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Author Topic: 09.U-0a setup with external USB DAC  (Read 62626 times)
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edward
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« Reply #45 on: February 28, 2008, 07:35:38 am »

Quote from: Steve N.
24/96 files are handled correctly by Vista providing you set 96kHz in the audio options

should I look into how to change "set 96kHz in the audio options" and run the test again on XXHE

No Dave, what Steve is referring to only applies to "shared mode". Changing your "shared mode" options will have no affect on Peter's "exclusive mode" test (nor XXHighEnd).
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« Reply #46 on: February 28, 2008, 09:29:48 am »

If USB specs tell that 48/16 is max, I don't see how a reprogrammed receiver chip can do anything about that. UNLESS you're not talking audio, but "data", which just is possibe too (at playing audio, might you know it or not ... my question is about that anyway).

sleeping

Who said anything about USB spec being limited to 48/16? That is false.

Terje also said:
Quote from: Terje
-Sound card of PC/notebook does not export more than 48kBps through USB
 or SPDIF whatsoever.
and...
Quote from: Terje
no sound card can shoot out 24/96

He said 48 kilobytes. (Which translates to 384 kilobits/sec) That would barely cover MP3s!!
Just look at the properties of a 16/44 song or 24/96. You'll see that they are 1411kbps and 4608kbps.

Where the heck does he come up with the idea that no card can shoot out 24/96??

Ok. It was late and I read Terje's post too fast. I read 48 kBps for 48 Hz, I read USB ship for USB, didn't understand the last half of hist post (soundcard output), and I guess I wanted to read what my expectations were. WERE.
Also, it is kind of confusing for me with a TwinDAC+ that does 96/18 over SPDIF and 44.1/16 over USB.

Obviously the speed of USB (even 1.1) is fast enough to do it, that's why I asked for "data", or maybe I should say asynchronous.
Also, I thought I read somewhere that indeed usbaudio.sys just doesn't allow for more. Well, this *is* all over the place, but for other reasons than natively not being able to.

I read a bit this morning, and if one thing is clear, it's that more people do not - than do understand these matters.
Also, by now it is fair to say that people must be very careful what they actually listen to, while Foobar just proves you get fooled without notice. This is not so difficult to see : when my DAC does 18bits at most, how can Foobar just play the 24 bits file ? How do people test it ? (and I used the 16 bit connection).
If your DAC actually allows for 24 bits, you *still* won't be knowing what e.g. Foobar makes of it ! Foobar, Vista, USB receiver chips ...

A response like that of Steve N. above says it all. He is asked a question by someone who knows what the culprits can be, is fed with the right stuff to test(ify), and the answer ... well ... shows that this DAC representative doesn't know anything of what is going on. Not to be harsh to someone like him, but the message is : you won't be knowing what you're listening to when you have to deal with these people who *do* make DACs but *do not* know about them. Well, to be honest, about the impact within the systems (OS) they are applied.

Mind you, it was less than one year ago that people like Gordon R. just couldn't tell himself what USB actually allowed and it may have been the Elias G. intervention that made others believe it was actually possible (96/24). Never mind if I twist some names or sequences, I only want to say that the TAS1020b wasn't doing the job either, without mutual exchange of knowledge on how to deal with the buffers properly. If I'd apply that chip but forget about the dedicated buffering I'd have 48/16 only.

Now what ?

I can tell you, I wouldn't care less. That is, about 96/24 files which -so far- hardly is provided in my music anyway. More importantly though, I wonder whether I'd really hear the difference. Pedal's post above ( http://www.phasure.com/index.php?topic=387.msg2746#msg2746 ) says some things maybe too, or anyway I recognize all of it. One thing though I wouldn't agree upon much :

Quote
On the other side, I think we can't have a high enough sampling frequency. Not because we can hear much above 20kHz (which is the limit for most red book DAC's) but because we can hear the sideeffect of the very steep lowpas filters at 22kHz. Pushing the sampling frequency up to 96k is certainly much better, soundwise.

... which is related to the technical impact of 24 bits vs. 16 and the derived functionalities from that. I refer to digital volume now which for me is just "under spec" if I had 16 bits only (and which is related to gain vs. sensitivity of speakers) and which is on par with 18 bits. But that's me and my system, and people would really need 24 bits to do it right when having more gain and less sensitivity.

