Hey hey ...

A:B-W-Y-R, B:B-W-R

This has been the configuration for many months. Maybe from of today we can change it to this one :

A:B-W-Y, B:B-W

with thanks to kurb1980.

Unless it is me or my ears or electrical charge from outer space ... this is such a refreshing one ... hard to believe it is real. Many things in there I never experienced before.

First odd thing : I never noticed a real change on day one of listening to this. Maybe the wrong music for it, hard to tell. But this goes in combination with day two, where it was one large jaw dropping experience (this was yesterday).

Odd: Bass sounds like with too much energy, but at second thought and "seeing through" it is about a layer under what you here foremost, that layer implying extra-low frequency.

Odd: When walking through the middle of the sweet spot (crossing the room from left to right and back), in the middle there's a clear increase of volume. I did not check with the SPL meter, but I guess it will be easily detectable. Anyway, like all adds up on the sweet spot.
I know, this may be logical to happen, but I never noticed it, and the increase is really quite much (feels like 6dB).

Odd: The depth of the sound stage is as flat as can be.
Huh ?
Yes, but in combination with a sound stage which never has been wider.
Huh^2 ?
Before trying it out, try to envision that. To me it seems impossible from theories. Still it is so. There is no depth but there is infinite width. And the most troubling issue : I can not detect where this bothers me. I mean, the speakers are not detectable anyway (and that with this flat stage ??) and it just is a most pleasant presentation.

There is detail in an again lower area which brings forward all kinds of sounds and voices and mumbling and a 100 things I never heard before. This also does a lot to cymbals and percussion instruments. I am also confident that this attributes to the (music) reality level ... not-normal.

But supposed you know my way of thinking, something quite else is going on ...

I (or we) already know of what happens to the "accuracy" as such, when the stage is made wider by what I call unnatural means. The sugar cubes are an example of that. You end up in a church but in the end all is more vague. The idea ? the energy in the sound waves is spread (by whatever voodoo force ) and therefore all becomes less accurate (it is torn apart). Get the idea ?
All right. This Lush^2 configuration works a kind of other way around;

The sound stage is wider, true. But the reason now is : all is compressed in a flat space (if I say it is 1 meter deep then I will exaggerate). And notice me dealing with the "energy does not get lost" law of physics. What I plainly hear is this :

All the energy is in this tight space (you could do the math on it, and compare with a depth of e.g. 3-4 meters as how it was previously). There it fights for space. This is the buzzing of the bass but which only happens when not listening at the right position. This is also related to the 6dB (?) of higher level which is perceived when being right in the middle (of the speakers) - hence at the sweet spot again. All there beams as should, comes together where intended and "adds up".

If by now you think I really lost it, then be happy as it is and leave it be.

All the again extra detail which is there now, comes from the far better accuracy because of the more compact sound (more compressed ?). Much more collides and much more literally energizes.

Warning : I have the clear idea that this now requires the Custom Filter Low setting, although I did not try plain Arc Prediction (each second of trialling things feels like a complete waste of time - haha).
The other and more important warning is about letting this config be for a first day and only judge at the second (it is too hard to imagine that I just missed this all on the first day).

I noticed that it was impossible for me to judge the quality of the "sound" because the music itself attracted too much (this is always a good thing, IMO). However, I was continuously and throughout pointed at the new sounds I heard everywhere. In amount, think twice as many (I am serious).
I coincidentally started out with some old hit (Turtles - Happy Together (1967)) and was open mouthed from the super clear sound and how undistortedly beautiful the lead singer brought this song to me. A-ma-zing. And with a channel separation of which I readily knew I never heard it before.

An other odd thing is that while on one side there's super speed exhibiting, there's an unsurpassed palpability at the same time. Listen to the snare drums and you'll understand what I mean. Full with body, never sharp and just how it sounds in reality.


Peter

Coen, you tempt me ...

