Jump to content
austinpop

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

Rate this topic

Recommended Posts

2 hours ago, mozes said:

System update: Radical change!

 

Well, I think it's about time to share an update about my system although it is still not where I want to end up with. Anyhow, I guess it is always WIP as usual.

 

Over the last few months, I have read various posts by @Romaz and @ElviaCaprice regarding their minimalist approach to their audio chain, that is going straight from DAC to speakers and the advantages this strategy yields. I became very intrigued given the isolation from AC mains, reduction of box count and increased transparency.

Of course, not any dac or speaker will work, so I had to make some tough decisions, and in the end I made up my mind. I sold my Naim gear, Brooklyn Dac and its VR Mini PSU, and replaced them with Chord Dave. Second, I bought two sets of speakers, Spendor D7 and Omega Super 8XRS with silver/gold wiring and high end Furutech binding posts. Why did I buy the Spendors and they are 90db only at 8 Ohms impedance? simply I fell in love with these speakers. They are one of the best speakers that engaged me regardless of price and I couldn't resist pulling the trigger. It seems, I have a weakness to everything made in the UK, having owned Naim, Neat, Chord and Spendor and who knows what more.

 

In the meantime, I also hunted a second hand tX USB ultra and I sent both tXs to SOTM who upgraded the DC wiring to silver, added Master clock inputs and modded a D-link switch from one of the tX units. I also got a 1m Lush USB cable that I added to my chain. To cut the story short, I am not going to talk about the many many tests I did and all the combinations I tried. I will simply share what my ultimate chain is so far which gave me the best sound ever till now :)

 

Nimitra (VR mini)>USPCB>Iso-Regen(LPS-1)>Elijah Audio USB cable (signal and ground only)>tX-USBultra(LPS-1)>USPCB>tX-USBultra(LPS-1)>1m Lush USB cable>Chord Dave>Spendor D7

 

Grounding: My Nimitra and Dave are grounded to Entreq Olympus Minimus and D-Link switch is grounded to Silver Minimus. Each of the D7 speakers have its negative speaker binding post grounded to an Aucharm Box.

 

This setup simply blows away anything I have had before in my system and I have tried real high end Naim gear (up to 500 series level). What I am getting is transparency and resolution like crazy! Music sounds just real, I have no better description.

 

The crown jewel of my system is undoubtedly the Dave, this little box amazes me. It is my dac, preamp and power amp and it gloriously drives 90db speakers to levels beyond my tolerance level and there is no need for any subwoofer. How can this little beats do that? only 2 watts!! I guess the answer is as @Romaz highlighted before, its ultra low impedance of only 0.055 ohms. I also bought a Rogue Cronus Magnum 2 tube amp to see what tubes offer. I tried Dave with the Rogue and it sounds very very sweet, but I prefer Dave directly to the D7s, I simply didn't want to compromise on transparency and resolution. With Dave driving the D7s directly, I put my ears on the tweeters and there is no noise at all, I mean literally ZERO noise, this is crazy, I haven't had any amplifier that it is DEAD quiet. This explains why I am enjoying now low level late night listening, as I feel that all the details are there, hard to describe what I am hearing, it is pure enjoyment.

 

My Tidal now sounds way better than my locally stored music in my previous system. You won't believe that this is streaming, it is like high res playing. The modded switch with Dave takes Tidal to a completely different level.

For now, I am waiting for the delivery of my Omegas which at 98db and 8Ohms impedance, will be a piece of cake for Dave, so I expect that I will get better dynamics and effortless delivery of music. Given what I am hearing with the D7s now, I am not sure that the Omegas will be better in all aspects.

 

Finally, big thanks to all the contributors to this thread and especially @Romaz and @ElviaCaprice for their advice and helpful tips.

 

To be continued!

 

 

Hey Mozes you are one courageous guy to completely redo your whole system. Kodos I'm glad it worked out for you.

Share this post


Link to post
Share on other sites
2 hours ago, romaz said:

As to the impact of these clocks to signal timing, I don't believe it has anything to do with it.  Referring back to John's Swenson's post on the REF10 thread that Rajiv provided a link to, the function of the clock in devices like the sMS-200, microRendu, Iso-Regen and tX-USB is not to time the signal (which is what a word clock is responsible for) but for "processing" the signal. How I interpret this is these clocks are necessary for the "functioning" of these components and so a good clock allows a component to function better.  I'll use my production line analogy once again.  When a production line operates smoothly and in a timely manner, less mistakes are made and less time and energy are wasted to correct any mistakes.  This would be similar to how Audiophile Optimizer improves SQ, by removing unnecessary background processes, you get fewer software errors and fewer latencies which also translates to less current draw and less noise being generated in the ground plane.

 

This is the area where even respected DAC designers have not bought in to what we have already observed empirically. You previously mentioned Rob Watts' skepticism. Over on the QX-5 thread too, Charles Hansen has described the commendable care and effort they've taken with low-phase-noise clocks for the DAC. But I have a pending question to him whether they applied the same care and low-phase-noise clocking to the "function" or "system" clocks, which also reside in the QX-5, for USB, Ethernet, and the ARM processor to run the Roon Ready module.

Share this post


Link to post
Share on other sites
4 hours ago, mozes said:

System update: Radical change!

 

Well, I think it's about time to share an update about my system although it is still not where I want to end up with. Anyhow, I guess it is always WIP as usual.

 

Over the last few months, I have read various posts by @Romaz and @ElviaCaprice regarding their minimalist approach to their audio chain, that is going straight from DAC to speakers and the advantages this strategy yields. I became very intrigued given the isolation from AC mains, reduction of box count and increased transparency.

Of course, not any dac or speaker will work, so I had to make some tough decisions, and in the end I made up my mind. I sold my Naim gear, Brooklyn Dac and its VR Mini PSU, and replaced them with Chord Dave. Second, I bought two sets of speakers, Spendor D7 and Omega Super 8XRS with silver/gold wiring and high end Furutech binding posts. Why did I buy the Spendors and they are 90db only at 8 Ohms impedance? simply I fell in love with these speakers. They are one of the best speakers that engaged me regardless of price and I couldn't resist pulling the trigger. It seems, I have a weakness to everything made in the UK, having owned Naim, Neat, Chord and Spendor and who knows what more.

 

In the meantime, I also hunted a second hand tX USB ultra and I sent both tXs to SOTM who upgraded the DC wiring to silver, added Master clock inputs and modded a D-link switch from one of the tX units. I also got a 1m Lush USB cable that I added to my chain. To cut the story short, I am not going to talk about the many many tests I did and all the combinations I tried. I will simply share what my ultimate chain is so far which gave me the best sound ever till now :)

 

Nimitra (VR mini)>USPCB>Iso-Regen(LPS-1)>Elijah Audio USB cable (signal and ground only)>tX-USBultra(LPS-1)>USPCB>tX-USBultra(LPS-1)>1m Lush USB cable>Chord Dave>Spendor D7

 

Grounding: My Nimitra and Dave are grounded to Entreq Olympus Minimus and D-Link switch is grounded to Silver Minimus. Each of the D7 speakers have its negative speaker binding post grounded to an Aucharm Box.

 

This setup simply blows away anything I have had before in my system and I have tried real high end Naim gear (up to 500 series level). What I am getting is transparency and resolution like crazy! Music sounds just real, I have no better description.