On another note, aren't a few of those 24 bit supporting DACs non-oversampling ? If so, how would this happen while no ladder (or similar) DACs exist with 24 bits ? dntknw
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« Reply #47 on: February 28, 2008, 02:37:02 pm »

Terje, it's wrong that soundcards can't deliver more than 48Khz on S/PDif. My XFi Elite Pro can deliver 96Khz out (and does) and so does a lot more cards that I know of. Of course, cheap on-board solutions have very weak software solutions often not giving the user the possibility to change the frequency of the bitstream manually and to above 48Khz.

More expensive soundcard solutions have this option.

As an IT Technician, my point of view is that USB should not be used for audio at all...with it's inherent flaws, noise etc. Many implemented USB solutions also have a transmission roof of 48Khz. The speed of USB is not really suited to any more...

If you're wanting to use e.g. 96Khz, use Firewire or S/PDif IMO.
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« Reply #48 on: February 28, 2008, 07:02:29 pm »

Quote
-The USB chip can not receive more than 48kBps (this is USB specification).
-Sound card of PC/notebook does not export more than 48kBps through USB
 or SPDIF whatsoever.
Sorry for not writing this properly.
It should have been 48khz not 48kBps. This is usb receiver chip spec.
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« Reply #49 on: March 02, 2008, 12:07:45 pm »

Dear Peter,

See DAC test below. Not much to brag about here...
My Eximus DAC is manufactured by APRIL, same company as maker of Stello DAC. Probably they all perform equally regarding digital input capabilities.

What does this means SQ-wise when using XXHighEnd? -Are we missing any sound quality [WHEN PLAYING REGULAR WAV FILES RIPPED FROM CDs - which is probably 99% what all of us are doing till now] because of this? Please advice. Several XX users have Eximus/Stello DAC's, so I believe this question is important.


* DAC Test_ Stello Eximus CD10.bmp (331.04 KB, 458x370 - viewed 1101 times.)
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« Reply #50 on: March 02, 2008, 01:09:33 pm »

Hi pedal,

(btw, is there any importance in the picture underneath your text ? I see a read cross only ...)

The answer is not all that simple (for understanding) I'm afraid;

Let me first say that if you do not want to use the Digital Volume from XX for your own reasons, for 44.1/16 playback it doesn't matter at all.
The point is in the if though ...

I determined that applying the digital volume is better than the analogue volume. Never mind why for now, but I reasoned it out en so far it looks ok. Additionally this - by accident - allows you to get rid of the preamp. And, as this is a major smooth-operator Happy holding back transients, creating noise and all, it slowly becomes a matter of "needing" the Digital Volume to just have a better sound quality.

Btw, that a preamp holds back and therefore better is to be avoided is notthing new, but -so far- we couldn't really see how to do it right with the ditgital volume and really too few bits available for it. Now, what is applied in consistency in XX is that with 16bits input and 16 bits output all is - or just is not sufficiently legit.
Btw, keep in mind that with 16 bits in (say, the CD) and 16 bits out (your DAC as it sadly seems) everybody on this planet will say STOP !
And when nothing "special" is applied at digitally attenuating, they are very much right ...

And mind you, all who apply official digital attenuating (like in the pro world (which includes RME and the like) will do so by means of internal 40 bit or more float processing, and XX does nothing of the kind ...
With this I only want to say that we're way off the usual tracks here, and my own theories including the practice as of now, tell me it just can be done. Ok, just, or just not;

As I said before, the just / just not is highly related to the gain you apply and the sensitivity of the loudspeakers;
At first, when I couldn't exploit my own 18 bits of the TwinDAC+ (XX could not do that yet) I determined that as long as I kept above 48dB attenuation (so, towards 42, 36) all was okay. But mind you, this was absolute listening without the possibility to compare the 16 bits DAC (useage) to the 18 bit DAC. I thought I could hear anomalies at 42dB, but I took them for granted NET. Together with that I would never play that soft, and it's merely -30dB or -24dB. At -30dB too, I again I thought I couldd hear the anomalies, and again I took them for granted, and again NET.

