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A novel way to massively improve the SQ of computer audio streaming


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Most important: please realize this thread is about bleeding edge experimentation and discovery. No one has The Answer™. If you are not into tweaking, just know that you can have a musically satisfying system without doing any of the nutty things we do here.

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4 hours ago, seeteeyou said:

 

It cracked me up whenever these claims were popping up while nobody really bothered to back things up with any kinda evidence whatsoever

 

 

They wouldn't even show any kinda numbers / data / measurements etc. but it's just "better" anyways. LOL

 

I don't mean to be disrespectful but sometimes we've gotta be realistic IMHO. Now I'm gonna be the bad guy since I'm spelling it out.

 

Most (>90 %) manufacturers doesn’t publish their clock measurements, probably because most people doesn’t know how to interpret them. Charles Hansen and his team is one of the most respected HIFI innovators. You don’t think they have actually tried if an external clock input to the QX-5 would degrade its performance or not?

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45 minutes ago, seeteeyou said:

Not really what I think or anyone thinks, it's just a matter of apples-to-apples comparisons. Talk is cheap and that's all.

 

BTW, let's compare SoTM sCLK-OCX10 to one of their best MV336

 

https://www.morion-us.com/morion-inroduces-xtal-clear-ocxo/


https://www.morion-us.com/quartz_oscillators/790/mv336/

http://www.morion.com.ru/eng/news/open/95/

http://www.morion.com.ru/eng/oscillators/ocxo/

http://www.morion.com.ru/catalog_pdf/31-MV336_REV_9.pdf

http://www.morion.com.ru/catalog_pdf/32-MV336M_REV_5.pdf

https://www.morion-us.com/catalog_pdf/mv336m.pdf

 

Just stealing (MAN, I'm such a petty thief) something from Rajiv that's posted on page 145

 

https://www.computeraudiophile.com/forums/topic/30376-a-novel-way-to-massively-improve-the-sq-of-computer-audio-streaming/?page=145&tab=comments#comment-722223

 Typical Phase noise(dBC/Hz)

                                                                                           My

                                       sCLK-OCX10                       Cybershaft    Mutec
     Frequency  Standard  Advanced  Reference       OP-14         Ref 10 (published)
         10Hz  :       -125         -135        -140                   -132.4              ≤ -142
       100Hz  :       -150         -160        -160                   -141.9              ≤ --155
         1KHz  :       -160        -163        -165                                             ≤ --160
       10KHz  :       -165        -165        -165
      100KHz  :      -165        -165        -165
      Noise floor:                                                                                        ≤ -166

 

https://www.morion-us.com/catalog_pdf/mv336.pdf#page=2

                                       Morion MV336
     Frequency         -              LN          ULN  
        0.1Hz  :       <-80          <-85        <-92
           1Hz  :       <-113      <-116        <-120
         10Hz  :       <-143      <-144        <-145
       100Hz  :       <-154      <-156        <-157
         1kHz  :       <-160      <-160        <-160
       10kHz  :       <-160      <-160        <-160
 

Maybe they're putting something even better inside QX-5 Twenty?

 

Is those measurements really made on the input of a DAC with a digital cables as cables would be needed for an external clock to work?

 

How many device was attached to the master/reference clock and did more device changed the measurements?

 

How does phase noise change with length of the cable at say 30, 50 and 100 cm?

 

Finally are those clocks good because they are external or can an identical good clock be used internal in a DAC or server with better result, because of much shorter cables?

 

* Maybe you know the answer to my above questions, but I believe most people doesn’t know how to interpret those numbers and are not aver under which conditions they there made.

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  • 2 weeks later...
13 hours ago, Beolab said:

 

The Puccini DAC + Master Word Clock with USB and dither, and it did cost untill 2016: 22.800 Euro , and today you buy it used for about 12.000-16.000 Euros in good condition depending on the age and Mark version. 

 

The Paganini Upsampler is about 10.000 Euros, but used you can find the whole stack for about 15.000 Euros. 

 

And dSC is the company that invented Asychronic USB standard with their own patent. And they are using VCOX (non oven based like MSB ) crystal ultra high precision clocks that cost around 6000 Euros for just the crystal Clock on the component market, in their DAC’s , Clocks , Streamers . 

So yes the USB input is connected to the VCOX clock. 