 

The crown jewel of my system is undoubtedly the Dave, this little box amazes me. It is my dac, preamp and power amp and it gloriously drives 90db speakers to levels beyond my tolerance level and there is no need for any subwoofer. How can this little beats do that? only 2 watts!! I guess the answer is as @Romaz highlighted before, its ultra low impedance of only 0.055 ohms. I also bought a Rogue Cronus Magnum 2 tube amp to see what tubes offer. I tried Dave with the Rogue and it sounds very very sweet, but I prefer Dave directly to the D7s, I simply didn't want to compromise on transparency and resolution. With Dave driving the D7s directly, I put my ears on the tweeters and there is no noise at all, I mean literally ZERO noise, this is crazy, I haven't had any amplifier that it is DEAD quiet. This explains why I am enjoying now low level late night listening, as I feel that all the details are there, hard to describe what I am hearing, it is pure enjoyment.

 

My Tidal now sounds way better than my locally stored music in my previous system. You won't believe that this is streaming, it is like high res playing. The modded switch with Dave takes Tidal to a completely different level.

For now, I am waiting for the delivery of my Omegas which at 98db and 8Ohms impedance, will be a piece of cake for Dave, so I expect that I will get better dynamics and effortless delivery of music. Given what I am hearing with the D7s now, I am not sure that the Omegas will be better in all aspects.

 

Finally, big thanks to all the contributors to this thread and especially @Romaz and @ElviaCaprice for their advice and helpful tips.

 

To be continued!

 

 

 

Mozes,

 

VERY interesting read. Thanks for sharing... and you've got me thinking furiously!

 

For those considering this, IF you don't have efficient speakers (like me... the various Eminent Technology LFT planar speakers I have here range from 81-84 dB 1wt/1m), there are 'booster' amps designed to provide more current to a load without imposing much of their own sonic signature. I believe 1 or some of Nelson Pass' First Watt series does this... and Paul Hynes published a design for the DIY'er he uses over on Audio  Circle a few years back. IF the source does not put out a high enough voltage, a well-designed tube or FET buffer can increase the final voltage delivered into the current buffer amplifier and the total wattage delivered.

 

That MIGHT be my final solution!

 

Greg in Mississippi

 

P.S. Paul's posts on his amps are here:

 

http://www.audiocircle.com/index.php?topic=66518.msg612458#msg612458

 

http://www.audiocircle.com/index.php?topic=86222.msg841368#msg841368

Share this post


Link to post
Share on other sites
3 minutes ago, gstew said:

 

Mozes,

 

VERY interesting read. Thanks for sharing... and you've got me thinking furiously!

 

For those considering this, IF you don't have efficient speakers (like me... the various Eminent Technology LFT planar speakers I have here range from 81-84 dB 1wt/1m), there are 'booster' amps designed to provide more current to a load without imposing much of their own sonic signature. I believe 1 or some of Nelson Pass' First Watt series does this... and Paul Hynes published a design for the DIY'er he uses over on Audio  Circle a few years back. IF the source does not put out a high enough voltage, a well-designed tube or FET buffer can increase the final voltage delivered into the current buffer amplifier and the total wattage delivered.

 

That MIGHT be my final solution!

 

Greg in Mississippi

Thanks Greg for sharing, very insightful 

Share this post


Link to post
Share on other sites
18 hours ago, ElviaCaprice said:

I gave SOtM liberty to use the remaining 3 clocks available on the NUC, 1 went to the txbexp pcie card.  So that is exactly what they did.  I did specify for them not to replace any USB or Ethernet port clocks, since I would be using the PCIE card to stream from.  I know they replaced the system clock but the other two??   Unfortunately I won't be able to use my new PH SR7 MR4 until next summer, stuck in Minnesota and just too expensive to ship down here to Costa Rica, never know what customs might charge.

The SCLK-EX board is free standing.

 

Here will be the chain I shall use for now, the PH SR7 will replace any temporary 12V inputs, there are 3.

JETWAY NUC JBC311U93-2930-B   (modified with 3 SCLK-EX clocks) (added mini pcie to PCIE 2X adapter) (txbexp pcie usb card modded with 1 SCLK-EX clock) => (USPCB) Chord 2 Qute => Omega Super 8XRS

 

For power, the NUC shall be powered via LIPO battery (I have two powerful 16AH ones here in CR for my fishfinder)

SCLK-EX and Chord 2Qute shall be powerd by 2 LPS-1's in a series 12V.

The Jetway NUC luckily has a SATA II port, not shown in specs.  So I shall power a 5V 2.5" HDD with an LPS-1 for data.  The txbexp card shall be powered by another LPS-1, 7V.

 

The addition of the SCLK-EX board and modifications wasn't cheap, I also added the master clock input.  All said and done it was just under $1300.  I only paid $100 for the NUC used and $150 for the txbexp used. 

 

I will test with and without the ISO Regen.  But from Roy's findings I expect the ISO Regen may be a burden to SQ.

 

I am going to be very curious to hear how your modded NUC sounds. I am looking for something like this - a cheap NUC, that can be sCLK modded to be an economical Roon server, while retaining the rest of the endpoint-based chain.

Share this post


Link to post
Share on other sites
50 minutes ago, austinpop said:

 

I am going to be very curious to hear how your modded NUC sounds. I am looking for something like this - a cheap NUC, that can be sCLK modded to be an economical Roon server, while retaining the rest of the endpoint-based chain.

Yes, with this NUC, which has two Ethernet ports, it would be fine for an added renderer chain.  In fact, one of the EX clocks could have been used for the Ethernet ports, if one wanted.   I bought mine on Ebay.  I wasn't planning originally to use this model but another Pico-ITX Jetway.  But the Pico-ITX didn't have a SATA port, so I opted for the NUC.  Also JRiver 64 bit can now download a whole album to memory, so having 4GB is better, especially for DSD.  The Pico-ITX only had 2GB.

Share this post


Link to post
Share on other sites
7 hours ago, romaz said:

Sorry for my delay in posting.  Hopefully, I can post my more definite findings soon.

 

Just to add to the discussion on clocks, it is unclear what clock frequency Ayre uses in its AX-5 Twenty but I suspect it won't be a 10MHz clock and you cannot compare phase noise measurements between clocks if they operate at different frequencies.  This is according to Chris Peters, CEO of Mutec.

 

When I asked him to compare for me the differences between the clock used in their well-regarded MC-3+USB vs their REF10, here is what he had to say:

 

"The problem is that comparing the output signals of the REF10 and MC-3+USB is nearly impossible because of the different clock frequencies of both devices. The phase noise depends directly on the clock frequencies. We have measured the MC-3+USB with the nearest clock frequency of 11.2896 MHz to the REF10. The difference is approx. 30 dBc. That means the phase noise of the REF10 is approx. 30 dBc lower as this one of the MC-3+USB."

 

When I asked him why I have been underwhelmed by my experience with certain "atomic clocks" in the past, here is what else he had to say about the REF10 and atomic clocks:

 

"The phase noise difference e.g. between the REF10 and the Antelope 10M or the 10MX is approx. 40 dBc. That is the reason why people do almost not hear any difference when connecting the 10M to the MC-3+USB. The MC-3+USB is better the so-called Atomic Clock of Antelope."

 

Now is a difference of approx 30dBc audible?  Based on my experience in Munich, very much so and enough of a difference that I went ahead and bought the REF10.  

