times my father had to play with microphone (see link in quote) and he hated it. He was also responsible (or made himself that) for the acoustics in the new concert hall back at the time. It just couldn't combine (as he told his 11 year old).

I will try to get that album and try to see wat you mean ...

(maybe I do this only to see whether we can talk about such subjects and whether it is possible to create filters which work out the same for everyone regarding such (ambiance) subjects)

Regards,
Peter

Hi Peter,

I am going to need to spend some more time listening to the Custom Filter to get a good handle on it.  It definitely sounds "warmer" and I do enjoy the resolution of steel strings that it has enhanced but I just wonder if some things sounds a little bit "the same" i.e. not enought variation betwwen the same instrument on different albums.  Not sure at the stage.  All was going well until I put of a couple of "warmer" sounding albums and I was not so convinced.

Anyway, I will listen for another week or so and then switch back to standard AP and report back.

Cheers,

Anthony

Thank you Anthony.

Things sounding the same is never a good thing (as you know from me / my thinking) so I sure would keep the focus on that for now (and I myself will try too).

Do notice though that it is not necessarily easy to judge this for real merits and that it really depends on the instrument;
Think about my example of a very low frequency sine really sounds the same everywhere because it is just that sine, and no matter from what album (hard to say "instrument") it has to sound the same when sufficiently undistorted. To a certain extent this is the same with higher frequencies; the more they tend towards sines the more the same they will sound.

Try to grasp that all the new filter actually does is making the higher frequencies less subject to distortion without added ringing. The other way around : when the higher frequencies are/become *more* subject to distortion, the resulting "tone" from it is very complex with a multitude of variations which can almost be called random. This is not about the wave form itself but what it produces for by-products.
In the end you might be talking about my own experience as written out yesterday, about the cymbals being "over". But not sure at all, so see upcoming post ...

I am almost sure that I talked about it before, but I think the first time I may have scratched (not posted) because it seemed to wild at the time (maybe one month ago). After that I wrote somewhere that the music reproduction became a sort of reference for the drum kit I listen to every day (son practicing). A really upside down thing ...

This will be a similar text to the one I never posted (at least I think I didn't) :

Whether the new filter or whether the combination with the NOS1a, things like cymbals got louder again; I talked about this yesterday.
Louder does not mean more square and tiring. I mean, that can be done too (including over 10 seconds sounding cymbals) but this is also (too) square and no good. Interesting, but tiring.
So what I talk about now is that actually a sort of opposite - a more square cymbal turns into a more higher pitched white. It becomes hash like when you first envision 6dB louder or so. This is how I name it to be "over". It gets too profound and the "white" sound is the same once the hits are louder. Now, I can try to find a technical explanation for it (logically a too low resolution) but my post here will tell something quite else ...

I now have the same with the live drum kit.
Btw frontmost logical explanation to *that* : my ears and all. But let's skip this for now.

So each day over here we also listen to my son's practicing. Not explicitly, but he's on the first floor above the music room - actually to the side of it. Hard to explain, but the room where the practicing occurs separates from the music room by a small glass wall (still this is two different levels for floor). Otherwise no doors are in that practicing room except for the entry door down stairs. All I'm saying is : I can outplay the guy only with well over 90dBSPL playback or better put : this usually doesn't happen so right through the music plays he.
I don't care since his practicing doesn't take ages per day anyway.

Meanwhile it allows me to unconsciously compare. So each single day I have been annoyed how far off the music reproduction still is from the real thing.
And since the NOS1a plus new filter (this happened about the same time for me) this annoyance has suddenly vanished. Also : it is not all so difficult any more to find drumming examples of which I now say "this is not so far from reality at all". And of course people will remember my self recording "complaints" about how only that recording was reall and all else a universe off.
Don't ask me how it can work (in combination) but the both - self recording and normal albums - are now much more closer (remember, the live recording was already totally real).

This is a very long lead-in to my observation that what I hear from the reproduced cymbals when smashed harder, sounds the same from the live drum kit when smashed harder. And what I hardly dare to write : the live rum kit annoys more.

