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Author Topic: World's first NOS 24/384 filterless DAC  (Read 338946 times)
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PeterSt
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« Reply #330 on: July 29, 2010, 09:43:04 am »

Quote
will it work with 8 channels? I can have all balanced outputs for what i care of that ^^
The most important thing is that all steps have the same impedance output seen by the amp (which IS different than without it).

It will work with 8 channels allright, if you only multiply the costs with 4. swoon
A kind of crazy, right ?

The impedance isn't influenced a sinlge bit though ...
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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 10/20 cores) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 30/-/1/1/1/ Q1Factor = 10 / Dev.Buffer = 4096 / ClockRes = 15ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 140.19  (max 140.19) / 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 = 35 / Nervous Rate = 10 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / *Arc Prediction Filtering (16x)* / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^2*A:B-W-Y-R, B:B-W-R* 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 A:B-R, B:B-R* Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
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*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
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« Reply #331 on: July 29, 2010, 10:32:19 am »

It will work with 8 channels allright, if you only multiply the costs with 4. swoon
A kind of crazy, right ?

Ahiahiahi, now i have to think about it a little bit. It'll depend on the base cost of the 8ch version which is not known yet, right?

But tell me one thing, if there would be 4 pots, can you factory set to have a different default (i.e 0, -4db, 0, -8db) and then change all at the same time with the remote? This would cover the different gain of the amps in a totally transparent way (probably better than doing it in the digital domain like i do now).
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« Reply #332 on: July 29, 2010, 02:44:52 pm »

So when will the 2 channel version of the NOS1 start shipping and what will the cost be?  Thanks.
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PeterSt
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« Reply #333 on: July 29, 2010, 04:07:28 pm »

But tell me one thing, if there would be 4 pots, can you factory set to have a different default (i.e 0, -4db, 0, -8db) and then change all at the same time with the remote? This would cover the different gain of the amps in a totally transparent way (probably better than doing it in the digital domain like i do now).

Well, actually it can, and I thought about telling you this, but then I couldn't see a good solution in it. But, I only thought of the different levels for the XOver channels, and didn't see a good reason to change the individual level of them *after* the XOver filters have been defined. And so I never mentioned it. Your reason though is a most legit one. But factory set ? hmm ... how to do that ? of course, if you'd tell me what the difference in gain is, it could. But it would be the most dangerous, and fully dependend on your amps (and changes to them ... others later maybe ...). So, what I would propose in that case is to have an additional small trimpot at the output of one such a "stage" to control the individual attenuation, which would be as legit as the whole thing not being in the signal path, *if* it works in the first place.

But for the 8 channel analogue stages, which as you know are not ready yet, it would be better to make that single ended from the ground up, which avoids the differential attenuation setup, so the 1600 drops to 800 again. This still allows for others to choose for balanced, if they only know it has its price.
And to remember : having both SE and Balanced in one cabinet won't go anyway, because there's no space for both connectors. This, unless the relatively expensive jack types are applied, which take both SE and Balanced in one terminal.

Hmm ...

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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 10/20 cores) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 30/-/1/1/1/ Q1Factor = 10 / Dev.Buffer = 4096 / ClockRes = 15ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 140.19  (max 140.19) / 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 = 35 / Nervous Rate = 10 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / *Arc Prediction Filtering (16x)* / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^2*A:B-W-Y-R, B:B-W-R* 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 A:B-R, B:B-R* Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
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 #334 on: July 29, 2010, 08:06:46 pm »

Well, actually it can, and I thought about telling you this, but then I couldn't see a good solution in it. But, I only thought of the different levels for the XOver channels, and didn't see a good reason to change the individual level of them *after* the XOver filters have been defined. And so I never mentioned it. Your reason though is a most legit one. But factory set ? hmm ... how to do that ? of course, if you'd tell me what the difference in gain is, it could. But it would be the most dangerous, and fully dependend on your amps (and changes to them ... others later maybe ...). So, what I would propose in that case is to have an additional small trimpot at the output of one such a "stage" to control the individual attenuation, which would be as legit as the whole thing not being in the signal path, *if* it works in the first place.

