Jump to content
IGNORED

exaSound e18 - e20 - e28 - Info and Experiences Post All Here


Recommended Posts

They were! The provenance of those tapes is what is at question.

 

Come away with me was recorded on Analog tape in Studio A at Sorcerer sound studio

 

http://www.sorcerersound.com/studio_a.htm

 

not 100% sure where Feels like home was recorded but was from analog sources too.

 

The rest were recorded at her home studio. Digital PCM files were sent for mixing and mastering to various studios from there.

 

You'll notice if you read the descriptions closely of the individual DSD albums on Acoustic Sounds that on "Come away with me" says from "original Analog tapes". Some of the others say "Original sources". That's usually a dead giveaway because they always make sure they mention it if they are from the Analog master tapes.

Link to comment
NativeDSD has literally hundreds of them. :)

 

An the other subject, I have found that the e12 sounds like it's sweetspot is DSD256 to my ears, and that HQplayer upsampling everything (that includes PCM) using the right modulators and filters, to DSD256 is the best I can make that DAC sound. A really musical DSD engine for under $2k. Yes, it plays PCM fine too, but DSD256 seems to be a panacea.

 

Try upgrading the clock to the E22 clock.

Link to comment
I brought up the topic of software claiming to be better at applying this algorithm than his chip. I asked him if the best algorithm theoretically possible was invented, where would the best place to apply it be? He said by far the #1 place is in the chip. If designed into the chip, the chip can do it much better than anywhere else. #2 would be inside a FPGA. #3 would be on a computer.

 

He also explained that there's much more going on in that chip than most people understand. Long story short the final word was it couldn't be done better than his implementation in his chip, if his chip was part of the playback chain.

With respect to him, thats exactly the answer I would expect from the designer of the Sabre chip. If he'd have suggested #2 or #3 that would have been a more interesting reply and more worthy of note as a definitive answer.

 

I'm sure if you ask Charles Hansen of Ayre or George of exaSound they would offer that #2 is far superior to doing it "on chip" and Miska of Signalsyt / HQPlayer would say that #3 has the potential to beat anything else as it offers much greater processing power.

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

Link to comment

Apologies in advance for the OT, folks. I'd suggest if anyone wants to talk further about filters, algorithms, etc., after this that we make a new thread for it.

 

This is where the grey area lies. Who has the best algorithm, and if you have the best algorithm, where's the best place to apply it?

 

This topic came up a couple weeks ago when the designer of the Sabre chip came over for coffee. (He happens to live down the road from me. I'm not going to mention names on a public forum)

 

The names of folks on the SABRE chip design team are public knowledge, but it's certainly up to you whether you care to disclose names or not. Several of them, including the team leader, are now with Resonessence, whose excellent page discussing digital filters I like to link from time to time: Digital Filters | Resonessence

 

I brought up the topic of software claiming to be better at applying this algorithm than his chip. I asked him if the best algorithm theoretically possible was invented, where would the best place to apply it be? He said by far the #1 place is in the chip. If designed into the chip, the chip can do it much better than anywhere else. #2 would be inside a FPGA. #3 would be on a computer.

 

Since the advantage of computer software is precisely that more sophisticated algorithms can be applied there, this is something like saying "If a Rolls Royce had the speed and handling of a Ferrari, which would you prefer to take for a long drive in the country?" You're taking the primary reason for choosing one of the options and assuming it out of the picture with the form of your question. Given the problem with the question, the answer does not provide fully informative guidance.

 

I am actually rather surprised he answered your question as asked, because as the Resonessence digital filter page I've linked above itself makes quite clear, there cannot be any "best" algorithm. This is due to the sheer mathematics involved in filter design:

 

All the various trade-offs discussed here...show us a fundamental relationship: as a filter is designed to be optimum in the time domain, it cannot be optimum in the frequency domain. The two are related in a very fundamental way, it is the same relationship that governs the whole world of physics....

 

In our perhaps less profound world of audio, the behavior in frequency is the Fourier transform of the behavior in time, and because of this, one gets worse as one gets better.