So if now people can think of what is really happening ... please tell me.

But like I said it the other time : somehow the reproduction is starting to be a reference for live. Or already is that.

The drum kit is in an unconditioned room. All "hard". Even glass ceilings. It now has to be so that cymbals reflect everywhere and the sound of them becomes a mess. A hash. White hash. Notice that I am not writing this because I named the reproduction like that in my previous post. It is really what I think when I hear the drum kit, and now in aftermath I see that through the speakers I actually hear the same. Both are equally bad.
But the music room is not conditioned either ...

Anthony, remember us talking about that Klipsch drumming recording and how the cymbals saturate in there (towards the end) ? That. But now in real live.
But for heaven's sake, why couldn't I hear this before ?

I will finish with my ever so much given example of NOT being bothered by cymbal sound through speakers when the attack of them is lacking, UNTIL you only *one time* heard it throug the speakers. After that you'll have brain damage when there's no attack again (but cymbal sounding). This, while you sure know the sound of live cymbals.
This too I can't explain.
So something is happening in our brains which makes things relative to whatever we are used to and won't be better anyway so no problem. And now the last one :

So my music room is not conditioned either. Still the recording rooms will be and now it can theoretically happen that a recording sounds better than live.
Next thing is that I judge the live cymbal as worse than the reproduction.
This really is so and no way that I like to think the impossible (those who say that live sounds worse than recorded should be banned from all audio forums). Still I do.
So is this a sort of measure that I'm finally there ? Of course I like to think *that* but I promised myself 8 years ago that I would never ever think that any more (while back then I did).

:wacko::wacko::wacko::wacko::wacko:
Peter

Interesting, I had two folks from the Philadelphia Area Audio Society in my listening room the other night and they both preferred the sound of standard AP via 186a verses the custom filter in 186i. They felt that the custom filter sounded too rolled off in the HF. Not sure I agree with their assessment.

Another thing, maybe its my imagination, but standard AP sounds different to me in 186a than in 186i. Seems like there is an active filter in 186i when AP is ticked instead of Custom, when plying at 8x oversampling.

Hi Peter,

a description with other impressions, but the same meaning.

The acoustic bass on the Brian Bromberg CD sounds with "Custom" very detailed, unbelievable, but the sound comes from a "deadened" ( evacuated) room with less reverberation and deacay of the instrument.

The "resin" sound of the violin on the Naoko Terai CD is less accentuated.

If you have one CD with an acoustic bass solo and another with a violin you should hear the difference.

The Naoko Terai CD is a terrific funky jazz album, same tracks are good to dance. (Market Place (a Weather Report tune) + Rio Funk)

Joachim

Hi Brian,

I  received a rev. "e" with the upgrage to the NOS1a and I have "acoustic hallucinations" that this revision is superoir to "i". Strange, because the 16xAP filter has not been changed.
So, it's time for a Wodka with Coke and ice (less headaches)

Joachim

Brian,


Only with 8x hence not with 16x ?

Peter

Yes, with 16x too. If I recall correctly when I tried to play 8x I had to go into the filter selection box and select the 176/192 filter in order for 8x AP to play at all. So 16X AP with the 705 filter sounded different from 8x AP with the 176/192 filter and the both sounded different to me to 16x and 8x AP in 186a. Hope this makes sense.

Well Brian, usually it does but this time I have some difficulties with it;

The first half of your sentence - especially without more elaboration - seems to tell the real story : You select the Custom Filter first and next have to select the 176.4 filter in order to let 8x play (this would be correct from my POV).

What you say next is that with same Custom Filter but now played at 16x, this sounds different. This is correct because this *is* not the same filter. Why ? because the AP part is different and it now operates at 16x instead of 8x. The filter on top of it is the same all right, but this is only technically so. For net result it is not because its input already is not (the 16x AP vs the 8x AP).

Still there ? I am. But I only followed your own text. And thus :
When the above is compared with 1.186a it can never be the same because there is no Custom Filter in there of which I just told (reasoned) that you were using that for that first half of the comparison.