Changing the gain of a power amp is more difficult and often cause much worse sound. Doing it in the source would be just perfect.
I would rather consider a jumper (internal) than another pot, since i wouldnt change in years. Bu yes of course it wont be in the signal path since itìs set as default by another mean. For factory set I meant set by you according to my needs (that i dont know yet haha).

Quote
But for the 8 channel analogue stages, which as you know are not ready yet, it would be better to make that single ended from the ground up, which avoids the differential attenuation setup, so the 1600 drops to 800 again. This still allows for others to choose for balanced, if they only know it has its price.

For me it doesnt matter if balanced or single ended, as long as SE sounds as good in the DAC part as balanced.
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Hardware:
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« Reply #335 on: July 30, 2010, 03:24:07 am »

Peter

Sorry if this has been covered before. I did a quick search but couldn't find the answer.

Your XXHE player/8 Channel DAC will have adjustments in each channel for: a) level b) phase c) variable crossover points with independently variable slopes d) delay?

I seem to remember you saying there would not be any facility for minimal EQ adjustment (usually to slightly pull down any response peaks in the bass) - correct?

Thanks
Frank
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PeterSt
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« Reply #336 on: July 30, 2010, 08:13:08 am »

Hi Frank,

a) No (I think Happy, b) Yes, c) Yes, d) No.

Ad a
See before posts addressed to Telstar. I judt didn't have in my mind to control that. But maybe it can't be avoided ... I mean, if I want my tweeter to nicely fit the mid, first the tweeter has to go down. Doing that by means of an analogue pot again would be a kind of wrongish, don't you (we) think ? But the point is, this makes all subject to "wrong" digital attennuation, and I could just as well make the current volume steps 0.1dB ...
Of course, it may turn out unavoidable that the whole band is influenced because of the crossover slope, and when that is so, better have the (most) fine digital attenuation. Once that is there, Telstar doesn't need his adjustments per channel anymore.

Ad d
Although in passive analogue some things can be done about this, it is not the way to do it ... I think. Better solve this in the physical driver alignment. But if you're talking about a subwoofer ... it should be in the subwoofer (overall phase).


About the EQ ... that is correct. But in the speaker filter (design) area it can't be avoided. But these should be parametric.


Hey, I guess you are somewhat ahead of me. yes
All very valid points Frank.
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W10-14393.0 - Oct 31, 2018 (2.10)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 10/20 cores) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 30/-/1/1/1/ Q1Factor = 10 / Dev.Buffer = 4096 / ClockRes = 15ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 140.19  (max 140.19) / 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 = 35 / Nervous Rate = 10 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / *Arc Prediction Filtering (16x)* / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^2*A:B-W-Y-R, B:B-W-R* 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 A:B-R, B:B-R* Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
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 #337 on: July 30, 2010, 10:54:05 am »

Ad d
Although in passive analogue some things can be done about this, it is not the way to do it ... I think. Better solve this in the physical driver alignment. But if you're talking about a subwoofer ... it should be in the subwoofer (overall phase).

Well, there are cases in which physical alignment can cause aesthetic issues. I dont need it i think, but it would be easy (i think) to implement and very useful for many, probably lossless (you tell me).
About subwoofers, think about who DIY them and dont use very adjustable amps... i dont see many 2kw subwoofer amps and if there are, too expensive to buy rather than build (ok this is only my case which is rather an exception).

Quote
About the EQ ... that is correct. But in the speaker filter (design) area it can't be avoided. But these should be parametric.

It's early to talk about it, but some drivers need some correction/EQ, yes parametric and with adjustable slope (Q). Not everyone can afford very expensive drivers with quite linear response, which are the MINORITY, anyway.
Mechanical equalization like is done with the driver construction or the speaker construction (horn size/shape etc), baffle size/shape etc. is THE best, second is digital correction IMO.