 

Another layer of this is that the mathematics governing digital filter design are not necessarily identical to those that govern our ear/brain system. As Resonessence has found, the on-chip filters they consider to be mathematically closest to "perfect" are not the same as those listeners prefer:

 

Again, we just have to trust the listening process, and that shows us (and certain key customers who act as beta-testers for us) that there is a predictable preference for what is not a mathematically precise filter in the listening experience. Interestingly, not all listeners seem to agree, some choose the mathematical perfection of the built-in Sabre filters, but we judge the majority of our key customers choose a slightly different filter, and in response to this, software versions 2.1 and later provide a choice of seven filters.

 

(Note, by the way, that they can't resist the marketing hyperbole of referring to the SABRE chip's built-in filters as "mathematical perfection" on the very same page where they say such perfection is mathematically impossible. :) )

 

He also explained that there's much more going on in that chip than most people understand. Long story short the final word was it couldn't be done better than his implementation in his chip, if his chip was part of the playback chain.

 

When designing DAC chip's, the #1 goal needs to be preserving the original source material in the most linear way possible. The result of this doesn't necessarily mean the most enjoyable listening experience possible, because if the source happens to be poor quality, the end result will be poor quality. This is where these software based algorithms can differ. They can be designed to sprinkle a bit of fairy dust on the music. This is the reason that some might find these algorithms superior.

 

Now if using these software algorithms brings you a more enjoyable listening experience, then that's great, use the software and enjoy it. But keep in mind that if the same algorithm was being applied in the chip, it would be able to do a better job.

 

That the chip would do the best job of implementing the "perfect" algorithm (caveat that there is no perfect algorithm, and that a chip lacks resources for implementing sophisticated algorithms versus computer software) is presented here as essentially an argument from authority: "Long story short, my knowledgeable designer friend says so." What about isolating the rest of the sensitive DAC electronics from the processing involved in implementing the algorithm? Many people (myself included) think offline conversion software has an advantage because of precisely this sort of consideration.

 

I'd like to hear more of the long story, i.e., I'm (as always) curious to learn more. If you could possibly persuade your friend to think about guest authoring an article here at CA, I'd very much look forward to it.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

Link to comment
With respect to him, thats exactly the answer I would expect from the designer of the Sabre chip. If he'd have suggested #2 or #3 that would have been a more interesting reply and more worthy of note as a definitive answer.

 

I'm sure if you ask Charles Hansen of Ayre or George of exaSound they would offer that #2 is far superior to doing it "on chip" and Miska of Signalsyt / HQPlayer would say that #3 has the potential to beat anything else as it offers much greater processing power.

 

Regardless of who has the superior algorithm, heavy lifting tasks can be preformed much higher efficiently and with much higher precision on chips and FPGA's. This is a well known fact regardless of what anyone may have biases towards. But this is way off topic so let's end this discussion on this thread.

Link to comment
Apologies in advance for the OT, folks. I'd suggest if anyone wants to talk further about filters, algorithms, etc., after this that we make a new thread for it.

 

 

 

The names of folks on the SABRE chip design team are public knowledge, but it's certainly up to you whether you care to disclose names or not. Several of them, including the team leader, are now with Resonessence, whose excellent page discussing digital filters I like to link from time to time: Digital Filters | Resonessence

 

 

 

Since the advantage of computer software is precisely that more sophisticated algorithms can be applied there, this is something like saying "If a Rolls Royce had the speed and handling of a Ferrari, which would you prefer to take for a long drive in the country?" You're taking the primary reason for choosing one of the options and assuming it out of the picture with the form of your question. Given the problem with the question, the answer does not provide fully informative guidance.

 

I am actually rather surprised he answered your question as asked, because as the Resonessence digital filter page I've linked above itself makes quite clear, there cannot be any "best" algorithm. This is due to the sheer mathematics involved in filter design:

 

 

 

Another layer of this is that the mathematics governing digital filter design are not necessarily identical to those that govern our ear/brain system. As Resonessence has found, the on-chip filters they consider to be mathematically closest to "perfect" are not the same as those listeners prefer:

 

 

 

(Note, by the way, that they can't resist the marketing hyperbole of referring to the SABRE chip's built-in filters as "mathematical perfection" on the very same page where they say such perfection is mathematically impossible. :) )

 

 

 

That the chip would do the best job of implementing the "perfect" algorithm (caveat that there is no perfect algorithm, and that a chip lacks resources for implementing sophisticated algorithms versus computer software) is presented here as essentially an argument from authority: "Long story short, my knowledgeable designer friend says so." What about isolating the rest of the sensitive DAC electronics from the processing involved in implementing the algorithm? Many people (myself included) think offline conversion software has an advantage because of precisely this sort of consideration.

