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Author Topic: 0.9y-4 - w00t  (Read 122742 times)
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Fidelio
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« on: October 06, 2009, 09:46:41 am »

Sorry, but fucking hell Peter clapping

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

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

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

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

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

Mani.

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

Where will this end ???

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

 unhappy

Dreaming about a 24 bit DAC.... aggressive

 Happy

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

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

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

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

Peter


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

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

Hey Leo,

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

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

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

Peter


* FirefaceDDS01.png (10.6 KB, 409x528 - viewed 1548 times.)
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A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
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« Reply #7 on: October 06, 2009, 10:19:48 pm »

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

OMG,

Peter you did it again,

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

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

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

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

Nice gadget that album counter Wink

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

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

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

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

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

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

Wow P, nice sound with the new ARC pred.

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

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

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

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

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

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

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

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

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

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

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

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

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

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PeterSt
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« Reply #14 on: October 07, 2009, 01:23:02 pm »

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

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

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

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

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


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

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

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



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

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

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

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

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

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

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

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

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

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

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

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


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

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

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

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

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

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

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

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

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