There was a very interesting opinion on a white paper from Harman international about speakers design and measurements and they differentiate the kind of adjustments needed because of the drivers/cabinet and because of the room. I know that you disagree on the second point, but the first... I do a notch filter in the digital domain, it requires higher attenuation than if i used a cap and a res... but i just didnt want to use them and the result has been defined "flat" (albeit is not in the HF, but nobody but me seems to hear them  grazy).
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Hardware:
OrigenAE H5 case | E5300 fanless |  8GB RAM | Winmate DC-DC fanless PSU | OS on SSD | Renesas USB3 pcie card | Belden highspeed usb cable | Audio-gd dac19 NOS with sigxer F1 | My_ref_FE mono amps | Albedo Apex speakers
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« Reply #338 on: July 30, 2010, 12:29:13 pm »

Peter

You are busy just getting the two channel DAC out, and that should be your priority.

Can I suggest that you continue to reconsider/rethink your configuration of the 8 channel DAC before finalising.

As I see it the most prevalent use of this 8-channel DAC will be by audiophiles wanting to run an active system. All of the features I asked about are what I would consider "basic" reqirements for such a system (Certainly in the Pro world, where it has been done successfully for ages - but not of course with the same resolution and fidelity as you are achieving).

Basic software requirements, usually available with an 8-Channel DAC would include:
1) Level adjustment per channel  
2) Phase adjustment  
3) Variable crossover points with independently variable slopes (i.e different slope high pass & low pass filters available at any crossover point)  
4) Simple delay for each channel [easy to achieve digitally and essential for good crossover design - physical alignment of drivers is not always possible or a design goal]    
5) Some EQ options.

Would building any of these software options into the DAC rather than XXHE make it easier?

Peter, please take some extra time to investigate before pushing yourself to get your 8-channel DAC to market.

Cheers
Frank
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« Reply #339 on: July 30, 2010, 05:18:51 pm »

No answer the first time....so may I ask again when will the 2 channel version of the NOS1 start shipping and what will the cost be?  Thanks.
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« Reply #340 on: July 30, 2010, 08:08:11 pm »

Yeah, I know. But I was waiting until the day was over, and I'd had some "results" on a stupid power inlet not doing what it promises (switching voltage between 115 and 230).

So, last Monday the cabinet should have gone into production (meaning the cabinet itself), but at assembling everything in the proto, this power inlet laughed at me within 2 minutes.
After fighting with and over this the whole week, a few hours back I finally received the message from the manufacturer that indeed it can't work.
So now I need another 100 of which I don't know where to get them at this moment, then they must be shipped from whereever it will be, and then at last I can try whether that works (but hey, why wouldn't it do what is promised eh ?), and finally I can let produce that cabinet.

Eerflappin, you are experienced in asking when the shipping date will be. I hope you are better than me in estimating it.
Btw, all other parts are here now, except for that volume trial where all apparently depends on one person which is only there next week.
Final price is 2900 excluding volume control *if* that works, which I can't try at the moment.

One thing I know, I am not ever going to design a DAC again, unless it's the same. It is all one big pile of hopelessly being dependend on suppliers which never do what they promise. I've been waiting for nearly 5 months for stupid capacitors. Come on now.
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W10-14393.0 - Oct 31, 2018 (2.10)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 10/20 cores) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 30/-/1/1/1/ Q1Factor = 10 / Dev.Buffer = 4096 / ClockRes = 15ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 140.19  (max 140.19) / 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 = 35 / Nervous Rate = 10 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / *Arc Prediction Filtering (16x)* / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^2*A:B-W-Y-R, B:B-W-R* 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 A:B-R, B:B-R* Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
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 #341 on: July 30, 2010, 11:58:21 pm »

Can I suggest that you continue to reconsider/rethink your configuration of the 8 channel DAC before finalising.
(...)