 

I'd like to hear more of the long story, i.e., I'm (as always) curious to learn more. If you could possibly persuade your friend to think about guest authoring an article here at CA, I'd very much look forward to it.

 

I agree way off topic for this thread.

Link to comment
Many people (myself included) think offline conversion software has an advantage because of precisely this sort of consideration.

 

Regarding offline conversion, my take has not been to trade any quality for speed, but instead optimize the algorithms to run as fast as possible using hand-writter assembler code. And rest is up to having a fast enough computer... I don't need to make things run on any particular CPU, if something doesn't run today, Moore's law will make it feasible in future.

 

I feel that we are roughly at the computing speed where we can do most of the things we would want to at best possible quality at typical high end audio electronics price level.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

Link to comment
Agreed, but I think Greg and Kevin are starting with the same analog tapes...wherever they came from. :)

 

Exactly.

 

I have been told that the session recording engineer (Jay Newland) on Come Away With Me by Norah Jones "recorded that release in ProTools at 48kHz. Any analog tapes would have to be copies of the 48kHz PCM masters."

Link to comment
Regarding offline conversion, my take has not been to trade any quality for speed, but instead optimize the algorithms to run as fast as possible using hand-writter assembler code. And rest is up to having a fast enough computer... I don't need to make things run on any particular CPU, if something doesn't run today, Moore's law will make it feasible in future.

 

I feel that we are roughly at the computing speed where we can do most of the things we would want to at best possible quality at typical high end audio electronics price level.

 

I would like to have the price of typical high end audio electronics to update my computer at the moment, but that will have to wait. :)

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

Link to comment
I would like to have the price of typical high end audio electronics to update my computer at the moment, but that will have to wait. :)

 

All depends on what you intend to do. If you'd like to run upsampling for 8 channels to DSD512 and run 4-way DSP cross-overs with room correction up to DSD256 input rates, you are going to need a hefty machine in price range of typical high end equipment ($5k or more).

 

But instead if you just like to upsample stereo, then you are fine with a normal PC...

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

Link to comment
If you'd like to run upsampling for 8 channels to DSD512 and run 4-way DSP cross-overs with room correction up to DSD256 input rates

 

Yes, please :)

 

I'll add tube amps to these too :P

Dedicated Line DSD/DXD | Audirvana+ | iFi iDSD Nano | SET Tube Amp | Totem Mites

Surround: VLC | M-Audio FastTrack Pro | Mac Opt | Panasonic SA-HE100 | Logitech Z623

DIY: SET Tube Amp | Low-Noise Linear Regulated Power Supply | USB, Power, Speaker Cables | Speaker Stands | Acoustic Panels

Link to comment

Later, I played the 1x DSD in my normal playback mode, which is on the fly conversion to 88k PCM, bass management xovers to my sub at 60 Hz and Dirac Live room correction. That was to us the clear winner sonically, easily exceeding and distinguishing itself positively from the 4 unprocessed originals. Some of the reasons for that are, I am sure, obvious.

 

I am sure that discussion, caveats or disagreement with this could rage on for pages. But, those are the honest anecdotal reactions from our listening that night.

 

I do not think the matter is closed for all time based on this. But, for now, it seems to confirm that my normal listening mode and signal processing may be optimal to my ears in my system for now vs. purer or higher rez source material.

 

Always a good thing to test in one's own system and find the sweet spot.

 

In my own, I have so far preferred doing organic acoustic treatment for the room response. In turn, this allows to avoid DSP processing for room treatment, and furthermore, I actually turn off my amp's DSP processing when listening to DSD.

 

In my system, DSD to my native DSD DAC sounds best, and I would not want to convert to PCM for processing. I used to do that with Audirvana+ but that was because the only external DAC I already had at the time was my soundcard which is only PCM capable.

 

I could envision using Miska's pure DSD-domain processing if necessary though. Cross-over capabilities for several filtered channel would be highly interesting to me.