 

Here are the phase noise tracings for Mutec's REF10:

 

59c7e68f07ca7_MutecREF10phasenoiseplot.thumb.jpg.184ba508433d6876fcf30f20f19240c9.jpg

So, how do you interpret this plot?  Not so easy, it seems.  There is a knowledgeable and well-respected poster here on CA (who I shall not name) who comments on the importance of phase noise measurements below 1Hz.  Well, when I showed these plot tracings to Lee in Munich, he told me the important measurements to look at are between 10Hz and 100Hz and it is between these frequencies that a really good clock will separate itself from other clocks.  In this regard, he felt the REF10 was stellar, perhaps the best he had ever seen and so it was clear to me that this is the frequency range he would be targeting with SOtM's master clock.  Here is what Chris Peters had to say about this (I have intentionally X'd out the name of the CA poster in question):

 

"I assume you are referring to the discussion in the CA forum regarding the audio-relevant measurement areas. And I think you are referring to "XXX" when talking about the range lower than 1 Hz. I honestly speaking do not know where or how "XXX" got this information. My beta testers have started trying to experience for audio most relevant frequency range since 2013 when we released the first version of MC-3+. So far we experienced audible differences between various oscillators when their specifications where different in a range between 1 Hz and 100 Hz distance from the carrier frequency of 10 MHz. Differences in this frequency range between the oscillators were audible best for my beta testers. So we optimized the REF10 for this frequency range specifically."

 

So, to a large extent, Chris Peters is agreeing with Lee of SOtM.  When looking at phase noise measurements of a clock, look at the measurements between 1Hz - 100Hz.  What is interesting is that with better, clocks, you definitely hear an improvement in bass definition but improvements are also clearly heard in the midrange and treble.  The best explanation I have for this is that if you improve the signal at 20Hz, for example, you would likely also improve its harmonic frequencies as well (ie 20Hz, 40Hz, 60Hz, etc).  As an example, I am currently testing Synergistic Research's "Black Box."  This is a $2k device that I had once assumed was just a glorified overpriced bass trap but it's impact is so much more.  As I am presently testing potential speakers in my large listening room with cathedral ceilings, while my room is not a resonant nightmare, there are clearly nodes in my room resulting in boomy bass in different areas.  With the introduction of this small box (which contains specially tuned passive resonators) into my room, not only does the bass boom disappear but midrange and treble clarity are greatly enhanced.  This Black Box is definitely staying put.

 

Anyway, back to clocking, as you further assess phase noise measurements, remember that with the REF10, phase noise measurements are taken from the BNC outputs and not from the clock.  Taken from the clock, the measurements would probably be even better.  I suspect any clock measurements Ayre might report will be provided by the manufacturer meaning the phase noise at the Ayre's outputs will most certainly be worse.  It remains unclear what SOtM's new clock measurements are based from although I have posed this question to Lee himself.

 

As a further example of how important this is, I have been testing clock cables of various price points and length.  Using various inexpensive DigiKey clock cables from the same manufacturer of various lengths, as you go from 20 to 40 cm in cable length, the SQ degradation is clearly audible.  This is why I had to send my gear back to SOtM.  Because they didn't have the really short clock cables in stock, they ended up using much longer clock cables in my build which I ultimately deemed as unacceptable.  Moreover, as I have tested identical length clock cables with my REF10 from companies like Pasternack ($40), Blue Jeans Cables (<$20), and Black Cat ($250) against the 700 Euro Habst clock cables that I purchased with my REF10, unfortunately, the differences are quite significant with respect to HF harshness and a very flat sound.  Not that the cheap cables sound horrible but when you replace them with the Habst, there's simply no wanting to go back to those cheap cables.  This is where those external clock doubters have a leg to stand on when they make their claims that external clocks don't add anything.  Cable length and cable quality DEFINITELY matters.

 

As to the impact of these clocks to signal timing, I don't believe it has anything to do with it.  Referring back to John's Swenson's post on the REF10 thread that Rajiv provided a link to, the function of the clock in devices like the sMS-200, microRendu, Iso-Regen and tX-USB is not to time the signal (which is what a word clock is responsible for) but for "processing" the signal. How I interpret this is these clocks are necessary for the "functioning" of these components and so a good clock allows a component to function better.  I'll use my production line analogy once again.  When a production line operates smoothly and in a timely manner, less mistakes are made and less time and energy are wasted to correct any mistakes.  This would be similar to how Audiophile Optimizer improves SQ, by removing unnecessary background processes, you get fewer software errors and fewer latencies which also translates to less current draw and less noise being generated in the ground plane.

 

Now, does a better clock guarantee better SQ?  Not always and if there is SQ improvement, this improvement can be variable.  For example, having replaced the stock Crystek clock in the ISO-Regen with the clock in the REF10, is it now on equal footing as the tX-USBultra (which is now also being clocked by the REF10)?  The simple answer is no.  While the Iso-Regen definitely is improved on the REF10, my tX-USBultra (which is also being clocked by the REF10) is still the better component to my ears.  In essence, and I have suggested this before, the best that a clock can do is to allow a component to perform at its very best but a good clock can never make a component perform beyond its physical capabilities.  Absolutely, it is the circuit as a whole that matters and the clock plays just a small part of it.

 

To elaborate on this topic further, while a good clock improves how a component operates, clocks contribute more than just timely functioning, they also contribute noise which can be measured in dB.  Since there is no such thing as a perfect clock, all clocks contribute some degree of noise, it's just a matter of how much.  When one bad clock in your digital pathway can contribute more than 30dB more noise compared to the REF10, imagine the cumulative noise impact of 8 noisy clocks in that same pathway.  To some extent, you can mitigate that noise by throwing an sMS-200ultra into the pathway between your server and your DAC but as many of us have experienced, this doesn't completely fix the problem since adding a reclocking switch before the sMS-200ultra and adding a tX-USBultra after it results in further improvement.

 

As I have now replaced all the clocks in my digital upstream from router onward, it is amazing how each clock replacement adds improvement but what I have also found is that the subsequent clock replacements seem to have less impact than before.  What I am trying to say is that with the clocks replaced in my server, the impact of the ISO-Regen and tX-USBultra is now considerably less.  

 

I know there are people out there who are critical of the "spaghetti solution" that we call the SOtM trifecta.  As an owner and originator of this trifecta, I have to agree it is a bit unsightly and cumbersome, especially when you incorporate the very inflexible SOtM dCBL-CAT7 into the mix (which I find to be indispensable).  This is one reason I went away from it but what I can say is that it doesn't have to be this way for those who are opposed to it.  With a low noise server (meaning a low power CPU, minimal RAM, avoidance of noisy storage drives, replacement of noisy clocks and driven by a clean, low-impedance PSU), this "spaghetti solution" is not only no longer necessary, but it is actually an inferior approach.  Not to suggest that those that have this spaghetti solution should move away from it (because this spaghetti solution probably still sounds better than most things out there) but that there are other pathways to achieving similar (and better) results.

 

Once again, while in Munich, Lee himself told me he considered the sMS-200ultra and tX-USBultra as his mid-level products.  He reserved his highest praise for his sMS-1000SQ server and the Ultra version incorporates the sCLK-EX and also his very best card, the tX-USBexp.  With this server, you can request to have the system board, Ethernet port and the tX-USBexp reclocked and everything fits nicely in one chassis.  Where the sMS-1000SQ probably can be improved (based on what people have told me) is with its PSU.  For those looking for an elegant, no-fuss, turnkey solution, I would suggest the sMS-1000SQ Ultra paired with a multi-rail SR7 and either SOtM's new master clock or the REF10.  Can this be improved upon?  Yes, I know it can and I will provide details of this in a further post.