ONLY and only when you did NOT select the Custom Filter but still had to go into the settings to chose the 176.4 filter because otherwise it would not play, ... then there I have the sense of you reporting something strange. But this is not what you are saying (plus it would imply a sheer bug I don't recognize).

Sorry for the long text, but ...

?

Peter

Sorry about the confusion. What you said above is what I was trying to say. I did not select the custom filter, was in AP, and was not able to play at 8x until I selected the 176/192 filter. Then and only then would 8x AP play.

Oh boy. Yes, I recall that now. This is a bug I solved by now here. But the effect of this seems other than you describe.

So now I must look in the program what I actually solved and whether you (we) actually activate something after all because of, well, what you describe.
I will do this tomorrow.

Thank you Brian.
Peter

Since hours I'm playing 16xAP (no Custom) with version "i", and a seamless switch to 8xAP is possible (on my computer system).
and the driver panel shows 176.400

Joachim

Edit: Forwards and backwards, and the driver panels follows accordingly

What I said above is WRONG. Sorry Peter. However, what I originally said is correct: the filters in 186i change the sound when AP is ticked and not Custom, and, because those filters are changing the sound of AP, the AP sound of 186i is different from the AP sound of 186a.

Had some time this afternoon and tested this. I was wrong that I had to change the filter in 186i to 176/192 in order to play in AP at 8x oversampling. The filter can be either 705 or 176/192 and AP 8x oversampling will play. HOWEVER, the sound when in AP (not Custom) is very different when you change from one filter to the other. It is very different again when changing from any configuration of AP in 186i to that same configuration in 186a. So, AP in 186i is not the same AP we've been listening to for the past months.

Don't know if any of this matters, but here it is what is happening.

OK, I'll have to look into this Brian.
Thank you.
Peter

Hi,

There are differences in the filter 16xAP (from release "e" to "i").

In release "i" the upper mids and highs are little bit more accentuated, what I prefer.

Joachim

Hi Peter

I've been quiet because I upgraded my DAC to 384,  meaning I was without music and the ability to further test these filters for about 10 days.

I've noticed that few responses have been given with respect to the experimental 705 filter.  I'm willing to try it but I need a version that runs at 384.   Which begs the question:  why not provide 384 versions of the two 705 filters?    ;)

(That was my way of asking for a preferred 384/352.)  :)

I can say that 352 has changed things.   That is, taking redbook all the way to 352 using the 176 custom filter has now resulted in my preference for that filter over the new AI.  At 176 AI had much of what I needed and Custom 176 was a little short.  But at 352/384 custom 176 filter very seldom has that dry sound and can be very clean and not hinting at lacking any critical harmonics.

This means that your filtering strategy is working over here the higher up in frequency we go, finally (something I bet all NOS1 users already know).  Meaning,  perhaps less ringing is needed as we go up in F.   Certainly some is needed though, IMHO.

All right Charlie. I will make "you" a 352.8/384 filter. For now I will do that as "speedy" as I did it with the 176.4/192 filter which means that it will use the same "technology" as the 705.6/768 filter and only optimized for the 352.8 frequency. Makes it comparable as well (sort of virtually).

Thank you again for your reporting.
Peter

After a fresh reboot, I also tried to revert to 1.186a and am seeing the same "preset errors," over and over.  I seems like it never stops so I shut down and went to sleep.   I reverted by copying the "a" versions of xxHighEnd.exe  Engine3.exe and xx.ahk (maybe the last  one was not needed?) over the i versions to see if 186a solves a synchronization problem I'm seeing now at 352 and 384 after I hit alt-e or alt-s in unattended mode, followed by alt-p,  which almost always causes missed synchronization,  and also at random times after my initial playback session ends when I proceed with further playback.

Except in my case, I have no preset file that works with 186a because I never actually ran 186a.  I installed "a" and then I probably  immediately installed 186h to get to the custom filter, which was the reason for upgrading.   Possibly I played one track, or part of a track but I don't remember.

So for me, should I delete the entire 1.186a folder and start again with a new re-install of 1.186a?

Two hopefully quick questions:

1) Is it best advised to reboot out of MinOS mode before I switch between 1.186x and 0.9z-9b?