Dedicated Line DSD/DXD | Audirvana+ | iFi iDSD Nano | SET Tube Amp | Totem Mites

Surround: VLC | M-Audio FastTrack Pro | Mac Opt | Panasonic SA-HE100 | Logitech Z623

DIY: SET Tube Amp | Low-Noise Linear Regulated Power Supply | USB, Power, Speaker Cables | Speaker Stands | Acoustic Panels

Link to comment

Here's a weid one I thoiught I'd ask here:

So I have two exaSound dacs inhouse, the e28 I have owned now for years, and the new e12 I reviewed a couple months ago. It has been my goal to comapre the stereo (2 channel) performance of these two dacs, since I has quite impressed with the lowly but more modern e12.

 

But first I had a shorter term goal; that is to move my HQplayer NAA running on WS2012 AO GUI mode to the more lightweight minimalist mode of Minimal Server. I quickly learned that in order to use the e12 there I had to load the exaSound.N1.exe console first (something GUI mode did automatically, thanks Phil) so I wrote a simple bat file to do that and, voila, the e12 runs nicely in minimal server mode.

 

So now it was time to simply move the USB cable to the e28. Wrong! No matter what I do (reboot HQP or NAA, reload the console, etc) NAA reports that the e28 does not handshake with the NAA ("ASIO not loaded, etc etc"). HQPlayer has the driver (as well as many others, including the 8 channel exaSound driver) in its driver list for NAA, but no music and no NAA reporting ASIO sample rates. Yet 5 minutes ago I just added a powered Amber Regen to the mix and lo and behold the handshake occurred. WTF? Are the USB (in my case JCAT card) power requirements different for the e28 to the e12?

 

So tonight I will compare once and for all. The e12 with parallel chips in play, the e28 with chip duties spread across 8 channels but slightly better spec'd analog stage and power filtering.

Link to comment
Here's a weid one I thoiught I'd ask here:

So I have two exaSound dacs inhouse, the e28 I have owned now for years, and the new e12 I reviewed a couple months ago. It has been my goal to comapre the stereo (2 channel) performance of these two dacs, since I has quite impressed with the lowly but more modern e12.

 

But first I had a shorter term goal; that is to move my HQplayer NAA running on WS2012 AO GUI mode to the more lightweight minimalist mode of Minimal Server. I quickly learned that in order to use the e12 there I had to load the exaSound.N1.exe console first (something GUI mode did automatically, thanks Phil) so I wrote a simple bat file to do that and, voila, the e12 runs nicely in minimal server mode.

 

So now it was time to simply move the USB cable to the e28. Wrong! No matter what I do (reboot HQP or NAA, reload the console, etc) NAA reports that the e28 does not handshake with the NAA ("ASIO not loaded, etc etc"). HQPlayer has the driver (as well as many others, including the 8 channel exaSound driver) in its driver list for NAA, but no music and no NAA reporting ASIO sample rates. Yet 5 minutes ago I just added a powered Amber Regen to the mix and lo and behold the handshake occurred. WTF? Are the USB (in my case JCAT card) power requirements different for the e28 to the e12?

 

So tonight I will compare once and for all. The e12 with parallel chips in play, the e28 with chip duties spread across 8 channels but slightly better spec'd analog stage and power filtering.

 

That will be a great comparison. Do you have the upgraded clock in your E28?

 

Another +1 for the E12 is the 1/3rd power consumption. When used with your linear power supply it won't be working as hard. This makes a huge difference in my experience.

Link to comment
That will be a great comparison. Do you have the upgraded clock in your E28?

 

Another +1 for the E12 is the 1/3rd power consumption. When used with your linear power supply it won't be working as hard. This makes a huge difference in my experience.

 

My e28 is the original (I think :) ). One exaSound is powered by the Hynes SR3-12 and the other by the Uptone JS-2 (stupid me my NAA is powered by the other side of the JS-2..I could have had both DACs with same ps and had NAA with Hynes but right now don't want to turn off either..I am anal about the digital clocks).

Link to comment
My e28 is the original (I think :) ). One exaSound is powered by the Hynes SR3-12 and the other by the Uptone JS-2 (stupid me my NAA is powered by the other side of the JS-2..I could have had both DACs with same ps and had NAA with Hynes but right now don't want to turn off either..I am anal about the digital clocks).