 

 

Thank you for sharing your finding, it is very eye opening. I am particularly very interested with this quote "With a low noise server (meaning a low power CPU, minimal RAM, avoidance of noisy storage drives, replacement of noisy clocks and driven by a clean, low-impedance PSU)". As I just got myself an asrock j3455b-itx mobo recently to use as a music server. It is 1.5Ghz CPU and I like it's atx connector type power supply rather than DC supply, with compare with my previous Atom mobo.

 

So I don't expect it to be a world beater as I am very green in this exploration, and I would be grateful if someone can point me the link to more definitive thread on what it means by low power CPU, min ram etc.

Share this post


Link to post
Share on other sites
1 hour ago, lateboomer said:

Thank you for sharing your finding, it is very eye opening. I am particularly very interested with this quote "With a low noise server (meaning a low power CPU, minimal RAM, avoidance of noisy storage drives, replacement of noisy clocks and driven by a clean, low-impedance PSU)". As I just got myself an asrock j3455b-itx mobo recently to use as a music server. It is 1.5Ghz CPU and I like it's atx connector type power supply rather than DC supply, with compare with my previous Atom mobo.

 

So I don't expect it to be a world beater as I am very green in this exploration, and I would be grateful if someone can point me the link to more definitive thread on what it means by low power CPU, min ram etc.

Nothing wrong with that mobo. it's an embedded one with fairly low power needs.  How do you plan to power it, since it's ATX?  Are you going to stream via USB?  What are you going to add to the PCIe lane?  How much memory? 

Share this post


Link to post
Share on other sites

I am glad you ask. I am using coolmaster 80 plus 400w normal power supply for it. I will consider seasonic 750w prime titanium suggested by Lmitche in the future as I maybe consider migrate to Asus z170 plus mobo with i7-7700 to enable up sample to dsd256. Currently Asrock can only do dsd128 running under Daphile server client setup.

 

Currently, the client is nuc 3815 powered by diy regenerative and lps. I don't think onboard DC supply is better than atx power supply. However, lps for atx  power supply is too challenging to implement. That is why I would consider Seasonic titanium. 

 

My config now:

Minipro ext HDD with Jitterbug >> asrock j3455b >> Ethernet >> drink dir-850l router with lps >> fmc and sfp fiber with lps for both FMC >> nuc 3815 with lps >> Regen Amber with lps >> USB DAC

 

For pcie lane, I closely follow just launched jcat net femto card. If I use Asus z170 as server, then I could swap this Asrock as client and consider jcat USB femto card.

 

But with FMC fiber implementation, I not sure do I still need to consider those femto cards or not.

 

One thing, current memory is 4GB x 2, I thought dual memory can help me to get to dsd256, it is just wishful thinking.

Share this post


Link to post
Share on other sites

Share this post


Link to post
Share on other sites

Quick question for @romaz since you're receiving your Hugo 2 recently / very soon, did you get a chance to ask SOtM anything about their willingness to modify the clocks of Hugo 2 for loyal customers like you?

 

DAVINA with both USB input and output should be on the horizon, that means we'll be able to capture the upscaled files of M-Scaler and save those 705.6kHz / 768 kHz wave files for offline playback afterwards.

 

In other words, we're still taking advantage of M-Scaler's power while the path should look like:

 

⇒ tX-USBexp+sCLK-EX combo

⇒⇒ tX-USBultra

⇒⇒⇒ Hugo 2

 

Another member (Bamber) was paying for multiple sCLK-EX boards and that's why SOtM was happy to make an exception to the rule

 

https://www.computeraudiophile.com/forums/topic/30376-a-novel-way-to-massively-improve-the-sq-of-computer-audio-streaming/?page=109&tab=comments#comment-691317

 

Share this post


Link to post
Share on other sites
5 hours ago, lateboomer said:

I read that jcat femto net card is developed from Intel i350t nic card. 

 

I wonder we could hack the Intel nic card to provide lps to it.

 

It uses i350 ethernet controller, but it's a proprietary design, not a modified intel card. 

 

Best regards,

Marcin

Share this post


Link to post
Share on other sites
17 hours ago, romaz said:

 

As a further example of how important this is, I have been testing clock cables of various price points and length.  Using various inexpensive DigiKey clock cables from the same manufacturer of various lengths, as you go from 20 to 40 cm in cable length, the SQ degradation is clearly audible.  This is why I had to send my gear back to SOtM.  Because they didn't have the really short clock cables in stock, they ended up using much longer clock cables in my build which I ultimately deemed as unacceptable.  Moreover, as I have tested identical length clock cables with my REF10 from companies like Pasternack ($40), Blue Jeans Cables (<$20), and Black Cat ($250) against the 700 Euro Habst clock cables that I purchased with my REF10, unfortunately, the differences are quite significant with respect to HF harshness and a very flat sound.  Not that the cheap cables sound horrible but when you replace them with the Habst, there's simply no wanting to go back to those cheap cables.  This is where those external clock doubters have a leg to stand on when they make their claims that external clocks don't add anything.  Cable length and cable quality DEFINITELY matters.

 

3

Hi Romaz,
You have found that shorter cables improve the sound quality. I wonder what length is your Habst. The shortest on their site is 50cm but for me even 30 or 25 would do. Do you think it is worth trying to order that length? 

Also I wonder if placing the MC3+USB on the Ref 10 would have any bad effects. That would give the chance to use the shortest cable. 

Thanks,

Zoltan 

Share this post


Link to post
Share on other sites
3 hours ago, romaz said:

Rob Watts will be introducing amplifiers that will connect to his DACs via digital interconnects (not analog ones) and will have the same resolution and transparency characteristics as DAVE directly driving speakers.  Essentially, these amplifiers will be "invisible" meaning they will have no character of their own.

This is very interesting stuff, a radical shift from all amps in the market! I hope Chord launches this new amp at CES 2018

Share this post


Link to post
Share on other sites
4 hours ago, romaz said:

 

Much has been said about how differences among servers are perhaps greater than difference among DACs.  In my own experience, I both agree and disagree with this observation and I will attempt to explain.  In my view, the DAC is clearly the more important component and why some DACs sell for >$100k.  Because this thread was never meant to discuss DACs, I have shied away from commenting but given Moussa's post, I figured I should comment although this will represent my last post on DACs on this thread.  In fact, this post will represent the beginning of my exit from posting in forums in general.  Life has become too busy.

 

When putting together an audio system, people will have their priorities.  I have already stated mine and they are simply (1) resolution and (2) transparency.  My reference isn't vinyl or tape or the million dollar setups one can hear at RMAF, Axpona or Munich, my reference is the live music I am often exposed to.  Most of what I listen to is unamplified acoustical music, whether it be large orchestral, small ensemble, choral music, or solo instrumental (especially organ but also piano and guitar).  When I am at an acoustical performance, whether it be classical or jazz, the first thing I notice is the acoustics of a venue and the resonances that venue provides.  The natural reverb and decay of instruments and voices are quite evident and from the stalls to the balcony or from one venue to another, they will vary.  