I have to say that I overall agree with you.
Some things are more important,some less but all can be useful and it's not difficult to do at software/drivers level.
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Software:
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Hardware:
OrigenAE H5 case | E5300 fanless |  8GB RAM | Winmate DC-DC fanless PSU | OS on SSD | Renesas USB3 pcie card | Belden highspeed usb cable | Audio-gd dac19 NOS with sigxer F1 | My_ref_FE mono amps | Albedo Apex speakers
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« Reply #342 on: August 14, 2010, 12:33:51 pm »

All,

First a few general, boring things.

It has become clear that this "channel integration" thing really is THE thing to hunt for. When I wrote about this the first time I was only used to it two days I think, so that was extasy all over. But actually -at running more and more albums- it is an amazing thing you just haven't heard *anywhere*. Yes, I'm pretty sure about that.
This too excels with certain types of music, and since this is all about separation - think about King's Singers. I listened too Good Vibrations of them the other day, and really thought it was the most beautiful I EVER heard.

Further (but this is really the last one) improvement has been made on the sound in general, which partly came from the lot being in its cabinet now (I perceived somewhat more body from that). I must add to this that this time I used the analyser to squeeze down every bit of noise which is in there, and what comes from the DAC can not be measured anymore (at the end of the intterlink) from an FFT. IOW, this is beyond -144dB, as far as the analyser goes, which thus shows -144dB. And I must say, I never heard completely *no noise at all* from my horn speakers (which as you may know by now are 115dB sensitive, and with the amps at full gain of 27dB).
The larger deal tough came from the DAC and gain stage now being "integrated" by means of two "click connect" PCBs, while the proto of the gain stage was connected with soldered wires and two connectors. And yes, that all matters. This time it brought the snare drum on Six Blade Knife (Dire Straits) to more reality, while it was sounding like firm cardboard before (what one can hunt for swoon).

And this brings me to the volume control ...

Let me (again nag) first that this just took over 2 weeks of wasted time. Wasted time to the sense of that it has to go (hence fit) in the cabinet (of which I'm producing a 100 at first) or not. I mean, I can't produce the definite cabinet when I don't know what will be in there.
So as I told the other time, I immediately ordered one of which I thought it would be good (I recall that was on a Thursday) and heard nothing. The next Sunday though I received an email it would be shipped the next Tuesday. And so I received it last Wednesday. A week later than could have been.
The last two days I have been working on it, to finally come to the conclusion that it can't work not being in the signal path. This is about adjusting a resistance somewhere which should be on a tracelength of 3mm, while here I can't do it otherwise than over a length of 20cm. So, distortion all over, and it didn't even attenuate.

At wrapping all up yesterday, I thought to at least check what would happen if I would use it as intended. Thus, in the signal path as any normal attenuator;
I have been measuring the whole afternoon, which merely was the result from not seeing bad things at all. Oh, I saw a decrease of almost a factor two on harmonic distortion, but you must know that isn't all *that* bad, while we're at -105dB, and this factor two implies 6 dB (thus now -99).
The only thing I could find as strange was the slowly increasing noise level beyond the audio band, but since this is inaudible for both the frequency (25KHz and up) and the level (-130dB), I actually could not "see" where it would be wrong. Not in hours of time.

And so, at listening time the volume control was in the chain, and I ran a couple of tracks that I'm familiar with.
Now, with the explicit notice that all is relative, and that I'm used to what I'm used to (which is, say, a far too high standard), I could hear immediately that it didn't "work". The width of the soundstage seems compressed (yea, must find some new phenomena sometimes haha), bass doesn't go deep at all, but worse, is coloured, I sensed some standing wave behaviour in the low areas and for sure perceived standing (buzzing) waves in the higher regions. It was mushy. It didn't fit anymore. Piano notes shouted. Te music seemed too loud to be comfortable.

I imagined "you" listening to it, and I imagined that you still could say "wow", not knowing better. But, it would be critical. It would be critical to "the best DAC ever" and I would not dare to bet on it.

