[1000a]

hit refresh I updated 3 times----The following concerns 2 channel audio specifically,

I have just spent tons of time reading on the lessloss website  that ideally the clock will not be used (bypassed) in a transport section, and only implimented inside an outboard DAC because that always yeilds the best audio by having a greater improvement. 

So I guess where are we in the current state of DAC's, CLOCKS and so on in regard to the SB?   So to be very specific I'm wondering where is the clock in an SB3 to outboard DAC set up?   I had also read some positive coments about using the MONARCHY DIP and their similar devices in the Paradisa DAC thread.

below is from the LessLoss audio website (their take is the CDP is slaved - by passing the clock inside completely) and taking the digi signal to their DAC where the Clock is sitting in the ideal place.

"Well, that's partly true. The entire audio business of CD transports and DACs is built on the totally backward setup of the CD player containing the Master Clock and the DAC being the Slave. This results in an entire pallette of innovations and experiments to lessen Jitter, starting from air drives to expensive digital cable technologies with complex math to reduce line-induced Jitter, to very carefully filtered power supplies, to all sorts of other things when striving to achieve the least possible Jitter. So we have the worst possible digital scenario resulting in the worst possible starting point to achieve jitterless signal transfer, because it is impossible to annihilate Jitter when the CD player is the Master Clock. If you're looking for quality, this is stupid, to say the least!"

see whole article
http://www.lessloss.com/about.html  (fasinating site to say the least - any comments on their ideas?) 

If this is an ideal I wanted to achieve is it doable with a SB3 and an outboard DAC?    or can it be approached using devives like the Monarky stuff or is that just more noise on the line and side skirtting proper implimentation? I've had some knowledge of this stuff before but now it eludes me and I am wondering how these ideas might be employeed in the SB situations.  Without dropping stupid $$ on a devive thats about to become the SB4.

I have found other chatter around here on the same wave length,

"Jim Levitt
So back to my original question: how are the AVA dacs designed to handle the incoming data stream so that "jitter is simply not an issue"? Do they buffer and reclock? Or are they dependent upon the clock from the transport? I have no doubt that the rest of the design is superior. I'm only asking about the interface with the transport and cable."   

[1000a]

much as I suspected I have found differing opinions on the implementation of this being worthwhile see:  http://forums.slimdevices.com/showthread.php?p=207728

I do think out of curiosity I will ask Lossless if there is a way to bypass the SB3 clock to use with their DAC.  Although I believe their DAC to be in the 1,500 price range.

  I always consider this "An improvement yes, but how much $$ for the improvement and just how much improvement?"  I am getting massive improvements thru DIY room treatments, halos on the tubes (they just released an amazing new driver and input damper called blackbery) and using magwire for DIY power cords, ICs and speaker cables.  So I am now terribly spoiled when it comes to improvements.  They need to be on a very serious scale for me to consider big $$ (1,000) on them.

so evidently just where the clock is best is disscussed by people, but I have little science type of insight - it just confuses me to no end sometimes, I know when I implimented the inexpensive changes above I recieved stagering improvements.  So I know when I hear it, but I am now spoiled and feel there is still alot on the table that can be improved by me for fairly low $$.

Thats what got me to mention the Monarchy DIP - thinking this might get some improvement (related to that whole debate-clocks reclocking and so on) without spending what for me is too much money for not enough bang.  I'll have to ask Nick at Prometheus about his clock in his new DAC introded at 340. 

[crooner]

I wouldn't worry too much about this.

As long as your DAC is high quality with true 75 ohm BNC connections and a good coaxial cable, there shouldn't be any crucial problems with the SB3. Jitter will be in the low picosecond range and shouldn't be detrimental to the sound. However, a Monarchy DIP might prove worthwhile checking into, specially for the price. I got one myself (should get it this week) and it will be fun to experiment with, that's for sure!

Also, similar effects can be obtained by simply using pulse transformers at both ends of the SPDIF link. The Monarchy, I believe, uses such transformers internally.

For those truly obsessive, word clock technology is the answer. With a Transporter and an external "atomic" clock oscillator (such as Teac/Esoteric's rubidium based unit), jitter would be effectively banished. Now, the question remains, if the sonic benefits warrant the super high price tag.

For me, a good quality conventional DAC, preferrably with internal reclocking, and a low noise correct impedance SPDIF connection delivers the musical goods without breaking the bank.