2) Is it best advised to reboot out of MinOS mode before I switch between 1.186a and 1.186i ?

Charlie, to avoid further hassle for you, I think so, yes.

Not reason to reboot (to answer your two questions).

Sorry for your inconveniences ...
Peter

Thanks.  And to revert from 186i to 186a do I need to overwrite just xxHighend.exe and xxEngine3.exe?   ...or also xx.ahk or whatever?

I have lots of hours up with the new Custom Filter in 1.186i with my NOS1 (not the a version) and I was really enjoying the music today so I thought it would be a good time to check back to standard Arc Predict.

Well, what to say, but Arc Predict sounds 'truncated', like the top end has been removed.  It is flatter sounding and a bit dull by comparison.  There is less jump and less dynamics and I guess that is to do with there seeming to be less HF information.

So wow Peter, I think that this filter is a very significant step forward and I can't wait to see what else you come up with.

Can you please tell me which sampling rates this filter works with or does not work with?  16/44.1 obviously is included, but are others like 24/192 or 24/88.2 etc. included in the supported Custom Filters? 

Cheers,

Anthony

Hi Anthony, thank you for that feedback.


At the moment I would allow to set all parameters by yourself instead of through the presets, yes. But this not assumed (it will be chaos) at this moment it requires more : the selection of the native sampling rate (together with the output sampling rate as it exists at this moment). This is not all that difficult to make.

What will be more difficult for me is finding the merit of such filter (parameter) settings. I mean, I don't think there will be any logic in it for me, so I would not know what to look for (best settings hence best output signal). And I think that chances are very fair that using native Arc Prediction will just be the best for the higher native sampling rates (88.2+).

Notice that a digital filter has two purposes :
1. Reconstruction (really only applicable to 44.1/48);
2. Bringing down high frequency "noise" (same signal but images beyond Nyquist (sampling rate / 2)).

What Arc prediction will do regarding Hires is #2 only. Oh, it will work on #1 as well, but this is quite moot because not needed unless we think that frequencies between 22.05 and 44.1 (for 88.2 sampling rate) are very audibe and need reconstruction too.
So what's left for the Custom Filter is bringing down the HF "noise" more but to a degree which is rather moot (because Arc Prediction at 705.6 (768) output already has it down sufficiently to be harmful (beyond 352.8 (384)).

So it's almost the other way around : When the Custom Filter is engaged by standard (for say 705.6 and anticipating 44.1/48 playback) it should not engage the filtering itself when Hires is in order and leave Arc Prediction as it is.

All again will be different when Arc Prediction is shut off in the first place (see grayed checkbox in the Filter Designer) and now *only* the Custom Filter is operative. But here too, you won't be able to make it yourself because it needs to show the results of it (though this I could graph) and now I also don't feel much like making such a filter for you guys because I just don't see the reason (for the better) for it.

Hopefully this is not too vague (I did not try to be this time :)).

Regards,
Peter

I think that this is true for 705K.  But don't forget us peons who can only go to 384K!  :prankster:

Charlie, the "say" was meant to be an example. But I also think that maybe you didn't get what I meant. So for 352.8 and the others it would count the same (don't engage that filtering). Tell me if it is not clear.
Peter

Let me clarify what I understood because I think things are different at 352/384 than they are at 705/768 and I think that even hi res needs help.  I understood that you might make a custom filter that behaves like AP when the source is hi.res. and like Custom (how it is now) when sourced with 44/48. 

What I'm saying is that even with hiRes source material, at 352/384 my ear is much preferring Custom over AP.  Hard to argue with the ear.

As an aside, I now hear the buzzing / zooming / mosquito-ing of the typical upsamplers that you've talked about.  I also hear that AP never has it, nor do the interpolative filters in HQPlayer.  With Custom, I can *sometimes* hear a little hint of the buzzing.  I accept it as it comes with a bunch of other goodies.  But usually when I hear it with Custom, it is in the recording itself; and I presume this is from the EQ processors that audio engineers use for vocals.  No doubt they ring, some more than others.   And I can check with AP for those recordings and sure enough the ringing is there for those recordings.  It's not a clean ring when it is in the recording but the essence is surely there.