 

 

I think it was Gordon Rankin that developed asynchronous USB data streaming with his Streamlength software.

 

https://www.stereophile.com/content/wavelength-audio-proton-usb-da-converter

 

http://www.usbdacs.com/Concept/Concept.html

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30 minutes ago, ElviaCaprice said:

Too funny, how did I not know it would come down to system components unable to correctly bring out the best SQ of the boutique component.  That you need other boutique components to get the best (correct) SQ. 

 

If the component is unable to accommodate the non boutique audiophile components properly, then it's basically worthless to most of us audiophiles. 

 

Thus it can't be tested properly and it retains it's bragging rights to those who don't want it compared to cheaper components.   They can continue the name dropping.

 

Please continue the test, Rajiv, with your own setup, regardless of the objection.  If the boutique component fails to live up to it's hype via USB or ethernet, then we know where it stands in practicality.

 

 

 

To get the “best SQ” of a Chord blu mk2 and to use its 1M taps you need to have a DAC that has dual BNC in. Someone always need to start for progress to take place.

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15 minutes ago, ElviaCaprice said:

What does blu mk2 have to do with a trifecta comparison? 

 

Was it not obvious? You complaint about “system components unable to correctly bring out the best SQ of the boutique component”. I just point out that the same thing has been done by a company you seem to like with Chord blu mk2. Maybe not all comments has to be about trifecta comparisons?

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20 hours ago, JohnSwenson said:

The switches mentioned are only 1/100 switches so cannot handle gigabit Ethernet. If you need gigabit for other devices go ahead and run them through a separate gigabit switch, using the FS108 or FS105 on a connection to the audio device.

 

I did not spend a lot of time trying all kinds of different configurations, in my tests I had the FS105 etc with just two connections, one to my main gigabit switch and one to the streamer. That does not mean that is the only configuration that will work, just that I didn't test any other configurations.

 

One thing I did find is that it is important to leave an empty port next to the one going to the endpoint. For example if the endpoint is on port 1, don't plug anything into port 2. (its actually more complex than that, sometimes you CAN plug something in next to the endpoint connection, but always leaving the ports next to it free will guarantee you don't have a problem).

 

John S.

 

Do you have any explanation or theory to why it’s better to leave an empty port next to the one going to the endpoint?

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On 2017-10-17 at 10:22 PM, JohnSwenson said:

It seems a lot of people are getting confused by my posts on the subject of network leakage, I will try and state things in a more concrete manor.

 

This is relating to a switch which is what your network endpoint into your audio system is connected, this may be streamer (microRendu etc) a laptop or other computer Mac mini, PC etc). We shall call this the audio endpoint (AE)

 

Leakage current can get into the AE through the switch in two ways, from the power supply powering the switch, or from the cable connected to the rest of the network. The leakage coming from the network comes from the SMPS powering THOSE devices.

 

If the switch connected to the AE is powered by an SMPS, grounding the negative of output of the SMPS will shunt the leakage from that supply, but the leakage from the NETWORK will still go through. There is one exception, see the next section.

 

IF the switch is one of FS105 and FS108, grounding the negative of the supply will get rid of BOTH the leakage from the SMPS AND the network leakage. NOTE, this ONLY happens for these two switch types. Grounding the supply to a different switch type does NOT block network leakage.

 

If you are powering a switch from a linear supply, this gets rid of the leakage going through the PS of the switch, but NOT the network leakage. The only way to get rid of the network leakage is to use one of the above switches AND ground the negative of the supply powering the switch, no matter WHAT that supply might be. (linear, SMPS, LPS-1 etc)

 

If you are using an LPS-1 to power the switch, see the above rules for ANY supply. ANY supply includes the LPS-1. Thus IF you have one of the two named switches and you are powering the switch from an LPS-1, you must ground the output of the LPS-1 in order to block the network leakage. This will only work with one of those two switches. Grounding the output of the LPS-1 will NOT block leakage if you are using some other switch. It will not help if the LPS-1 is driving some other type of device. Thus there is no reason to ground the output of an LPS-1 if it is NOT driving one of the above named switches.