At this moment I can't decide;
*If* it would be in there, there would always be a direct path to the (different set of) outputs as well. But heck, why to do it in the first place ?

1. To comfortably attenuate the sound ?
Maybe, but in the very end non-sense. When talking about other players it is not much of a problem to begin with, and when talking about XXHighEnd a lot can be done to make it more comfortable. It can be made more fast (responding), it can be made more direct (like choosing the number of steps to in/decrease). But no, there won't be a physical knob. So what.

2. To attenuate with smaller steps
Yes, but this is very relative. This one works with 64 1dB steps, and using the knob isn't even comfortable, knowing that you need 10 full cycles or so of the knob to go from 64 to 0. I didn't try it with the remote, but imagine it takes ages to find "your" volume, because the steps can't be addressed directly, and it takes a "wait time" when the up/down key is continuesly pressed to start moving, and after that it will go too fast to stop where you want, or will go too slow to be comfortable.
Allright, where this is 1dB, XX works with 1.5dB. Well, I tell you that the annoying thing at the moment is those small steps when you want larger. It really really never occurred to me that I couldn't find "my" volume in those 1.5dB steps. So ... if you really think you need smaller steps (like Telstar) better think twice if this may be nagging, while in reality you are after the best sound quality. So, if I *tell* you this is nagging, maybe it helps ! haha

3. To protect from static
Is this only stupid theory ? I mean, when we play loud with our well matched amps, we play near full volume, and the static will be as bad as without the volume.
Of course this is different when we use 700W amps on speakers who need 50 only, so we only talk about that (and these people sure will exist, so why not take it as a legit reason).
For me it is clear that we just need an inherent solution for this. This is not a panic-mute button (you won't be able to press it from shock) but something which detects it and just cuts the signal. In theory this can be done in-DAC, but I think it will be too complicated to have a program running in there which can "see" this, like the Crack Detection in XXHighEnd can. The Crack Detection in XXHighEnd by itself works allright, weren't it that it is hooked in the chain too soon. I mean, after the Crack Detection process there's more which can mangle the data, so theoretically it can be done later. However, while it can be there at the latest at filling the buffers towards the OS and driver, there's always the driver at latest which can do things wrong. Not that we need to expect it will do things wrong (I never encountered that except for one report of a user with an explanation which goes in that direction), but we can imply it ourselves so much. Thus, when we squeeze down the latency so much that we aren't able to pass on the bytes in the proper sequence - meaning just skip one - we'd have it. On this matter, notice that an audio sample comprises of more bytes, and if one is left out you'd have static. This too I never encountered, which may imply that the OS (or driver ?) is very much responsible for keeping these things together, so if we really could rely on *that*, then I could guarantee it in the software.
But then this is only *my* software, while the DAC won't prevent you from using Foobar etc. ...

This brings me to the last subject for today : the inputs.
As it looks now, it won't be possible to use the available digital input (SPDIF) for measuring means. This never has been a subject much, but for me it is a kind of important. This is in the area of people "testing" the DAC (could be a review), while without such an input there is not much to test. The input is physically there allright, but it implies such a bad "stream" that this itself masks all the good stuff from the DAC. Most probably this is because the SPDIF input will route via the OS, or better : I can't get the SPDIF input passing through to i2s (which I use), unless I switch on the "Listen to this" in the device's properties. So, while the both are available on the same board (think like a sound card), the OS apparently is used for the routing, and nothing than harmonic distortion comes from that (like over 30%).
I didn't get around yet to testing the analogue inputs. But I need to add something to this :

I guess I made a thinking error on combining two "connection" situations where either at this moment won't allow analogue inputs to work (but the combination does while it can't exist as combination). The one situation needs a space in the cabinet which isn't there because the PCB with the clocks occupies it (and which I won't move farther away because it should be as close to the DAC as possible), while the other situation will be in the PC, but it will have a too jittery clock connection because the clocks are in the DAC cabinet. And please keep in mind : this "inputs" thing was about "your measuring" (so not really recording as such) which by itself needs the same clock for playback and the recording (of that playback).
Hmm ... something to sort out. But maybe I'll leave it be; It ever was a nice "withgoing" feauture, but if it's not "withgoing", then bye.
Anyone there who needs 400 Neutrik connectors ? you can also use them for outputs ! hahaha Cry
Ok, I'll see what I can make of it.