Anyway, I hear the trade off that you are making with these filters and can see that it is like walking a tight rope, picking the right balance between the two extremes.

I can say, it is amazing that Custom measures better than AP because AP sounds cleaner.  But sound is sound.

All right. By now I may think differently again and that even Hires should be filtered in the "non-AP way". So no worries.

Also : Because at this moment you can only use the 176.4 filter while using 352.8 this will ring unnecessarily.
Nothing wrong with your ears !

Peter

An attenuation of no less than -6db has been recommended with the custom filter.  In my case I'm talking about the 176 filter, used mostly at 352 and also at 176.

For sure audible clicks are heard at 0db.  For a while I ran at -3db of attenuation because I'm running deep class A on my ams and due to heat I have limited headroom.   The purpose if this post is to confirm that -3db is not enough.  When I run at -3db I don't hear clicks, but it is possible to hear things that don't sound right.  Further attenuating to -6db solved that problem. 

It's still hard for me to believe that the intentional ringing is reaching with peaks of  200% which corresponds to -6db,  but it appears to be the case that 141% which corresponds to -3db is for real.   

I've not tried -4.5db and may never try it unless I actually need the headroom / volume.

This is regarding the 176 Custom Filter, run usually at 352.


Correction: The quoted text above (from me), while true sometimes, is not true most of the time.  It depends on the compression level of the recording.

I've been running three weeks now at the recommended -6db and xx has turned out to not be my favorite player in this time.  The sound became drier, even with the custom filter which had solved the dryness.  AI sometimes came to the rescue but takes too long to load to use every day. 

So yesterday, I came across a dry sounding situation that was only dry with xx and not with other players.  Hmmm.  This was on the highRes version of the Indigo Girls All That We Let In,  This recording I know has peaks that never go above 90%, and I was upsampling from 96 to 384.   So I changed the attenuator from -6db all the way to 0db.  The dryness was gone!!!   :veryhappy:

The same is holding true for most recordings that I've tried since then.

And so it appears that at -6db there is less "ringing" going on, and less reconstruction of the higher harmonics, than at 0db.   

My standard setting right now is -1.5 db and if I hear anything funny I drop to -3 db or -4.5 as needed.  The more compressed the recording, the lower I must go.

Thank you for your effortless sharing Charlie !

Peter

I've been trying many different settings for the new AI, including all of the dither options ...including none.

At first I was trying to get an asymmetrical filter to work out.  I was trying values between 17% and 5% pre to post ratio.  And while an asymmetrical filter can sound good,  the best "new AI" settings I've found is for a full minimum phase filter with straight TPDF dither.  The settings for this are 0% and then selecting TPDF from the dropdown menu for dither.

I wish the divisor for the filter bandwidth were active,  I'd try opening it further if I could.

And yes for dither straight TPDF is sounding better than any of the noise shaped variants.

I am sorry, but what do you mean - "divisor" ?

Hi Peter

Sorry for the delay.  Busy week.

There's a bandwidth or slope setting,  I don't have it in front of me right now.  As I recall it shows Fs/2.  It appears to be clickable with a drop down, but it isn't.  I guess this is grayed out.   I was hoping I could change the value to something like Fs/1.8.   

What I'm really trying to do is open the slope or control the slope, either the starting point, the ending point, the rate, or all three.  One interesting one would be starting from 16.3K and ending at say 22.05.   Another would be starting at 20K and ending at 24K or 30K (under the assumption that the original AA filter at the mastering stage did its job, as an option anyway).   

Why?   Because to my ear 48KHz Fs recordings sound so much better than 44KHz Fs recordings.   48 can sound almost as good as 96 or 88, just shy, or sometimes indestinguishable.

And okay, other ratios may be better than these.

I acknowledge that "opening" the filters does not always work.  For example, DSD sounds best with the lowest setting in my DAC, denoted by "<50KHz."   Meanwhile the chip default curves are 50, 60, and 70 KHz and none of these are acceptable.   One reason is that the quantization noise in DSD is full scale.   Whereas in PCM the quantization noise is over the last few least significant bits.   It seems that PCM has the advantage for this type of added noise.