But many years ago I worked on it for as long as was needed to find out that this is about a deep down Windows driver bug. This, with the notice that KS Special Mode is highly illegal to begin with, so there really won't be anyone feeling compelled to solve that bug. And when the driver is used in normal fashion, this bug doesn't surface ...

Peter

No imagination, I think. Btw, if possible for your DAC, try to use KS Special Mode. If it works without ticks or buffer errors, it is "the best" (sadly not possible for NOS1 users).

Anyway, you are completely correct in your stance that the upsampling changes the buffer sizes inherently, etc. This is exactly why I created the NOS1 such that electrically this all does not matter (and the software cooperates  ).

Also notice that most DAC's will let themselves "overrule" for the filtering, hence, an e.g. 88.2 will not engage filtering for 44.1 (read : will not roll off under 22.05). This means that "our filtering" will be effective in maximum fashion. But if the DAC does not understand this way of working, then the overruling merely becomes "messing with". And then you will have quite contradictionary filters on top of it all. This is why a DAC actually should be Non Oversampling; now we can guarantee that our filters will be applied for 100% (not messed with even the slightest).

And oh, notice that each Q1 setting below zero, is a special application for the WASAPI engine. Maybe set xQ1 to 1 and Device Buffer Size to 1024 (I recall that ever back I made it for these settings and I forgot whether this is all taken into account). Minimum of Q1 for WASAPI is -4.

Have fun !
Peter

Hello Zheng - Happy New Year indeed !

I solved this a few days ago. All you now have to do is wait for the new version.

Btw, I think it is not commonly known by others which recognize the issue as "the last track in the playlist", that this happens at a "Format Change". So in the middle of the playlist it can happen just the same, if only a next track is of another Sampling Rate than the current one. The current one now will cut (and for how much depends on the SFS).

Kind regards,
Peter

Bert, Yes, you can very well be right. I already forgot about your description but had in mind the advised Custom Filter-Low setting and yesterday from the first second it all came to me as more musical. Maybe this does not 100% cover what you are saying, although it could when thinking of the "too technical" you refer to. Along with it, I have the idea it is more pleasant to listen to, while I not readily noticed downsides. Maybe a tad less spoken bass ?

Thanks !
Peter

Hi Nick - I am sorry, but both end up at the result of 150, and I am not aware of any possible SQ influence by means of the individual settings of each.
This is for Kernel Streaming. For WASAPI the story is different ...

Best regards,
Peter

Although this would be true (like "Event Mode" or whatever other software calls it) is not in there at all. It is just (indeed) that mere raw mode which existed only at the very beginning of WASAPI. But that should not matter.
What does matter is the denotation of DIYinHK at the driver, which tells that using the highest speed is at your own "risk" etc. or whatever it tells or ever back told about it (maybe this changed). So this is (or was ?) some self-implementation of the IIRC XMOS driver and it is tweaked or tricked. So yeah ... maybe that is it.

On an other note, be aware that not all software implements WASAPI in Exclusive Mode (and IIRC JR does not or at least allows the situation of "not"). This means that resampling may take place if a sampling rate is not supported.
If Foobar did not change things, then that would be the better candidate for comparing because Foobar does not (did not) allow working outside of Exclusive Mode (same for XXHighEnd and that thus can incur for the culprit easily).

Please be aware that Q1 is quite crucial for WASAPI to work (and that the error in particular indeed can come from this). At getting it to work, always have xQ1 at 1 and best is to have the Device Buffer Size at its default of 4096. The default of Q1 is 14 (so start there) and vary it with 7 and 20 (not everything between it). If this does not help, set the Device Buffer Size to 1024 and try the 3 Q1 settings again.
If that again does not help, set back the Device Buffer Size to 4096 and never look at it again (it probably does nothing these days, except for being part of the calculation which Q1 x xQ1 also is part of (think like Q1 x x Q1 x DevBufSize).