First the Crack Detection I'd say.
Peter




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For the Stealth III LPS PC :
W10-14393.0 - Oct 31, 2018 (2.10)
XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 10/20 cores) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 30/-/1/1/1/ Q1Factor = 10 / Dev.Buffer = 4096 / ClockRes = 15ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 140.19  (max 140.19) / 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 = 35 / Nervous Rate = 10 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / *Arc Prediction Filtering (16x)* / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^2*A:B-W-Y-R, B:B-W-R* 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 A:B-R, B:B-R* Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
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 #343 on: August 14, 2010, 01:37:59 pm »


And this brings me to the volume control ...


Hi Peter,

in my opinion you should deliver the first 100pcs DAC as planned without volume control. GO for the best SQ as intended.
This was your intention until the "volume control" came up and you should not waggle.
As I see from your story, there is a lot of research involved,  and we would like to have the "World's first NOS 24/192 filterless DAC" 

NOW

without volume control and the best SQ available.

Enthusiastics may have a look to the" Weiss ATT202 Passive Attenuator".

Joachim

------------------------------------
Windows VISTA SP2> XXHE 09z-2 - Adaptive mode -  Q1/2/3/4/5 = 1/0/0/0/0 / No Invert / Playerprio = Low / ThreadPrio = RealTime / Scheme = 3 @ UnAttended / Arc Prediction * > Acer , Intel i750>|- Weiss Minerva -|- Dartzeel NHB-108B > speakers: Anat Referenz II Studio>


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« Reply #344 on: September 08, 2010, 10:52:19 pm »

Wednesdy 25th of August

After months of attempts we (Gerard & Leon) get in the car on our way to Peter.
Our aim ……….listening for the very first time to the  Phasure NOS1 DAC and with
a bit of luck admire the machine. Gerard promised Peter to arrange the lunch and
so we went on our way packed with all kind of bred, sandwiches and anything
you can think of to put on top of the bred.

Thanks to the Tomtom we easily managed to find Peter’s place and at about half
past twelve we arrived over there. Worthwhile mentioning is that Peter had this
house being designed according his own architecture and when you first see the
place you cannot help but admiring it, what a beautiful place to live.
In order to get to know each other a bit we talked to Peter, Ciska (his wive) and
their son, of whom we must admit have completely forgotten his name (sorry
Peter), prior to the lunch. There was plenty for everybody, so soon we had an
animated conversation between the family and ourselves. One of the items we
discussed during lunch was Peter’s fruitless attempt to incorporate a volume
control into the DAC.  Let’s not try to explain all the different ways he tried to
achieve his goal, just important however was that even the day prior to our
meeting he made another attempt, resulting in first a completely disfunctioning
PC system and after resolving that matter a system with audible less SQ than
before. Talking about this we made the remark that working with digital music
you handle bits and bytes which is an exact science, so in other words you should
be able to solve that by measuring stuff. Peter’s answer was a kind of a snoring
laugh and the remark that we should not forget to put that in writing ea the fact
that it is an exact science. Well ……… what do we know!!! However, we all know
how important the SQ thing is for Peter, so the volume control was disabled for
our listening session.