This happening now fairly consistently and therefore often enough to report.  Regarding the newAI vs. the 176 custom, I don't know how this is possible but it appears that the 176 custom filter at 352 can have more buzzing / zooming than the newAI filter using 0 (min phase) and TPDF as its variables.   This is not supposed to be the case.   Very strange.

As always,  ArcP none, or whatever is in the recording.

[BTW,  if it ever mattered, only now is my Sig remotely close to what it has been for months and most of this thread,  except that wasapi is new within the last few weeks.  The one thing I'm quite certain is inaccurate is the Q5 setting; it might be 5]

Hey Charlie,

Just to let you know : always reading.

I must say though - it gets quite messy for me. I mean, not at all that you don't make sense, but it becomes mighty difficult for me to make sense. There's just too many things different from what I do or have set;

The first example would be the OS which is W7 in your case and to which I most definitely can not listen any more (just one pile of distortion).
And if you *then* start about "more buzzing" with the custom filter opposed to a most-ringing setting in the AI filter ... well you wondered yourself of course, but how to help out now ...

Something else - also not making it more easy - I myself am back to normal Arc Prediction. Maybe not forever, but at least since 3-4 weeks by now. The "whether needed" (Custom) seems to depend on external factors I am not conclusive about yet. But an example : change interlink which does not work out until you move to the other filter.
What's possibly going on here is that "illegalities" in the filter (AI being the most legal of them) *or* show, *or* are filtered by the interlink. Or the other way around : when filtered too much it may require normal AP to push back some of the (square) detail.
Just far away thinking; nothing formal.

:scratching:
Peter

A simple hypothesis would be interaction, ..with the ringing already present in the recording.  

And so the most natural sounding interaction is the one that goes unnoticed.

This would also explain why it is so difficult to pick a favorite filter amongst, for example, Miska's 15 or so filters in HQPlayer.  All sound different, and all eventually lead to a situation where a change to a different filter improves.

I found two good examples of where newAI filter using 0 (min phase) and TPDF as its variables sounds much better than the new Custom.  I post these two titles so that you may try them for yourself if you happen to have this music in your collection.

The first is Traffic Far from Home.  This is in the class of popular or rock music.  Here, using Custom, Steve Windwood's vocals sound as if only midrange is present.  Assuming your system might reproduce what I'm hearing, it sounds almost as if he is singing through one of those hand-held "megaphones" or institutional outdoor speakers used for public announcements.  For sure the feeling that something is missing.  Now switch to AI.  With AI it sounds like a professional recording, proper, with an entire range of high frequencies blended in with the midrange.  It sounds like a real person singing.

Next is Antal Dorati and the London Symphony Orchestra, as released by Mercury Living Presence Léo Delibes Sylvia & Cappélia, Disc 2.  This is classical music and there are all sorts of different stringed instruments.  The focus here is on the delicate sound of the strings.  With Custom, it sounds good;  but a little bit like a CD, which it is, and so has a little bit of that digital hardness.   Now switch to AI.  With AI it sounds wonderful, like real strings.  Forget CD sound, it sounds like HD material or SACD.  There's just a beauty to the strings that is not there with Custom.  Folks,  it sounds like a live concert, very smooth, very analog, and with a very nicely balanced high end.  Further,  with this recording, there's no buzzing / zooming or distortion as I'd normally expect with the extra ringing.   

Both were played at volume levels that sounded best for each:  -3 db for Custom,  and 0db for AI.  (And for Traffic I think it was -4.5db for Custom)

Because it takes so long to load AI at 8x, I tend to use Custom.  So making the above realizations can take some time.

Hi Charlie,

With these two clear examples I am surely going to try.
Next up would be to incorporate "AI" in the Custom Filter (settings), although I can't tell at this moment how difficult that would be (read : when ever I'll have the probably long time needed for that).