 

There is one exception to the last point. IF the LPS-1 is driving an ISO REGEN there can be a situation where the whole audio system is floating with respect to earth ground and a charge can build up which can show up as clicks and pops. ONE earth ground in such a system can alleviate this. ONE way to do this is to ground the negative of the supply powering the ISO REGEN. If this supply is an LPS-1 then you can try grounding the output of the LPS-1 to see if it fixes the clicks and pops.

 

Grounding the INPUT to the LPS-1 can help in other situations by shunting the high impedance leakage.

 

I hope this makes things clear, I'm running out of ways to say this.

 

John S.

 

 

 

 

 

 

 

If I get a switch with one of FS105 and FS108 and grounding the negative of the output of the SMPS I will get rid of both the leakage from the SMPS and the network leakage, ok. Is it best to have the switch on the same outlet as the AE or can I have it on another outlet next to the router or does it not matter?

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  • 2 weeks later...

If it’s one thing I have learned about building audio systems it must be; there are more than one way to skin a cat and what’s best in one audio system is not necessary best in another.

 

Take me for example. I have 2 computers, one big with JCAT NET card with all fans turned off and 1 fanless mini PC. In both cases the computer has been used only to store and play native files (no up sample) and with an uR/mR as output and Offramp 5 as DDC. My fanless mini PC is stock feed by an HDPlex 200W. To me the sound of my bigger PC is better even if it has a big ATX power supply. My experience tells me that getting a low power PC is not enough and many other things is of more importance. To put the computer in another room and power line make for best sound to me that have UltraRendu as EA, no matter which computer I use. The noise from a computer are not only traveling thru power and digital cables, computer radiate a lot of EMI and RF noise that are airborne.

 

Then building a mini-computer with “normal” computer parts some compromises has to be made. Placing all components like processor, ram, hard drive, mother board, switching regulators, clocks, inputs/outputs very close together means EMI and RFI will infect each other more than in a full- size box there they can be placed further apart. Components that are used in a stock NUC and laptops are also often of low capability and quality, use more switching power internally to keep heat and power low, use more non discreet components, cheap cables and painted circuit boards, one PSU for powering all components etc.

 

This doesn’t mean that’s I think that it’s impossible to make a dedicated mini audio PC/server even better than a full-size PC, I do, just that low power by itself is not more important than quality of parts. Both quality power, descent spaced box, PC in another room/AC and quality parts would be optimal I think.   

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On 2017-11-01 at 9:48 PM, ElviaCaprice said:

 

I don't follow this logic.  You say a bigger ATX mobo performs better than a mini PC (whatever that is), but then you use a renderer which is basically an extremely small computer.  So which is it, smaller or bigger? 

The renderer showed the way.  Smaller is better and even better yet better clocking along with quality power supply.  It's not hard to envision going direct from as small/low power as possible of a mobo modified via sCLK-EX as the ultimate endpoint. The trifecta (sCLK-EX) pointed the way.

Server=>DAC=>Speakers

 

No I didn’t say a bigger ATX mobo performs better than a mini PC. I merely exemplify that in my audio system, with an UltraRendu as a render and to my ear, low power in a PC is not as important as the quality of components in the PC. With a bigger PC one can more easily use quality parts like a JCAT NET card which also can be powered from a separate PSU.

 

It’s correct that the uR is basically an extremely small dedicated audio computer. With a NAA or render the PC (that handle Roon/Jriver and play files etc) and its power supply is of less importance, especially if placed in another room and on another AC. Nevertheless, to me, the PC still has an impact on the sound, reduced, but still audible. The noise coming from CPU, memory, hard drives, clocks, regulators and other hardware and software is not totally cured by any PSU or by moving it away. I know that here and on many other forum ATX power supply are believed to degrade sound compared to almost any LPS. I think it’s an exaggeration that are related to the fact that it is a switched power supply. Let me explain. Yes all other things held equal a good liner power supply is often better or even much better than a switched power supply. Is all other things equal in my and many others case? I don’t think so. With an HDPlex PSU my whole NUC is feed 12v or 19v and a lot of switching regulators has to transform it to proper voltage for the hard drive, USB, Ethernet etc etc. With an ATX power supply on the other hand the motherboard is feed its proper voltage and the hard drive its proper voltage the same with most other components. Less voltage regulation made by cheap switching regulation is maybe more important than a reasonable good switching ATX PSU?! Placing all components like processor, ram, hard drive, mother board, switching regulators, clocks, inputs/outputs very close together means EMI and RFI will infect each other more than in a full- size box there they can be placed further apart.