Please keep in mind that there is a "balance" between more attenuation and the SQ getting worse on one side, and the less attenuation and less distortion (that's what it plainly comes to), or IOW the louder you play the better you can hear distortion obviously, BUT the distortion by itself gets less. At 0dB there's no distortion left on these matters ...

With the repeated "NET" I refer to the overall improvement, and *if* I can hear distortion somewhere a. I'm not even sure it is "that" distortion I hear and b. because of a. and the overall better result, I don't care.

Mind you, this is all with the 16 bit DAC.
With the 18 bit DAC though, the distortion (if audible at all at the given level) officially comes forward 12dB "later". Thus, when with the 16bit DAC audible distortion is there at -42dB, with 18 bits the very same distortion would be there at -56dB. Now here the mentioned balance comes in again, because when you normally play at -30dB, there is no way there's audible sound at all at -56dB. Try it (not with your ear in the speaker, but at normal listening distance).
To make a long story short, I am sure that I hear the difference between the 16bit (used) DAC and the 18 bit DAC both at -30dB. Also, when you A-B this, it becomes more apparent what it actually is you hear with the 16 bit DAC and which you took for granted. It's a kind of roughness that I can *not* really explain from missing bits as such, but of which I know by experience that they are needed to fill out "holes" at certain frequency levels (or better : from certain instruments). This is similar to how I eliminated "roughness" by just adding gain to the highest frequencies in my loudspeaker filter (which lineairly goes up from 5K to 20K by 16dB !).

With 18 bits I'm satisfied at -30dB, but maybe with 20 bits I'm more satisfied ...
Suppose that 18 bits would be the "standard" for legitimally play back at -30dB, then from that follows (it's just math) that with 16 bits all is as legit at -18dB.
When you are not able to play at -18dB (because it's too loud) the relatively simple solution is to adjust the gain of your amp. NOT by means of attenuating the existing gain !

The latter comes down to using the digital attenuation in the first place. So when you just don't use that, there's really no problem with 16 bits files that a more than 16 bit DAC can solve. Not that I know of.
Doubling or Upsampling to 88.2 KHz was created for better SQ (mainly Doubling !) but if it works out like that ... YMMV. For me it does not, because it needs a larger buffer in the soundcard (from 48 to 96 samples), and -again- NET it is not better because of that.

Please keep in mind, all of the above is related to 44.1/16 files only.
Once it would be a standard to have 96/24 etc. files, the "problems" are obvious, because you just won't be able to play them.

I'm not sure if you'd need any advise on bringing back your DAC to April Music, but if you spent that amount of money with the idea it could do 96/24 ... well ... The proof that you could be mislead is sufficiently available in this topic. yes. Along with that, the person involved not really to blame for it as well.
For the future : when the specs of a DAC do not specifically mention input samplerates and bits, better assume 48/16 as max. For that matter the CD10 does mention 96/24 for input explicitly. Whether *this* then is true (over USB !) I don't know.

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

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #51 on: March 02, 2008, 01:16:53 pm »

Dear Peter,

See DAC test below. Not much to brag about here...
My Eximus DAC is manufactured by APRIL, same company as maker of Stello DAC. Probably they all perform equally regarding digital input capabilities.

What does this means SQ-wise when using XXHighEnd? -Are we missing any sound quality [WHEN PLAYING REGULAR WAV FILES RIPPED FROM CDs - which is probably 99% what all of us are doing till now] because of this? Please advice. Several XX users have Eximus/Stello DAC's, so I believe this question is important.