I'll probably jump on the 'word clock' bandwagon, when prices finally drop!

[1000a]

I wouldn't worry too much about this.

As long as your DAC is high quality with true 75 ohm BNC connections and a good coaxial cable, there shouldn't be any crucial problems with the SB3. Jitter will be the low picosecond range and shouldn't be detrimental to the sound. However, a Monarchy DIP might prove worthwhile checking into, specially for the price. I got one myself (should get it this week) and it will be fun to experiment with, that's for sure!

Also, similar effects can be obtained by simply using pulse transformers at both ends of the SPDIF link. The Monarchy, I believe, uses such transformers internally.

For those truly obsessive, word clock technology is the answer. With a Transporter and an external "atomic" clock oscillator (such as Teac/Esoteric's rubidium based unit), jitter would be effectively banished. Now, the question remains, if the sonic benefits warrant the super high price tag.

For me, a good quality conventional DAC, preferably with internal reclocking, and a low noise correct impedance SPDIF connection delivers the musical goods without breaking the bank.

I'll probably jump on the 'word clock' bandwagon, when prices finally drop!

thanks a mill crooner

I was wondering if anyone was gonna pipe up on this stuff, sense the post I have have been around the net 2 times I found the end, and of course crashed into word clock stuff.  pricey I'll wait also (thanks for the knowledge I can relax, getting a little obsessive not me) --that is something to look forward to I love the trickle down cycle of the digital world in audio and photo, last years caddy is on its way to the somewhat entry level easily in 2 yrs if one is patient and does not have to have it yesterday!

I definitely have 75 ohm and so on - good stuff its just this DIP is starting to attract me and I can send the signal to the DAC with AES/EBU balanced cable instead of my Stereovox coax, also hoping that might up the anti.  Oh God please tell me a balanced cable is cheap and there ain't no diff between any of them!

Please post your feed back on the DIP I would be very interested to hear about it.  Did you order the Classic or the 48/96?  I think that's the only thing that's got me holding on my order. I do believe its certainly worth a shot.  Some on the Paradisa thread are very happy with their's.  I guess I am trying to future prof slightly!  The SB3 does not send the info in a manor for my DAC to do the upsample thing it is capable of.  It does not seem to matter sound is excellent. 

So I guess I am trying to second guess potentialities of the soon to be released SB4, all the while having my eyes focused on the Promitheus NOS DAC.

[crooner]

Actually, being the cheapskate that I am, I got a used DIP off eBay for peanuts. Should be able to resell for virtually no loss if I don't like it! 

I believe it's the very first version from the late 1990s, 16/44 (and 16/48), of course!

The Stereovox is a killer cable. I use the HDXV, and couldn't be happier. It's gonna take some good dough to improve it. Actually, in some circles, BNC 75 ohm is preferred to 110 ohm AES/EBU. Actually, the best (aside from Word Clock of course!) would be SPDIF2, but that's rarely implemented outside the professional world.

I recall reading the technical notes on some of my XRCD's and they state their preference for SPDIF2 over AES/EBU. And those remasterings sound amazing!

As for the upsampling thing, it actually only works for conventional 16/44 sources. High rez signals don't need it for obvious reasons. I'm a little weary of its benefits, unless you get something like the dCs Elgar that costs a fortune!

[richidoo]

Since you're into reading you might enjoy this:

Mother of Tone

It will take a while, but fun and has many potential rewards
Rich

[miklorsmith]

I own a Lessloss DAC and have another with superclock input and companion clockless transport in-house currently.  The latter setup is configured to do the exact comparison you're looking at.  It just arrived and I haven't formed any opinions on it yet.

The SB3 has an internal clock just like any disc player.

I was thinking about getting my SB3 modded for clock slaving, per the Lessloss recipe.  It's a fairly simple operation for a modder, but then your SB3 won't work with any other DAC - problem.  If I find it IS that great, I'll have it done.  It seems to me there should be a clock-active/deactivate switch possible like on the CD transport I have but so far that doesn't seem true.  Maybe the SB3 innards are too compact.

The Lessloss guys have devoted their operation to jitter smashing, it'll be interesting to see where this goes.

I think the current DAC price is around $3k with options that can go higher.  Without commenting on the clock master stuff, it's a very nice standalone DAC in its own right.