 

With a sCLK-EX, better PSU and superior regulators I guess most or all of the negative impact I can hear in my more or less stock NUC can be treated. So like I said, there are more than one way to skin a cat and what’s best in one audio system is not necessary best in another.

 

Food for thoughts. Is let say a small Paul Hynes LPS better than a much more powerful Paul Hynes LPS?   

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1 hour ago, Johnseye said:

 

It isn't low power as much as clean power, meaning as little noise as possible.  Low power helps achieve that.  I think clean power could have a bigger impact than many components.

 

 

This is a good question.  I don't know what challenges there are in building a higher power PSU.  This would be a good question for @Superdad

 

Yes it’s all about clean power, but it’s not the measured output on the PSU that is most important, it’s on the input of the components that are meant to be power that really counts. Hence the importance of less cheap switching voltage regulators.

 

A well-made ATX PSU, even if it capable to output many hundred watts, does not do that if you use reasonable low powered hard ware and software and only use it for streaming audio.

 

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6 minutes ago, Johnseye said:

 

 

So how can we determine when a motherboard or an external component has a high quality power delivery system?

 

By knowing the pros and cons of all component used or just listen while changing one component at the time, so called trial and error.  

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24 minutes ago, Johnseye said:

 

Trial and error is unrealistic.  The shear number of motherboards and components available, not to mention how quickly they product cycle makes it a near impossible task.

 

There must be specs and known information on build quality, component part quality and design that enable us to differentiate.  This is done with tubes.  It's also done to an extent with capacitors.  V caps and Mundorf come to mind.  

 

You can near it down by type and what others that have compared them to in an audio PC. One reason to share subjective

experience like we do here, collective knowledge. Take the memory for the OS as an example, the M2 that is commonly used in laptops and NUCs for its low power and small footprint and cheap price. It is known to make lot of high frequency noise. I read it and put the OS on my Samsung SSD instead and yes better, but still not as good as my big PC.

 

Yes you can of course use measurements as guidelines if can get hold on relevant data for audio and you also know well how to interpret them.

 

Yes it’s possible to measure some parameter on tubes and capacitors. But many of the best sounding tubes are NOS and doesn’t necessarily measure better, often it’s the opposite.

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37 minutes ago, ElviaCaprice said:

No emotion, here, Larry, just straight forward impressions from my own listening experience.  So where were you on the mechanisms involved???  Clean power is the only explanation?  Yet, you haven't even heard it yet, a sCLK-EX component?   Seems that in the spirit of this empirical thread it might be wiser to first have tried the component in question before looking for scientific answers.  Plenty of empirical evidence point to the sCLK-EX being what is said, here in this thread.  If you still question it's value, then I doubt any explanation scientifically is going to sway you to want to take action and go the sCLK-EX route yourself.

 

Which other streaming solutions have you personally compared your sCLK-EX against?

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  • 1 month later...
  • 3 weeks later...
9 hours ago, austinpop said:

 

I think we can also credit the isolation built into the Zenith SE ethernet port. As Nuno wrote on this other forum:

 

The ethernet ports on the Zenith are no ordinary Ethernet ports you find on normal "consumer-grade" boards. The ethernet ports have 4 separate isolation transformers each to further isolate the connection (4 small grey boxes right before each ethernet port). We also do extensive anti-vibration treatment of the chassis, the LPSU and even the SSD, converting vibration into very small amounts of heat. This makes a very audible difference in sound quality.

 

The Ethernet board is important, no doubt. It made a big difference going from mobo Ethernet to JCAT LAN. Probably of even more significance if bridging them like I do.  

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  • 2 weeks later...
2 hours ago, austinpop said:

Update: Zenith SE with or without sMS-200ultra

 

One difference between @flkin and @Kritpoon's recent findings with the Antipodes DX3 and the SOtM trifecta, and @limniscate's and mine with the Zenith SE, has to do with the presence of the sMS-200ultra.