 

It has been stated that the reverberation time in a large venue like Carnegie Hall measures between 1.8 to 2 seconds.  At the Alice Tully Hall in the Lincoln Center just a mile away, this more intimate arena has a shorter reverberation time of 1.4-1.5 seconds.  Which is preferable depends on whether I am listening to a solo guitar, four string quartet or a full orchestra but regardless, I very much enjoy hearing the acoustics of a great building and never would I prefer to hear music in an anechoic chamber.  This is where most DACs stumble and where I find the DAVE excels.  This is also where I find PCM superior to DSD.  DSD provides you an expansive and a soft "tube-like" sound but this softness, which can be a wonderful way of masking the harshness of many chip DACs also results in a diffuse and imprecise presentation with respect to depth and timing and my careful A/B of my own recordings has convinced me of this.  As someone who values the accurate spatial portrayal of a live musical performance, I have found that a good music server can provide much but a good DAC can provide more.

 

When talking about resolution, as we look at our PCM files, we are provided 2 types of information:  (1) bit-depth and (2) sampling rate.  For Redbook, this means 16/44 which translates to 16-bits of dynamic range and a sampling rate of 44 kHz.  While DR is important, I contend that sampling rate is much more important with respect to a DAC's abilities.

 

When people talk about dynamic range, most people think about how loud and dynamically a DAC can play when really, it's about how quietly a DAC can perform that is important.  With regards to DAC performance, Rob Watts equates DR to the "hiss level" of the DAC and the greater the DR, the less likely you are going to hear "hiss" when no music is playing.  There is a DAC (that I will not name) that sells for >$100k and boasts a DR of 173dB (or 28.8 bits of dynamic resolution) as if we should be impressed by this.  For those that know better, this performance metric is useless since most believe most humans are incapable of hearing beyond 21 bits of dynamic resolution.  Just as important, most ADCs are also limited to about 21-bits of DR and so when people talk about 24-bit recordings, they often don't contain a true 24-bits of dynamic range.  Even at 24-bits (or 144dB) of dynamic resolution, for those who choose to look at DR in the traditional way of how loudly something can play, listening to any sound at 144dB SPL would be considered lethal. Now this is what people fail to realize -- as soon as you connect DAVE (or any DAC) to an outboard headphone or speaker amp, you now have thrown away the DR capabilities of your DAC because now, you've buried the DR performance of your DAC into the much higher noise floor of your amplifier.  For those who use an outboard amplifier with their headphones or speakers (this means most everybody who do not own a Chord DAC), you're basically listening to the much more limited dynamic range of your amplifier which is typically between 16-18-bits.  

 

With regards to sampling rate, I will explain why I consider this to be the more important spec with regards to DACs and this is why most DACs cannot match the performance of the very best turntables.  Sampling rate gives you a measure of timing resolution and this provides you not just spatial information such as depth but also timbre accuracy and the layering of fine detail.  With analog sources, you are hearing a continuous waveform and SQ is limited only by the quality of the gear that transmits this waveform.  As such, it is generally easier to get great sound from an analog setup such as a turntable.  With digital, an ADC is responsible for sampling the analog waveform a specific number of times per second and the larger the number of samples that are taken, the fewer the gaps of missing information there are and the more fluid or "analog" the recording sounds.  In theory, a waveform that is sampled 176,000 times per second (hi-res PCM) will sound better than a waveform sampled only 44,000 times per second (Redbook).  If that waveform is sampled an infinite number of times, then from a mathematical standpoint, your digital file becomes equivalent to your original analog waveform but as we know, infinite sampling is not possible based on the technology we have today and so this would suggest that digital can never truly equal analog.

 

However, there is the practical matter of the limitations of human hearing that potentially make it possible for digital to equal analog.  Most scientists agree that the human brain/ear has the ability to discern 2 separate sounds if they occur at least 5-7µs (microseconds) apart and so this represents the limits of a human's auditory time resolution abilities.  This means that when 2 sounds occur 10µs apart, as an example, we can hear 2 discrete sounds but when these 2 sounds only occur 4µs apart, instead of hearing 2 discrete sounds, we hear only one blended sound.  This is the rationale for why digital sounds "discrete" and why analog sounds "continuous."  With Redbook, as previously stated, sampling occurs 44,000 times per second and this equates to a time resolution of 20.8µs.  Anyone comparing a CD to vinyl in a resolving setup should easily be able to discern that with a CD, information is clearly missing.  As you sample more often, let's say 96,000 times per second, time resolution improves to 10.4µs and while this represents a significant improvement, most ears will likely still be able to detect that an analog source provides more information.  When you use an ADC to sample a file 192,000 times per second, time resolution now improves to 5.2µs.  In theory, at this sampling rate, a digital file should sound virtually indistinguishable from the original analog wave form and so this is the basis for why hi-res files were created.  This would suggest a 24/192 hi-res PCM file should sound equivalent to the original analog waveform.

 

For those who have done careful listening, however, with most DACs, 24/192 does not equal analog and even DXD or DSD256 files still can't match the resolution of the very best analog setups.  At most audio shows you attend, when you ask a certain exhibitor to give you their very best presentation, if they have a turntable or a reel-to-reel present, quite often they will switch to their analog source and, in fact, I have witnessed this many times.  As a further example, having visited the Magico factory in Hayward, CA recently, they have arguably the finest listening room assembled in the world today.  This room cost them $250k to build and has the equivalent of a floating floor and no parallel walls to avoid standing waves.  Short of an anechoic chamber, it perhaps has the lowest noise floor of any listening room and they use this room as their lab.  In fact, it is how they voice their speakers including their $600k Magico Ultimates and their soon to be released $175k M6.  Here is a photo of that room:

 

59c82952949e6_Magicolisteningroom.thumb.jpg.a471bfd9423c20e2b246c5ff6099f573.jpg

 

Because Berkeley DACs are the local favorite, they use a Berkeley Reference 2 DAC (Berkeley is headquartered nearby) fronted by a Baetis Reference server.  However, when they wish to present their very best, they revert to their turntable.

 

The reason is not so much because this sampling theory is faulty but because ADCs have limitations.  It is the reason why such technologies like MQA were created and why many DACs oversample.  Those in the NOS (non-oversampling) camp suggest that NOS DACs sound more natural but NOS strives only to reproduce the best that the ADCs can offer, warts and all.  Oversampling is much more ambitious and strives to overcome the limitations of the ADC by interpolating the missing bits of information through the use of sophisticated mathematic filters.  If the oversampling is done perfectly, a 16-bit Redbook file originally sampled at 44kHz per second should be audibly indistinguishable from the original analog waveform and this is the basis for the long tap-length filters that Rob Watts has been championing for decades but also the basis for what HQPlayer tries to accomplish.  As to who does it better, I will leave it for others to decide for themselves but having listened to both approaches, I much prefer Rob's approach.  As to the benefits of oversampling to DSD vs PCM, people will have their preferences, I have already stated mine.  

 

Regarding why some people fail to recognize great differences between DACs, I hear this all the time and I believe there are several reasons.  As both a headphone and a speaker listener, I have found both types of listening to have their advantages.  Headphones have the ability to portray fine detail better while speakers can image and soundstage better.  

 

DAVE is unique because its headphone output doesn't utilize a separate headphone amp.  When you plug a headphone into DAVE's headphone jack, you are actually listening to the DAC itself.  This means your headphone is tapped to DAVE's full bandwidth, ultra low noise floor (-180dB), dynamic range, and time resolution.  Moreover, what is unique about DAVE is it has no noise floor modulation and so whether you listen to music at low levels or at DAVE's peak levels, noise floor remains at the same ultra low levels.  There is simply no cleaner, clearer, more transparent way of listening to music than this.  The problem with headphone listening is that headphones do not portray depth well, certainly not as well as speakers and so to this degree, a lot of DAVE's performance cannot be fully realized through headphones alone.