Thanks once again ...
Peter

In contrast, this album The Very Best of the Manhattan Transfer plays by far the best with Arc Prediction upsampling at the highest sampling rate I have available, 352.8.
It plays hyper cleanly with arc prediction, very analog, and the vocals sound lush without the help of any added ringing/reconstruction. 

With Custom (176 @ 384) it sounds very good, but if you focus on the vocals there is clearly some buzzing going on from unneeded reconstruction.  So this recording has plenty of post processing on the vocals to get a certain sound.

With the newAI (set to 0, TPDF) this album plays with clearly audible distortion in the vocals (only).  There is too much ringing / reconstruction going on.

----------------->  Not tied specifically to this album:

I've noticed with these three upsampling methods that the vocals climb in the mix with added ringing (more added harmonics) and fall in the mix with less.  In one extreme we have newAI with the most ringing and where I perceive the vocals as louder.  In the other extreme we have arcPrediction with no ringing where I perceive the vocals a little lower in the mix.  In the middle we have Custom.

What's interesting to observe between these methods is that, because the ear is most sensitive in the vocal range, I tend to adjust the volume based on the vocal's loudness or perceived loudness.  The minimum step on my dac for volume is 1db.  So in practice, after I adjust the volume to compensate for the vocals, it is the band / instruments that vary in the mix.   The end result is that with Arc Prediction I get the most "band" in the mix, with newAI I get the least amount of "band," and with Custom I get somewhere in between.

This is all very subtle.   Even though subtle, it can make a big difference in perception and overall experience.  With more band from arc prediction can come the  perception that arc prediction brings the best sound stage and imaging (which appears to consistently be true).  At the other extreme we have the newAI.  Custom comes in very close to Arc Prediction for perceived sound stage, usually.

Awesome description ...
!

Hi Peter

Whenever it comes, I'm happily anticipating a 384KHz Custom Filter.  And even if it is beta, I'll gladly try*any one*.   This is because the 192 is not right at 384, with too much buzzing, and with ArcP being too much on the clean / dry side most of the time.   

Much has happened here with my DAC; I've had to return it due to that same initialization / synch problem, and so the fixed DAC works right  but I've had to break it in a second time and so it has been a moving target.   

I was previously running with the slow roll-off filter and now that has changed as I'm now back to the brickwall on the DAC.   It sounds better right now.  And believe it or not the best sounding filter within xx is back to the new "AI" with minimum phase settings and TPDF dither.  This could be because it is a preprocess or because it is at least tuned to 384 and maybe I just can't hear the buzzing because it is too high in frequency.  Along with this, Miska's asymFIR filter at 384 is the best sounding while xxHighEnd is still the most impactful by far.

But I highly suspect that any custom 384 filter in XXHE will be, both  the best sounding and most impactful. 

Bring it ON!!!! :shout: :derisive:

Hey Charlie,

I looks I just finished a very long "project". The filter stuff has always been postponed until the end, which should be about now. So first I wil make a 384 filter, especially for you and because you have been so patient. You will receive it as soon as I have finished it. Only you. :)

Peter

Thanks!

So ...

I think I just spent 3 days of tuning filters for 768, 384 and 192 (not for 96). All tested with 16/44.1.
And that was even by automation (let the program change the filter parameters automatically and meanwhile watch the analyser).

All with the least of ringing and all without changing the phase anywhere.

And the change in SQ ?
hahaha ho ho ho
Not a downside that I can hear (all is as accurate) - only upsides.

Peter

I guess we can look forward to a new upgraded Xxhighend soon!!!

How will you find time to make the Clairixa's?

Robert

Awesome.  Looking forward to hearing your filters.

To anyone subscribed to this topic, there is a new 2.01 version that now supports these filters.

In one word, they are UNBELIEVEABLE!!!!!   :soundsgood:

[Edit: Only as of May 31, 2015 are my settings accurate in my signature lines below,  in case anyone is interested in my settings.  They are important.]

 :) :) :)

Hi Peter,

Is it possible to know your settings within 2.01.  :)

Joachim

Joachim, you are right - I must update my sig.
I will do that soon.

Regards,
Peter