 

If you'll recall, before I got the SE, my optimal configuration was:

  • Router > Win 10 bridged > shunted switch > modded switch > modded sMS-200 > ISO-Regen > tX-USBultra > Codex DAC
  • Cybershaft OP-14 providing the reference clock to the trifecta (switch, sMS, tX) via the tX master clock input
  • Roon Core on Win 10, music files on NAS

After much experimentation, I determined the new optimal configuration (i.e. with maximum SQ) was:

  • Router > shunted switch > modded switch > Zenith SE > tX-USBultra > Codex DAC
  • Cybershaft OP-14 providing the reference clock to the tX and switch via the tX master clock input
  • Roon Core on Zenith SE, music on local SSD

Based on this finding, I sold my ISO-Regen and modded sMS-200, along with their respective LPS-1 PSUs. 

 

Fast forward to this week. Since I had ample PSUs on hand, along with Eric's Ref 10 and sMS-200ultra, I decided to revisit the question of whether the sMS-200ultra still has a place in a Zenith SE topology. This is in light of @flkin's observation that in their experiments, they found the combo of Antipodes AND SOtM trifecta to sound the best. In the context of my system, the question can be framed as follows: Does the addition of the sMS-200ultra in the path add or detract SQ? I compared the following 2 topologies:

  • Baseline SE chain: Router > shunted switch > modded switch (LSP-1.2) > Zenith SE > tX-USBultra (SR-4) > Codex DAC, vs.
  • SE with full trifecta: Router > shunted switch > modded switch (LPS-1.2) > Zenith SE > modded switch (LPS-1.2) > sMS-200ultra (LPS-1.2) > tX-USBultra (SR-4) > Codex DAC
  • NOTE: there is only one modded switch - the reason it is shown twice in the chain above is to indicate data flow. Connectivity-wise, both the SE and the sMS-200ultra attach to the modded switch
  • Both sMS-200ultra and tX-USBultra reference clocked by the Ref 10

Result: The baseline SE configuration sounds better, as before. The introduction of the sMS-200ultra back into the chain seems to add a small bit of harshness and a flattening of the image, while delivering no perceptible increase in resolution, as one would expect with another reclocked component in the chain. There is a certain calm, inky-black smoothness to the SE that gets lost when you add the sMS-200ultra back in.

 

I don't really think that, based on this, one can make any associative conclusions of the DX3 vs the SE. I would caution against it. For my part, I am just glad that I could recoup some of the expense of the SE by selling the modded sMS, ISO-R, and 2 LPS-1s!

 

I presume you perceive some SQ gain with a shunted switch directly followed by a modded switch. But do you have any insight to why and can’t the modded switch be shunted instead or is the reason for having one switch followed by another switch not to get the benefit of a shunted switch?

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18 hours ago, austinpop said:

 

This convoluted setup was prompted by John Swenson's findings with regards to switches, SMPSes, and ground shunting. He has several postings across various threads that cover this, but this one from my bookmarks is probably a good one for an overview in regard to switches:

https://www.computeraudiophile.com/forums/topic/30376-a-novel-way-to-massively-improve-the-sq-of-computer-audio-streaming/?do=findComment&comment=730832

 

And this one gets into his observation that only some switches seem to block noise through the magnetics in each port:

https://www.computeraudiophile.com/forums/topic/35129-a-novel-way-to-massively-improve-the-sq-of-the-microrendu-ultrarendu/?do=findComment&comment=723196

 

The bottom line is that we don't know the universe of switches that block noise correctly. John has tested and confirmed that the Netgear FS/GS 105/108 switches are good. 

 

Given this data, the question for me was - what about my modded Zyxel switch? Obviously, if I could get John to test it on his measurement setup, I'd know. But John has made it clear he can't spend his entire waking life testing switches, and who can blame him! 

 

In considering the choices, it was actually John who came up with the suggestion of preceding the unknown (leakage characteristics) switch with a known one like the Netgear. 

 

As for SQ, when I added the Netgear prior to the modded Zyxel, I wasn't sure I heard much of an SQ difference at all. In keeping with my practice of periodically revisiting my optimizations to confirm they still matter, I will redo the experiment with and without the Netgear in my latest setup to see if it actually helps SQ or not.

 

Okay, let see if it’s better with or without the Netgear switch, when you have done some testing.  