For WASAPI this is all about timing and how the endpoint device (DAC) talks back about this. So if that does not happen for this particular rate ...

On a not unimportant side note : I can't see per the logging where it realy goes wrong or what part of the code is executed. Maybe this is because it has been 100 years by now that I used WASAPI - maybe this is because I removed redundant code. But the fault and abrubt ending in the log file I don't recognize.
Btw, it should be so that you get one or two more error messages, one of them showing a sequence of numbers. Am I right ? if so, can you give me a screencopy of that ?

Peter

Is it possible that you send me that track (via FileMail etc.) ?
I don't think it is the track, but I need to be sure. And then I compare the logging with what happens over here.

Hi Maxi,

Right from the beginning with XXHighEnd (say within the year) it became obvious that once Q1 (which in the end also is a buffer somewhere) became too low, people observed an inside-out phase behavior. I guess that never changed (although today it would be about going from the higher SFS to a more extreme lower one).

The other day I also encountered such a sensation, and because it is less than a week ago (as far as I can tell now) it could be the same experience (as back than, as yours ?).

Why this happens is beyond me and all, but I suppose it has to be related to a more acurate representation of "the sound", whatever that exactly is. But if I observe the stage to be wider (more extending from the speakers) then in my book this will be about some form of accurace (better pinpointed "waves"). And that in itself could show the inverted phase better.

Point is (to some degree) that a proper absolute phase hardly can exist. This is mightly difficult to explain, but is already related to the response of mechanical and electrical devices in the recording chain, as well as in the playback chain. So if someone hits a drum, what part of the wave do you think is the first one with the highest excursion ? I say : this is random and depends on many things not under any one's control. Of course, the first wave front should be positive (more pressure) because in realtity that was so. But whether this shows in e.g. the digital file ? ... at random. And when this appears to be negative, then bad luck - then it (literally) sucks.

Point of course is that if one person hits a drum and the bass player is supposed to pluck his low E or whatever at the very same time, but should be of equal phase, at least at the attack. But there is already no way that will happen because of the behavior of the two too different instruments.
The same absolute phase in the sustain phase then ? Well, even with e.g. two bass players with the same bass instruments that would be the most difficult. If they are the slightest out of tune then within the second of that sustain they are out of phase.
Etc. etc. etc. and we don't even talk about how the various recording devices are electrically wired (half of them could work in opposite phase and if not half than maybe one, like you have the idea you have it wrong and we all have it right).

So that is my personal idea of the existence of it ...
... Which does not prevent from what you perceive is a generally known thing. If people only did not forget about it. And if what happens does not suit you, well, then you invert the phase ...

Regards,
Peter

Maxi, I tested this and had the impression that the sound got a bit more "brittle" over here. I am not sure though.
So as it looks, the other way around from your experience. And if true indeed, than one of us can have the absolute phase inverted (which easily can be me).

But really, I never had the idea it could be permanently audible for me;
I will be experimenting some more with it. At least it is interesting ...

Kind regards and thank you again,
Peter

Thank you Georg. I don't think I was ever able to test that (the NOS1 uses a special connection to achieve that sampling rate).

Best regards,
Peter

Hi Numlog,

Can you switch on logging (at the bottom in Settings), kill XXEngine3.exe (click blue led in the right hand border), press Play on that file, let the error appear on screen, do nothing further and now grab tha latest X3 and X3PB two log iles. You can attach them on a post in response.

Btw, I am not 768KHz will work in XXHighEnd by other DAC's than the NOS1. So maybe recheck if that really works, because I have doubts about it (and then the general issue is that it was never made for normal DACs doing 705.6/768 - which in the not so far past just did not exist).

Peter

Maxi, you mean the effect of Inverted Phase was not to noticable ?
If that is indeed what you mean then I will be very interested in tonight's playback session. It would indeed be so that it would be the first time that it really is obvious(ly noticeable).

Thanks !
Peter