 

Clock upgrades are a massive improvement. Try it and see for yourself. A $30 part can make a bigger difference than $1000's spent elsewhere. So easy to do as well.

Link to comment
Clock upgrades are a massive improvement. Try it and see for yourself. A $30 part can make a bigger difference than $1000's spent elsewhere. So easy to do as well.

 

Are you saying it's field upgradable with no solder iron? Where does one get a compatible femto clock chip?

Link to comment
Are you saying it's field upgradable with no solder iron? Where does one get a compatible femto clock chip?

 

 

 

Everything's field upgradable if you know what your doing :)

 

Yes it is a soldered in piece, but it's not a hard job with the right tools and experience. I recommend using a air gun as it's a SMD component. If you have no experience with this, any electronic service shop can do it. I took my Exasound to one of these places and he did it while I waited. Took him 5 minutes and he charged me $25. So was a $49 investment altogether, but brought my E20 into E22 territory.

 

The clock in the E20,E28 and E12 is a Crystek CVHD 950-100. What really matters when it comes to clock specs is phase noise in the range of human hearing. The whole "Femto" clock talk is just a bunch of marketing hype. It is based on a completely useless spec when it comes to audio. It's the jitter at 100KHZ spec. 100 KHZ is far above the range of human hearing. Jitter gets worse the lower you go in frequency. Phase noise represent jitter. So the go to spec when it comes to audio clocks is the 10hz phase noise spec. It's a negative number so the higher it is means lower noise, so higher number means better.

 

The Crystek CVHD 950-100 has a 10hz phase noise spec of -86DBc/HZ. It's pretty decent. The new "Femto" clock Exasounds uses (along with a bunch of other companies like Wyred 4 sound) uses the Crystek CCHD-575-50-100. It's 10hz phase noise is 90DBc/HZ. Only a difference of 4 Db but enough to make a drastic improvement in sound.

 

The 575 series clocks are a smaller form factor, but the contact pads are exactly the same so they line up to the same pads on the Exasound board.

 

Where to buy? they sell them all over but digikey is a good place.

 

Here's the CVHD 950-100:

 

CVHD-950-100.000 Crystek Corporation | 744-1213-ND | DigiKey

 

And here's the "Femto" CCHD 575-50-100

 

 

CCHD-575-50-100.000 Crystek Corporation | 744-1454-ND | DigiKey

 

 

Wow look at that. Exactly the same price right to the penny. Why would they even offer the lower performance clock since they are both the same price anyways?

 

1: Marketing. (good reason to buy the E22 instead)

 

2: Maybe use up old stock of the CVHD-950-100's?

 

Who knows.

Link to comment

The clock in the E20,E28 and E12 is a Crystek CVHD 950-100. What really matters when it comes to clock specs is phase noise in the range of human hearing. The whole "Femto" clock talk is just a bunch of marketing hype. It is based on a completely useless spec when it comes to audio. It's the jitter at 100KHZ spec. 100 KHZ is far above the range of human hearing. Jitter gets worse the lower you go in frequency. Phase noise represent jitter. So the go to spec when it comes to audio clocks is the 10hz phase noise spec. It's a negative number so the higher it is means lower noise, so higher number means better.

 

The Crystek CVHD 950-100 has a 10hz phase noise spec of -86DBc/HZ. It's pretty decent. The new "Femto" clock Exasounds uses (along with a bunch of other companies like Wyred 4 sound) uses the Crystek CCHD-575-50-100. It's 10hz phase noise is 90DBc/HZ. Only a difference of 4 Db but enough to make a drastic improvement in sound.

 

This assertion (the "lower freq jitter performance is more important for SQ") matches what I read a while back on the DIY forum, where people were looking at several different clock types with interesting specs. I'm not a digital engineer, but if as you say the 575 clocks have 4dB better performance than the 950s at 10Hz, I'd expect the performance and thus SQ to be much better still in the KHz range and above--and to my ears, that's the range where timing anomalies are most noticeable and annoying. Vocal consonants, cymbals, upper registers of piano and especially violins, with all their overtone series, and spatial cues. The system that can unsmear a well-recorded baroque violin section while retaining the rosin of the attacks and hall bloom so that it sounds like a blend of individual instruments in a recognizable space and not an amplified dentist's drill is the system I can love.