After lunch it was time to listen to the output of the Phasure NOS1 DAC.  whistle
 First we had a look on the oddly but very functional shaped unit. Remember, the design
never should compromise SQ therefore the design of the unit is somewhat
different from what you would expect from a standard DAC. The PC and the
whole sound system is set up in the living room of the family. And what a place to
listen to music! It’s big, without too many obstacles and various places where you
could stand or sit down to enjoy listening to the music. Peter pointed out the best
place to have the music coming to you in the most balanced way, which
coincidentally turned out to be the bar.  drinks Being very docile people we immediately
followed his advice Wink
We were being asked what kind of music we would like to listen to and
immediately explained that no matter what, he would start the session with some
music he himself had chosen for us. These turned out to be recordings from “The
Beatles” Not particularly the kind of music you would expect, but he did it for a
good reason. So we started the session with listening first to Get Back and
secondly to Yellow Submarine. What happened was that you actually could
believe you were there in the studio while the music was recorded. The details
and the way the sound came to us made a rather poor recording quality quite
acceptable and nice to listen to. As example you could actually hear the separate
voices coming to you from different places during the simultaneous singing of
John Lennon and Paul McCartney. The biggest surprise for us was with Yellow
Submarine, closing your eyes you would totally accept having the Beatles in front
of you while playing with their feet in the rolling waves of the Northsea,
unbelievable.
As said befor, Peter had a meaning with letting us listen to the Beatles. He
wanted to prove that with the right equipement it is possible to have a very
acceptable SQ with even very old and/or poor recording quality.  Well, he
definitely made his point, despite the fact that the Beatles are neither the taste
of one of us, it was definitely not a burden listening to it. On the contrary it made
very clear the value of the DAC.
After discussing the first part we were asked to point out some music we would
like to be played. We came up with “Dead Can Dance” the song Sanvean,
Bubbling Under of the album Motion Picture of Yello some numbers of John Lee
Hooker and a Case of You from Diana Krall. Now something strange happened
and for me (Leon) totally unexpected. The song “Sanvean” didn’t moved me, at
all. The sound was all right and basically I couldn’t put my finger on it, but the
voice of Lisa Gerard didn’t spring out the way she normally does. This song is one
of the most beautiful songs I know from Lisa Gerard and the number never seizes
to touch me, even when I play it at my own place with a definitely lesser system
compared to Peter’s. But not this time. Even Peter, not knowing this song at all
was not happy with the way it sounded. In fact he asked me if this was the way
her voice should sound and if knew the violins were actually synthesizers. Now
personally I don’t mind much whether they are violins or synthesizers, it’s the
voice doing the trick for me and the emotion she gives me with this song and the
emotion just wasn’t there.
In order to check things out, Peter chose a number of Emma Shapplin, I think it
was “Un Sospir’di Voi” of the album Etterna, again a woman’s voice as biggest
part of the music, This song we know all three and we agreed that also Emma
Shapplin didn’t sound the way she should. Also her voice didn’t spring to live the
way we know. Strange and it was obvious for us that did was somewhat
unexpected and annoying for Peter.
After these women, it was time for the other selections and this time the DAC
never failed. Regardless whether we listened to the blues of John Lee Hooker, the
by piano supported jazz of Diana Krall or the electronic of Yello it all came to us
in rich details and very very well balanced. With wonderful placing of the
instruments in the space and an extreme purity from the lowest low to the
highest high.
Between our selections Peter played also his choices, so we also had the pleasure
listening to tracks from Jan Akkerman, Ben Harper, One Step Closer van Shulman
en Taj Mahal.
One thing we both concluded afterwards is how different audiophile people listen
to music playback compared to ordinary mortals such as us. Someone like Peter,
who is used to listen mercilessly critical to quality of recordings and quality of
playback, notice details we are totally unaware of unless clearly being pointed
out at.  This makes judging the new DAC not an easy job. Peter expects the
highest level criticism where first we have to deal with a set of equipment which
is not comparable with the systems we have at our respective homes, and
secondly you talk with a guy who works professionally with music recordings and
has a very clear goal to achieve with his DAC. That goals is make the best DAC
ever for any kind of music, regardless if it is classical music, hip hop, trance or
just ordinary rock. After several tracks we were asked quite direct to express
what we thought, felt or noticed while listening to the track. Now we don’t know
whether is was the Schumann Synchronizer  or just the way we normally would
respond, but we tried to adjust our way of listening to all the tracks to the way
Peter is obviously judging his own system. One thing is for sure however, despite
the Schumann Synchronizer we never heard it rain throughout the whole
afternoon  rofl rofl Wink.
There is one more thing to tell about our experiences this afternoon. Peter told us
that he recently had learned a lot about what goes wrong at recording studios
and how that of course affects the SQ of playback. For that reason he recorded,
at his own home, his son playing drums and cymbals. For me personally (Leon) I
congratulated myself with not giving my son a drum set when he grew up. But for
showing the capabilities of the NOS1 DAC it was very convincing. Playing the
recorded session and us standing outside on the porch it was as if his son was
actually playing the drum set inside the house, you just couldn’t tell the
difference. The cymbals just sounded the way cymbals do when you hear them
real live. Truly remarkable!
Okay, so now it is time to get back on these two women who didn’t want to do it
our way, Lisa and Emma. The day after our visit we got a mail from Peter
telling us what he did after our session. We told earlier in this report the fact that
Peter was not happy at all with the SQ of the tracks of Lisa Gerard and Emma
Shapplin. We also told about Peter’s attempt to get his DAC the best answer for
ALL different styles of music. So he mailed us that he decided to rule out every
possible influence of the volume control by totally disconnect all of it from the
DAC, including the power. As we understand that made a big difference. 
Peter explained it to us in Dutch, so down below you will find our translation of it
into English (Peter, sorry if the translation isn’t exact as you wrote it)