 

The problem with listening to speakers with DAVE (or any DAC) is that DAVE's performance is largely buried in the amplifier you use to drive your speakers.  While DAVE's performance still shines through, its performance is blunted as you end up inheriting many of the limitations of even the finest speaker amplifiers.   Just like with outboard headphone amps, no speaker amp can match the performance characteristics of your DAC and so what you get with even the finest amps is a diminished photocopy of the original.  

 

Throw in a preamp, no matter how good, and this further adds to a loss of transparency.  That is just the nature of adding components to your analog chain.  Unless you are using a preamp for sound tuning (ie tube linestages), or you have an amp that demands a certain preamp to function optimally or unless you have multiple sources you need to switch among including a turntable, with DAVE, the very best preamp is no preamp at all.  Just like with amplifiers, no preamp can match DAVE's performance with respect to distortion characteristics, noise floor, speed, dynamics, or time resolution.  Even more, Rob programmed into DAVE the ability to attenuate down to whisper levels with absolutely no loss in resolution.  That means that as you attenuate DAVE to its lowest level (-75dB), DAVE is still outputting full resolution, something that no preamp can match.  

 

In the photo below is VAC's very highly regarded Master preamp (about $30k):

 

59c8b1c00f056_VACpreamp.thumb.jpg.375cee006d04903d4d7b282706624c0d.jpg

 

Kevin Hayes, VAC's designer, was kind enough to allow me to compare my DAVE driving his wonderful VAC tube amplifiers both with and without his Master preamp:

 

59c8b207a8935_VACpreampwithDAVE.thumb.jpg.4f43aeaa034e0bfaa90fe35fd4059bc3.jpg

 

It was the forgone conclusion of most people in the room that the sound through the attached Harbeth speakers would be vastly better with the Master preamp in the chain.  They were surprised when this was not the case.

 

Here is another example of a dealer's DAVE driving an $11k Constellation Inspiration Stereo Amplifier both with and without Constellation's $9k preamp.  To both the dealer's and my ears, SQ was better without the preamp and so when this dealer sells a DAVE, he no longer tries to promote the sale of a preamp:

 

59c8b3c8e95ff_DAVEconstellation.thumb.jpg.58e4b52d266ff41781b465b735bebfa8.jpg

 

And so what Moussa and ElviaCaprice are hearing is something that is very unique.  Through their high-efficiency speakers, they are hearing the full potential of their Chord DACs limited only by their choice of cabling and speakers.  With either the Omegas or the Voxativs I am using, I am hearing every bit of detail that my best headphones can provide while also the imaging and soundstage that only speakers can provide without the resolution and transparency robbing  impact of an outboard preamp or amplifier.  At the present time, I am trying out a pair of $25k Martin Logan Renaisssance Hybrid Electrostatic speakers in my large listening room, which I find to be very resolving and transparent.  These speakers are currently being driven by a pair of Pass Labs XA60.8 class A monoblocks ($13.5k for the pair).  While I cannot deny how wonderful this sounds when fronted by my DAVE, compared to DAVE directly driving my more modest pair of Omegas, this latter setup still sounds more resolute and more transparent.  This is possible only with Chord DACs because only Chord DACs (as far as I'm aware) have output impedances that are low enough to directly drive speakers.  In the case of DAVE, which has an output impedance of 0.055 ohms, this equates to a damping factor of 145, which is stellar.  Soon, Rob Watts will be introducing amplifiers that will connect to his DACs via digital interconnects (not analog ones) and will have the same resolution and transparency characteristics as DAVE directly driving speakers.  Essentially, these amplifiers will be "invisible" meaning they will have no character of their own.  They will have class A output and the first amplifiers will output either 20 watts stereo or 70 watts in monoblock form.  This technology is supposed to be scalable where 200 watts of amplification will be possible.  

 

Furthermore, as I have alluded in other posts, I have added Rob's new M-scaler to my DAVE.  This is incorporated into Chord''s new Blu Mk 2, which is a CD transport that also includes a USB and BNC SPDIF input.  This increases DAVE's TAP resolution to just over a million TAPS.  This is a milestone that suggests Redbook is now completely indistinguishable from the original analog waveform and Rob didn't believe it would ever be achieved when he first conceptualized it back in the 80s but because of the rapid advancement of FPGA technology, this indeed has been achieved.  Practically speaking, this results in a massive improvement in DAVE's resolution, so massive that the collective impact of my server mods which includes 8 clocks being replaced pales in comparison to what Blu Mk2 provides.  For those of you who own a Chord DAVE, I would suggest you prioritize getting a Blu Mk2 beyond anything else discussed on this thread.  Combined with Chord's upcoming "digital" amplifiers, there will be no more resolute or transparent way of listening to a digital file.  Despite all of this, I am finding, however, that the quality of the music server still matters.

 

Truly an epic post!   Thanks so much for taking the time to compose it.

Share this post


Link to post
Share on other sites
1 hour ago, mozes said:

This is very interesting stuff, a radical shift from all amps in the market! I hope Chord launches this new amp at CES 2018

This has been done before by Lyngdorf. Look at the TDAI2170 and the SDA2400 and even before that. Truly amazing gear. These are true digital amplifiers without the drawback of other so-called class D amplifiers. No coloration, at all and with built-in room correction. 

Share this post


Link to post
Share on other sites
4 hours ago, romaz said:

 

When putting together an audio system, people will have their priorities.  I have already stated mine and they are simply (1) resolution and (2) transparency.  My reference isn't vinyl or tape or the million dollar setups one can hear at RMAF, Axpona or Munich, my reference is the live music I am often exposed to.  Most of what I listen to is unamplified acoustical music, whether it be large orchestral, small ensemble, choral music, or solo instrumental (especially organ but also piano and guitar).

 

I understand the focus on resolution and transparency as I've been down the same path.  I suspect your objective has been to eliminate all sources of color which could be introduced by components.  That is to say, not in the recording itself.  Whether that be through noise of any kind, tubes, cable characteristics, etc you're looking for the most neutral sound possible from your auditioned components.  You're also focusing on acoustic music.  Does that mean you're not using any electric instrument recordings in your tests, and that your test recording tracks are a sampling of acoustic music?  Those who listen to vinyl or electric amplified music, or those who target characteristics like musicality or PRAT may be after different results.  That's not to say you aren't after those as well, just that they aren't as high on your list as resolution and transparency.

 

4 hours ago, romaz said:

  For those who use an outboard amplifier with their headphones or speakers (this means most everybody who do not own a Chord DAC), you're basically listening to the much more limited dynamic range of your amplifier which is typically between 16-18-bits.  

 

Are you saying all Chord DACs are unique in this regard because as far as you know no other can drive speakers without an amp?  The amp I use (Benchmark AHB2) is rated at 135db.  Given the DAVE is rated at 127.5db why would I be limiting the dynamic range by using this amplifier?

 

4 hours ago, romaz said:

It was the forgone conclusion of most people in the room that the sound through the attached Harbeth speakers would be vastly better with the Master preamp in the chain.  They were surprised when this was not the case.

 

Here is another example of a dealer's DAVE driving an $11k Constellation Inspiration Stereo Amplifier both with and without Constellation's $9k preamp.  To both the dealer's and my ears, SQ was better without the preamp and so when this dealer sells a DAVE, he no longer tries to promote the sale of a preamp:

 

When you say SQ was better, by what characteristics are you defining SQ?  Transparency and resolution only?  After eliminating as much noise as possible you're left with coloration, soundstage, dimensionality or depth and other characteristics a component can add/change to the sound.  Did the DAVE best the Constellation and VAC in all characteristics?