Hi Pedal,
I think whether we are missing any SQ when playing back 16/44.1 wav files is up in the air.  If I remember correctly, Peter doesn't care for upsampling/doubling of 16/44.1 files ... digital volume,,,???, I think he feels you don't miss anything if not used aggressively, (mines set at 0 though; I use my analog vol). ***edit - see Peter's words above***
I'm guessing the usb on our units can only take in 16/44.1 & 16/48.  Other people talk about how they like the upsampling/doubling, there units can take that signal in, ours can't...idk.
But,
Coax input should accept more rates for April products?  I think that's what I understand?
So...
What about using something like Empirical Audio's Offramp I2S (doesn't the eximus have I2S input?; Empirical adds this input to benchmark,,, I wonder if he could do it for the Stello....), or the Offramp Turbo 2 (for spdif out) http://www.empiricalaudio.com/ These units take in usb and output I2S or spdif respectively.
This from his site too, "All will pass 16/44.1kHz, 24/96kHz, MP3 and iTunes unmodified files"... I wonder if it will pass more than that now?(site could be old?)
or
UD-10 USB Audio Converter http://www.trendsaudio.com/EN/Product/USB_Audio_desc.htm
Perhaps if we use the coax inputs, our dacs will do more.  Maybe just need to find a usb to coax (I2S?) converter that passes hirez signals...untouched hehe
I'm not so interested in upsampling/doubling as I am playing back hirez files when they become available.

This is what is on the product page of my Stello usb dac

"As a professional D/A converter, the DA100 can handle any input data up to 96 kHz/ 24 Bit as well as the normal CD standard 44.1 kHz/ 16 Bit." ... "True 24Bit Delta-Sigma DAC"  Would have been nice if they had mentioned that the usb input was incapable of that....
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« Reply #52 on: March 02, 2008, 08:39:01 pm »

Can I understand something:

If you have a 24bit/96 wav file, but your dac doesn't support 24... that you can set XXHE to 32 bit dac and it plays the file... yea?

also

I went to the Linn site and they have a lot of flac files at 24/88 would a dac that does 24/96 play these? no, right?  I mean if it did the files are being changed and not played back at the native rate...

Here's my goal... I wouldn't mind trying upsampling doubling twisting bits & pieces with XXHE ... but first and formost I want it and the dac to play back files perfectly in their native format... what ever XXHE (transport) sends to the dac I want the dac to play that without manipulation...just trying to find an audiophillic well made dac that  accepts ... well..how about these for starters (or the most important ones):

Start Audio Device Analysis Exclusive Mode support
040FFD38

   Supported : 2|16|44100|176400|4
Not supported : 2|24|44100|264600|6
Not supported : 2|32|44100|352800|8
   Supported : 2|16|48000|192000|4
Not supported : 2|24|48000|288000|6
Not supported : 2|32|48000|384000|8
Not supported : 2|16|88200|352800|4
Not supported : 2|24|88200|529200|6
Not supported : 2|32|88200|705600|8
Not supported : 2|16|96000|384000|4
Not supported : 2|24|96000|576000|6
Not supported : 2|32|96000|768000|8
Not supported : 2|16|176400|705600|4
Not supported : 2|24|176400|1058400|6
Not supported : 2|32|176400|1411200|8
Not supported : 2|16|192000|768000|4
Not supported : 2|24|192000|1152000|6
Not supported : 2|32|192000|1536000|8

Seems to me that within the immediate futures we're at least going to have some decent choices of hirez files...


Should I just get a Fireface (they should come out with an I2S output ???) I have one firewire into my lapto for the HDDs.  If I get a card with another firewire input ... would these two things get along and not collide?
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« Reply #53 on: March 02, 2008, 11:48:52 pm »

Hi Dave,

Maybe I don't understand all of your post, but

Quote
If you have a 24bit/96 wav file, but your dac doesn't support 24... that you can set XXHE to 32 bit dac and it plays the file... yea?

No. You can just leave it at 44.1/16 and it plays the file (and you won't hear the difference). Mind you, the DAC needs 96 KHz for this. In your case you won't be able to play the file at all (why ? because XX doesn't downsample from 96 KHz to 44.1 or 48 Khz -> it could, but it doesn't seem the right thing to do).
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For the Stealth III LPS PC :
W10-14393.0 - July 17, 2021 (2.11)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

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

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« Reply #54 on: March 02, 2008, 11:55:21 pm »

Hi pedal,

(btw, is there any importance in the picture underneath your text ? I see a read cross only ...)
No, just an error.