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  • 2 weeks later...
21 hours ago, Solstice380 said:

 

I agree that we’d expect the regens to provide enough current - P10 is suppose to spit out 75A for some amount of time - but you have me curious again to try my amp direct.  Like you, I’m thinking we also under estimate the requirements of our components on an instantaneous basis.  Do we need the rise time of lightning?  ⚡️? ?

 

The spec of regenerator like PS audio P5 or P10 doesn’t show the amount of idle current/voltage/resistance. A P10 is only regenerate a small amount of idle power and if more power is needed it will regenerate more. If a P10 would have all 75 A on tap (all the time) it would consume a lot of power, get very hot and be much less efficient.

 

It’s maybe easier to understand a power regenerator if compare to an amp. A class A-amp with high bias always have all power on tap and don’t need any microprocessor to regulate the amount. A class A/B amp always have some power on tap (low bias) and a microprocessor that regulate than more power is needed. A class D amp have no power on tap idle and its microprocessor regulate the output power from 0 to 100 %.

 

A bigger power regenerator probably have a bit more power on tap all the time. But like with amps it’s not only the bias or quantity of power that’s of importance, the quality, power swing, impedance, resistance and filters/DC blockers used and so on are of equally significance.

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On 2018-03-25 at 3:53 PM, BigGuy said:

IIRC, I was told that the early versions like the P300, P500 were Class A based and as expected ran very hot.  In fact, a second version of the P300, etc had a cooling fan.  I have been using the P300 for front end components since it was first offered and have been using external cooling to extend the unit life.

 

I was under the impression that the latest series P5, P10, and now P20 might be Class D since even the largest does not have cooling?!

 

P300 and P500 was not class A, they was high biased class A/B with 50 percent efficiency.

 

I don’t know much about the bias in the new line of regenerators from PS Audio. What I know is they have a new type of DAC in the latest series of P5, P10, and P20 that generates DSD instead of PCM.  

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  • 1 month later...
10 hours ago, Superdad said:

 

I’ve had a Spring L3 for at least 20 months.  It took 6 weeks of playing before I thought it was anything special (for a long time even my old base version Geek Pulse had better bass than the Spring).  But then it really started to open up.  So do give it time. 9_9

 

Do you play NOS or with OS?

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  • 5 weeks later...
15 hours ago, davide256 said:

hmm, bought a  Startech GigE NIC (PCI) to try the bridged connection with my onboard GigE NIC (PICe), works wonderfully for audio resolution improvement as per start of thread but getting episodic BSOD crashes, EXPOOL_CORRUPTED. Have on  order an Intel dual port NIC (PCIe), hoping the cause is just a bad Realtek driver for bridging use that will go away with just the Intel NIC's in service. Wondering if anyone else has seen like problems with adding a second NIC?  Only other variable

is that USBridge link is FE vs router link is GigE

 

If you have update to version 1803 in Windows 10 you perhaps lost the file sharing on home network function. I did and could not use the bridged connection anymore.

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  • 1 month later...
12 hours ago, Cornan said:

Is’nt this NUC thing all about isolation? You can use any fancy words that you like (latency or what ever) but in the end it is all about isolating the music source, core and streamer from each other and from the ac mains in any possible way. Same rules applies to Tidal (source router which needs to be isolated) as it does to local files (source harddrive/NAS which needs to be isolated). The more you can separate and isolate each part of the chain the better (also inbetween devices ofcourse). If you start looking at it in terms of isolation instead of all the other ”words” it is way easier to understand where you are heading IMO. Just my 2 c.

 

ps. Isolation is the least possible impact to the signal and power chain. 100% isolation is just fiction. ds. 

 

I think it’s about using devices that generates as little noise as possible to start with. The next thing to do is to isolate the noise that we can’t prevent from being produced. Sound very easy in theory, but is very hard in practise because in reality we have to deal with all the things that can have a negative effect on other devices in the audio system, but that is not well known/specified as being as source of noise in literature/theories.

 

In principle we can illustrate this by having one X-axel and one Y-axel, there all types of noise generated by a device (or a whole system) can be put on the X-axel and the all types of isolation can be summed on the Y-axel.

 

Even if I think that you are in some way correct in that “we” can use “fancy words” to explain some specific aspect, like the different types of noise there is. OTOH when we discuss real specific implications instead of general theories it’s often necessary to distinguish between different types and sources of noise, otherwise it’s next to impossible to understand and do something about them.

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