 

An excellent write up as usual Roy.  Your insight and experimentation will be sorely missed in any forum you participate.  Does this mean you will no longer be experimenting, or just that you will no longer be sharing your results?

 

Share this post


Link to post
Share on other sites
5 hours ago, romaz said:

 

Much has been said about how differences among servers are perhaps greater than difference among DACs.  In my own experience, I both agree and disagree with this observation and I will attempt to explain.  In my view, the DAC is clearly the more important component and why some DACs sell for >$100k.  Because this thread was never meant to discuss DACs, I have shied away from commenting but given Moussa's post, I figured I should comment although this will represent my last post on DACs on this thread.  In fact, this post will represent the beginning of my exit from posting in forums in general.  Life has become too busy.

 

When putting together an audio system, people will have their priorities.  I have already stated mine and they are simply (1) resolution and (2) transparency.  My reference isn't vinyl or tape or the million dollar setups one can hear at RMAF, Axpona or Munich, my reference is the live music I am often exposed to.  Most of what I listen to is unamplified acoustical music, whether it be large orchestral, small ensemble, choral music, or solo instrumental (especially organ but also piano and guitar).  When I am at an acoustical performance, whether it be classical or jazz, the first thing I notice is the acoustics of a venue and the resonances that venue provides.  The natural reverb and decay of instruments and voices are quite evident and from the stalls to the balcony or from one venue to another, they will vary.  

 

It has been stated that the reverberation time in a large venue like Carnegie Hall measures between 1.8 to 2 seconds.  At the Alice Tully Hall in the Lincoln Center just a mile away, this more intimate arena has a shorter reverberation time of 1.4-1.5 seconds.  Which is preferable depends on whether I am listening to a solo guitar, four string quartet or a full orchestra but regardless, I very much enjoy hearing the acoustics of a great building and never would I prefer to hear music in an anechoic chamber.  This is where most DACs stumble and where I find the DAVE excels.  This is also where I find PCM superior to DSD.  DSD provides you an expansive and a soft "tube-like" sound but this softness, which can be a wonderful way of masking the harshness of many chip DACs also results in a diffuse and imprecise presentation with respect to depth and timing and my careful A/B of my own recordings has convinced me of this.  As someone who values the accurate spatial portrayal of a live musical performance, I have found that a good music server can provide much but a good DAC can provide more.

 

When talking about resolution, as we look at our PCM files, we are provided 2 types of information:  (1) bit-depth and (2) sampling rate.  For Redbook, this means 16/44 which translates to 16-bits of dynamic range and a sampling rate of 44 kHz.  While DR is important, I contend that sampling rate is much more important with respect to a DAC's abilities.

 

When people talk about dynamic range, most people think about how loud and dynamically a DAC can play when really, it's about how quietly a DAC can perform that is important.  With regards to DAC performance, Rob Watts equates DR to the "hiss level" of the DAC and the greater the DR, the less likely you are going to hear "hiss" when no music is playing.  There is a DAC (that I will not name) that sells for >$100k and boasts a DR of 173dB (or 28.8 bits of dynamic resolution) as if we should be impressed by this.  For those that know better, this performance metric is useless since most believe most humans are incapable of hearing beyond 21 bits of dynamic resolution.  Just as important, most ADCs are also limited to about 21-bits of DR and so when people talk about 24-bit recordings, they often don't contain a true 24-bits of dynamic range.  Even at 24-bits (or 144dB) of dynamic resolution, for those who choose to look at DR in the traditional way of how loudly something can play, listening to any sound at 144dB SPL would be considered lethal. Now this is what people fail to realize -- as soon as you connect DAVE (or any DAC) to an outboard headphone or speaker amp, you now have thrown away the DR capabilities of your DAC because now, you've buried the DR performance of your DAC into the much higher noise floor of your amplifier.  For those who use an outboard amplifier with their headphones or speakers (this means most everybody who do not own a Chord DAC), you're basically listening to the much more limited dynamic range of your amplifier which is typically between 16-18-bits.  

 

With regards to sampling rate, I will explain why I consider this to be the more important spec with regards to DACs and this is why most DACs cannot match the performance of the very best turntables.  Sampling rate gives you a measure of timing resolution and this provides you not just spatial information such as depth but also timbre accuracy and the layering of fine detail.  With analog sources, you are hearing a continuous waveform and SQ is limited only by the quality of the gear that transmits this waveform.  As such, it is generally easier to get great sound from an analog setup such as a turntable.  With digital, an ADC is responsible for sampling the analog waveform a specific number of times per second and the larger the number of samples that are taken, the fewer the gaps of missing information there are and the more fluid or "analog" the recording sounds.  In theory, a waveform that is sampled 176,000 times per second (hi-res PCM) will sound better than a waveform sampled only 44,000 times per second (Redbook).  If that waveform is sampled an infinite number of times, then from a mathematical standpoint, your digital file becomes equivalent to your original analog waveform but as we know, infinite sampling is not possible based on the technology we have today and so this would suggest that digital can never truly equal analog.

 

However, there is the practical matter of the limitations of human hearing that potentially make it possible for digital to equal analog.  Most scientists agree that the human brain/ear has the ability to discern 2 separate sounds if they occur at least 5-7µs (microseconds) apart and so this represents the limits of a human's auditory time resolution abilities.  This means that when 2 sounds occur 10µs apart, as an example, we can hear 2 discrete sounds but when these 2 sounds only occur 4µs apart, instead of hearing 2 discrete sounds, we hear only one blended sound.  This is the rationale for why digital sounds "discrete" and why analog sounds "continuous."  With Redbook, as previously stated, sampling occurs 44,000 times per second and this equates to a time resolution of 20.8µs.  Anyone comparing a CD to vinyl in a resolving setup should easily be able to discern that with a CD, information is clearly missing.  As you sample more often, let's say 96,000 times per second, time resolution improves to 10.4µs and while this represents a significant improvement, most ears will likely still be able to detect that an analog source provides more information.  When you use an ADC to sample a file 192,000 times per second, time resolution now improves to 5.2µs.  In theory, at this sampling rate, a digital file should sound virtually indistinguishable from the original analog wave form and so this is the basis for why hi-res files were created.  This would suggest a 24/192 hi-res PCM file should sound equivalent to the original analog waveform.

 

For those who have done careful listening, however, with most DACs, 24/192 does not equal analog and even DXD or DSD256 files still can't match the resolution of the very best analog setups.  At most audio shows you attend, when you ask a certain exhibitor to give you their very best presentation, if they have a turntable or a reel-to-reel present, quite often they will switch to their analog source and, in fact, I have witnessed this many times.  As a further example, having visited the Magico factory in Hayward, CA recently, they have arguably the finest listening room assembled in the world today.  This room cost them $250k to build and has the equivalent of a floating floor and no parallel walls to avoid standing waves.  Short of an anechoic chamber, it perhaps has the lowest noise floor of any listening room and they use this room as their lab.  In fact, it is how they voice their speakers including their $600k Magico Ultimates and their soon to be released $175k M6.  Here is a photo of that room:

 

59c82952949e6_Magicolisteningroom.thumb.jpg.a471bfd9423c20e2b246c5ff6099f573.jpg

 

Because Berkeley DACs are the local favorite, they use a Berkeley Reference 2 DAC (Berkeley is headquartered nearby) fronted by a Baetis Reference server.  However, when they wish to present their very best, they revert to their turntable.