Quote
Let me first say that if you do not want to use the Digital Volume from XX for your own reasons, for 44.1/16 playback it doesn't matter at all.
Question: IF I had a DAC who accepts more than 16bit, then, would the use of digital volume in XX introduce a SQ improvement? Or to ask in another way, is there any “secret” trick hidden in the XXs volume feature which betters the SQ?

-Or is the SQ improvement due to the omission of the preamp (shorter signal path etc)?

Please specify.

-------------------

Apart from this, I must state that the Eximus is build to a very high standard, and it is the best sounding DAC I have heard. The SQ of a DAC is based on several factors, and although Eximus doesn’t accept hi-rez through it’s USB input, it excels in other areas. For my ears, it sets a new standard for neutrality and resolution. It will remain in my system for a long time to come. And who knows, maybe there will be a future upgrade choice for it’s digital input?

BTW: Eximus offers 4x upsampling internally, before the D/A converting takes place. It’s a worthwhile SQ improvement on all quality recordings.

----------------

After a short initial listening test today, I feel that vU2 is another improvement compared to T-5. U2 is simply more transparent. I hear more recorded depth and more acoustic space surrounding each instrument. The bass is tighter. I can play several dB louder (always a good sign) without listening stress.

This is when playing through my external preamp as usual (XX volume = 0). My system needs between 20 and 30dB attenuation, depending on song material. (Some CD’s are recorded louder than others). So, no way to relay solely on the XX volume due to Eximus’ 16bit restriction. But I could eventually do -12dB digitally with XX and the rest -8 to -18dB on my active preamp. Are there any theoretical reasons (SQ) for me to try this out?
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Hardware: Stealth Mach III > Lush^2 > 24/768 Phasure NOS1a/G3  > active preamp > 3-way active XO > amps > ribbon/dynamic true line source speakers.

Settings all settings as recommended by Peter by October 2019.
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« Reply #55 on: March 03, 2008, 01:19:03 am »

Quote
Let me first say that if you do not want to use the Digital Volume from XX for your own reasons, for 44.1/16 playback it doesn't matter at all.
Question: IF I had a DAC who accepts more than 16bit, then, would the use of digital volume in XX introduce a SQ improvement? Or to ask in another way, is there any “secret” trick hidden in the XXs volume feature which betters the SQ?

-Or is the SQ improvement due to the omission of the preamp (shorter signal path etc)?

Please specify.

The latter for sure, but without that still.
Explanation is (too) difficult though, and I'd rather awaiten my own approval which -this time- takes more than a few weeks of listening.
It's merely a reasoning why things are wrong otherwise (using analogue volume) than any tricks being applied, although the means the digital volume is applied is all the way legit in the first place (you could call that a trick as such).

Quote
This is when playing through my external preamp as usual (XX volume = 0). My system needs between 20 and 30dB attenuation, depending on song material. (Some CD’s are recorded louder than others). So, no way to relay solely on the XX volume due to Eximus’ 16bit restriction. But I could eventually do -12dB digitally with XX and the rest -8 to -18dB on my active preamp. Are there any theoretical reasons (SQ) for me to try this out?

Yes, these theoretical reasons definitely exist, but I'd rather that you (or anyone) first testify yourself that the results are as I intend (i.e. "WOW"), than creating placebo's all over that won't help anyone including me (or the other way around Happy).
Please keep in mind, this is a "way off track" theory, that made me start doing it anyway, and since it worked out for me (placebo !!) I so far think I'm right.
As you know, audio is  difficult, and the last thing I'd want is everybody just wanting to hear the same as e.g. I do. There's too much copying on the Internet already, and I want not only myself to stay far from that, but "you" just the same. No way to learn otherwise !
So far, all bits and pieces brought by some of you, are in 100% line of my own perceivement, and they are all but one on the positive edge. The exception, so far, is the lower bass end (as was brought by SeVeReD), and exactly that I recognize too.

I know, a great deal of these things you want to know because you really can't try it currently, but at this moment I really don't want to be all mighty while things are so much different that I keep on being in doubt myself what's wrong and what's right. You could say that *I* need the confidence from you all to be 100% sure. Only at that stage I will, say, "order" you.
I hope you can understand this (as well as the good cause).

Peter
« Last Edit: March 06, 2008, 12:30:03 am by PeterSt » Logged

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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