First put on Sanvean (which I didn’t know before). Well, it took me less than 10
seconds to make my judgment. First the “violins”, clearly audible now as having
way to much the same vibrato for being real violins, obviously a real synth. But
further? brrrr, I definitely should listen more to her. The whole room filled up now,
say just like with Yello. After that it wasn’t possible anymore to listen specifically
to the “violins” caused by that overwhelming voice. Very full and definitely a lot
bigger than yesterday. Well, this is how it is supposed to be I thought.
A little while later I realized that yesterday I squeezed down the volume a bit,
thinking “maybe they find it too loud” You know (Gerard) … Irritant. Well, this
time “I had her”  finally on the same level as Yello without any second feelings
about it and so I just thought of this.Than Emma ... I don’t know (Leon) if you know
that specific track, but what I totally didn’t  understand yesterday was why her voice
came up from the right speaker.
I never experienced that before, very odd. And now? Just from the midst of the two
speakers, the way it should be. The “thunder” from the beginning of the track, all
of a sudden was real thunder again and after listening for 20 seconds the shivers
ran down my spine. Flat? …… never. Just powerful as ever (the reason why I put
this track on opposite to Sanvean in the first place, but, what amazingly, didn’t
work at all)


We believe Peter on his word when he writes this. The quality of what we heard
yesterday makes it hard to believe that this system would not be capable of
dealing with powerful women voices. Nevertheless Peter we sure wouldn’t mind
convincing ourselves again by listening to Lisa and Emma a bit further.
We like to conclude this report with our wish to try the Phasure NOS1 DAC at our
own place. From what we heard throughout the day we are convinced this DAC is
potentially one of the best and will upgrade the way we enjoy listening to music
to a new level.   

We like to thank Peter and Ciska for there hospitality!  thankyou


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Januari 2017 XX 2.05
OriginAE x11v Home build HTCP ~ Asus x79 mobo 3930K 12 core underclock 500 MHz,
16GB, *Windows 10 Pro 64 bit build 14393.0* *from RAM*, music on music server / 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 = Not the best (OS from RAM issue) / Time Stability = Off (OS from RAM issue) / Custom Filter Mid 705600 / -> USB3 (Silverstone both sides isolated = Sw#3 of NOS1a = Up) -> *Clairixa USB 15cm* -> Intona Isolator -> Clairixa USB 0.70m -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink ->> Metrum Amps ->> Metrum Acoustics ESL Open Baffle.
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