 

The reason is not so much because this sampling theory is faulty but because ADCs have limitations.  It is the reason why such technologies like MQA were created and why many DACs oversample.  Those in the NOS (non-oversampling) camp suggest that NOS DACs sound more natural but NOS strives only to reproduce the best that the ADCs can offer, warts and all.  Oversampling is much more ambitious and strives to overcome the limitations of the ADC by interpolating the missing bits of information through the use of sophisticated mathematic filters.  If the oversampling is done perfectly, a 16-bit Redbook file originally sampled at 44kHz per second should be audibly indistinguishable from the original analog waveform and this is the basis for the long tap-length filters that Rob Watts has been championing for decades but also the basis for what HQPlayer tries to accomplish.  As to who does it better, I will leave it for others to decide for themselves but having listened to both approaches, I much prefer Rob's approach.  As to the benefits of oversampling to DSD vs PCM, people will have their preferences, I have already stated mine.  

 

Regarding why some people fail to recognize great differences between DACs, I hear this all the time and I believe there are several reasons.  As both a headphone and a speaker listener, I have found both types of listening to have their advantages.  Headphones have the ability to portray fine detail better while speakers can image and soundstage better.  

 

DAVE is unique because its headphone output doesn't utilize a separate headphone amp.  When you plug a headphone into DAVE's headphone jack, you are actually listening to the DAC itself.  This means your headphone is tapped to DAVE's full bandwidth, ultra low noise floor (-180dB), dynamic range, and time resolution.  Moreover, what is unique about DAVE is it has no noise floor modulation and so whether you listen to music at low levels or at DAVE's peak levels, noise floor remains at the same ultra low levels.  There is simply no cleaner, clearer, more transparent way of listening to music than this.  The problem with headphone listening is that headphones do not portray depth well, certainly not as well as speakers and so to this degree, a lot of DAVE's performance cannot be fully realized through headphones alone.

 

The problem with listening to speakers with DAVE (or any DAC) is that DAVE's performance is largely buried in the amplifier you use to drive your speakers.  While DAVE's performance still shines through, its performance is blunted as you end up inheriting many of the limitations of even the finest speaker amplifiers.   Just like with outboard headphone amps, no speaker amp can match the performance characteristics of your DAC and so what you get with even the finest amps is a diminished photocopy of the original.  

 

Throw in a preamp, no matter how good, and this further adds to a loss of transparency.  That is just the nature of adding components to your analog chain.  Unless you are using a preamp for sound tuning (ie tube linestages), or you have an amp that demands a certain preamp to function optimally or unless you have multiple sources you need to switch among including a turntable, with DAVE, the very best preamp is no preamp at all.  Just like with amplifiers, no preamp can match DAVE's performance with respect to distortion characteristics, noise floor, speed, dynamics, or time resolution.  Even more, Rob programmed into DAVE the ability to attenuate down to whisper levels with absolutely no loss in resolution.  That means that as you attenuate DAVE to its lowest level (-75dB), DAVE is still outputting full resolution, something that no preamp can match.  

 

In the photo below is VAC's very highly regarded Master preamp (about $30k):

 

59c8b1c00f056_VACpreamp.thumb.jpg.375cee006d04903d4d7b282706624c0d.jpg

 

Kevin Hayes, VAC's designer, was kind enough to allow me to compare my DAVE driving his wonderful VAC tube amplifiers both with and without his Master preamp:

 

59c8b207a8935_VACpreampwithDAVE.thumb.jpg.4f43aeaa034e0bfaa90fe35fd4059bc3.jpg

 

It was the forgone conclusion of most people in the room that the sound through the attached Harbeth speakers would be vastly better with the Master preamp in the chain.  They were surprised when this was not the case.

 

Here is another example of a dealer's DAVE driving an $11k Constellation Inspiration Stereo Amplifier both with and without Constellation's $9k preamp.  To both the dealer's and my ears, SQ was better without the preamp and so when this dealer sells a DAVE, he no longer tries to promote the sale of a preamp:

 

59c8b3c8e95ff_DAVEconstellation.thumb.jpg.58e4b52d266ff41781b465b735bebfa8.jpg

 

And so what Moussa and ElviaCaprice are hearing is something that is very unique.  Through their high-efficiency speakers, they are hearing the full potential of their Chord DACs limited only by their choice of cabling and speakers.  With either the Omegas or the Voxativs I am using, I am hearing every bit of detail that my best headphones can provide while also the imaging and soundstage that only speakers can provide without the resolution and transparency robbing  impact of an outboard preamp or amplifier.  At the present time, I am trying out a pair of $25k Martin Logan Renaisssance Hybrid Electrostatic speakers in my large listening room, which I find to be very resolving and transparent.  These speakers are currently being driven by a pair of Pass Labs XA60.8 class A monoblocks ($13.5k for the pair).  While I cannot deny how wonderful this sounds when fronted by my DAVE, compared to DAVE directly driving my more modest pair of Omegas, this latter setup still sounds more resolute and more transparent.  This is possible only with Chord DACs because only Chord DACs (as far as I'm aware) have output impedances that are low enough to directly drive speakers.  In the case of DAVE, which has an output impedance of 0.055 ohms, this equates to a damping factor of 145, which is stellar.  Soon, Rob Watts will be introducing amplifiers that will connect to his DACs via digital interconnects (not analog ones) and will have the same resolution and transparency characteristics as DAVE directly driving speakers.  Essentially, these amplifiers will be "invisible" meaning they will have no character of their own.  They will have class A output and the first amplifiers will output either 20 watts stereo or 70 watts in monoblock form.  This technology is supposed to be scalable where 200 watts of amplification will be possible.  

 

Furthermore, as I have alluded in other posts, I have added Rob's new M-scaler to my DAVE.  This is incorporated into Chord''s new Blu Mk 2, which is a CD transport that also includes a USB and BNC SPDIF input.  This increases DAVE's TAP resolution to just over a million TAPS.  This is a milestone that suggests Redbook is now completely indistinguishable from the original analog waveform and Rob didn't believe it would ever be achieved when he first conceptualized it back in the 80s but because of the rapid advancement of FPGA technology, this indeed has been achieved.  Practically speaking, this results in a massive improvement in DAVE's resolution, so massive that the collective impact of my server mods which includes 8 clocks being replaced pales in comparison to what Blu Mk2 provides.  For those of you who own a Chord DAVE, I would suggest you prioritize getting a Blu Mk2 beyond anything else discussed on this thread.  Combined with Chord's upcoming "digital" amplifiers, there will be no more resolute or transparent way of listening to a digital file.  Despite all of this, I am finding, however, that the quality of the music server still matters.

Interesting post, however, what many people tend to forget is that most new albums that are pressed on vinyl, have been through the digital machine. Very few pure analog recording studio's still exist today. Even quite old recordings have been digitally remastered before they are pressed onto vinyl. Something to think about. Honestly, I never heard a top class analog system in my life. Something I should do in the near future, with the risk of being somewhat disappointed for the rest of my life. This already happened to me in the digital domain, the day I heard a Steinway-Lyngdorf model D. Sorry this is my second post mentioning Lyngdorf. Guess what I have at home :-)

Share this post


Link to post
Share on other sites

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

×