Link to comment
This assertion (the "lower freq jitter performance is more important for SQ") matches what I read a while back on the DIY forum, where people were looking at several different clock types with interesting specs. I'm not a digital engineer, but if as you say the 575 clocks have 4dB better performance than the 950s at 10Hz, I'd expect the performance and thus SQ to be much better still in the KHz range and above--and to my ears, that's the range where timing anomalies are most noticeable and annoying. Vocal consonants, cymbals, upper registers of piano and especially violins, with all their overtone series, and spatial cues. The system that can unsmear a well-recorded baroque violin section while retaining the rosin of the attacks and hall bloom so that it sounds like a blend of individual instruments in a recognizable space and not an amplified dentist's drill is the system I can love.

 

Yes of course jitter is improved at all frequencies. The higher you go in frequency, the lower the jitter.

 

Checkout the phase noise plot for the 575-50-100ImageUploadedByComputer Audiophile1437489640.429943.jpg

Link to comment
  • 3 weeks later...
linux driver - have checked with exaSound - they did respond quickly saying it's not available. not a major deal breaker i guess for a player (be it squeezelite or HQ or something else that people love to use) windows and mac os are providing more then enough options.

it's just me liking the linux :)

 

on the other side they do provide extra features with their driver(s) for win and mac so one can complain but not too much :)

 

I'm looking at several small size options for audio serving to place on my audio rack next to my Exasound E22 - ethernet to small computer, then USB to Exasound. My options today force me to add $140 for Windows Pro (for remote desktop control) to a $300 computer build, instead of being able to stand up a cheap tiny single purpose Linux computer. I find I can control what all goes in in a Linux box much better than I can playing whack-a-process with Windows.

Link to comment
Almost every review I read of the e22 gushes about how great the DSD is. Am I the only one who prefers it doing redbook? Admittedly DSD has never really sat well with me, but still, even with that the exa is a phenomenal PCM DAC. Do people maybe think it wouldn't be worth the price as a good PCM DAC? I think it is...

 

DSD does sound wonderful on my Exasound E22, but I don't think it sounds better than 192/24, or the higher rate PCM. (Distracted right now listening to some great music... so can't remember...) But I definitely agree with you that it makes redbook sound much better than any other dac I've used.

 

The thing that really strikes me about it - and anyone else who's heard my system, even though I don't bring it up, they do - is how it has its way with live music. It puts the whole band or orchestra right there in the room, and makes the vocals almost tangibly 3D. Whatever it's doing with the fine detail that got captured about the recording space and ambient noise, which there isn't much of in studio recordings, is pretty amazing. Some of the Chesky binaural recordings are almost creepy in dimensionality, they've got a lot of ambient space content in them.

Link to comment
The thing that really strikes me about it - and anyone else who's heard my system, even though I don't bring it up, they do - is how it has its way with live music. It puts the whole band or orchestra right there in the room, and makes the vocals almost tangibly 3D. Whatever it's doing with the fine detail that got captured about the recording space and ambient noise, which there isn't much of in studio recordings, is pretty amazing.

 

You should list your system. I'd love to know what choices in gear and cabling helped you achieve this outcome.

Digital:  Sonore opticalModule > Uptone EtherRegen > Shunyata Sigma Ethernet > Antipodes K30 > Shunyata Omega USB > Gustard X26pro DAC < Mutec REF10 SE120

Amp & Speakers:  Spectral DMA-150mk2 > Aerial 10T

Foundation: Stillpoints Ultra, Shunyata Denali v1 and Typhon x1 power conditioners, Shunyata Delta v2 and QSA Lanedri Gamma Revelation and Infinity power cords, QSA Lanedri Gamma Revelation XLR interconnect, Shunyata Sigma Ethernet, MIT Matrix HD 60 speaker cables, GIK bass traps, ASC Isothermal tube traps, Stillpoints Aperture panels, Quadraspire SVT rack, PGGB 256

Link to comment

Create an account or sign in to comment

You need to be a member in order to leave a comment

Create an account

Sign up for a new account in our community. It's easy!

Register a new account

Sign in

Already have an account? Sign in here.

Sign In Now



×
×
  • Create New...