[1000a]

keep us posted on your findings, excellent and thanks a mil

[Bob Reynolds]

You might find the article Clock Jitter, D/A Converters, and Sample-Rate Conversion by Robert Adams (Analog Devices) of interest:

http://www.theaudiocritic.com/back_issues/The_Audio_Critic_21_r.pdf

[miklorsmith]

I'm not going to use a 13-year-old article to inform myself about something so new and changing as digital audio.

[NewBuyer]

...Also, similar effects can be obtained by simply using pulse transformers at both ends of the SPDIF link. The Monarchy, I believe, uses such transformers internally...

The Monarchy units use a pulse transformer at the coax output, but unfortunately not at the coax input.

The GW Labs DSP uses pulse transformers at  the coax input and the coax output - and also has AES/EBU input and output capability.

[Bob Reynolds]

I'm not going to use a 13-year-old article to inform myself about something so new and changing as digital audio.

Obviously, that's your choice. You know what they say about ignoring history. Exactly what about jitter and dacs do you think has changed?

For those with an open mind and a desire to learn something from an accepted expert, enjoy the article (though it's not an easy read).

[miklorsmith]

Let me flip the question back on you - what, exactly, about digital audio HASN'T changed in the last 13 years?  What digital products are you aware of that are still acknowledged as "competent" much less "exemplary" from that period?

[Bob Reynolds]

Let me flip the question back on you - what, exactly, about digital audio HASN'T changed in the last 13 years?  What digital products are you aware of that are still acknowledged as "competent" much less "exemplary" from that period?

SPDIF is still the predominant data interface format. Bits are still bits. The sources of jitter have not changed. The use of async sample rate conversion as a means of managing one source of jitter has not changed.

[miklorsmith]

Fair enough.  Point/counterpoint.

[Bob Reynolds]

One thing that changed from the early days of digital audio to the time of that article and continues  today is that the analog sections are much better. Regardless of what happens in the digital domain you still have to convert the current from the DAC into voltage. According to those early articles, the output sections added enough distortion to reduce the effective resolution by a few bits. It's possible that poorly designed/implemented analog sections account for larger sonic differences reported among digital components than the digital sections.

[1000a]

...Also, similar effects can be obtained by simply using pulse transformers at both ends of the SPDIF link. The Monarchy, I believe, uses such transformers internally...

The Monarchy units use a pulse transformer at the coax output, but unfortunately not at the coax input.


The GW Labs DSP uses pulse transformers at  the coax input and the coax output - and also has AES/EBU input and output capability.  But

Hi New Buyer

After hearing about the GW labs DSP a couple of times now and doing more reading I am going to try that one vs the Monarchy, but I thought it is generally considered better to run a opti cable from the transport to the DSP or DIP and the coax to the DAC.   Simple enough I can simply try both.  Does the pulse transformer not effect the optical input??

Hi Crooner

Where do I buy or check out the pulse transformers, if I wanted to investigate that also?

thanks in advance 1000a

[NewBuyer]

Hi New Buyer

After hearing about the GW labs DSP a couple of times now and doing more reading I am going to try that one vs the Monarchy, but I thought it is generally considered better to run a opti cable from the transport to the DSP or DIP and the coax to the DAC.   Simple enough I can simply try both.  Does the pulse transformer not effect the optical input??

Hi 1000a, as I understand it, the pulse transformer only applies to digital signal delivery through wire (as in coax or AES/EBU), not optical - you are correct. I tried connecting the SB3 to the DSP using both coax and then glass optical, and found the glass optical connection to be noticeably cleaner and better sounding. I do not know what would account for this - perhaps the difference is due to the even better isolation (between SB3 and DSP) when using the glass optical connection, or the lack of coax-cable impedance issues, etc. But I'm only guessing here. I hope you will please post your impressions too, after you have a chance to try it out.

[1000a]

Hi New Buyer

After hearing about the GW labs DSP a couple of times now and doing more reading I am going to try that one vs the Monarchy, but I thought it is generally considered better to run a glass cable from the transport to the DSP or DIP and the coax to the DAC.  Simple enough I can simply try both.  Does the pulse transformer not effect the optical input??

Hi 1000a, as I understand it, the pulse transformer only applies to digital signal delivery through wire (as in coax or AES/EBU), not optical - you are correct. I tried connecting the SB3 to the DSP using both coax and then glass optical, and found the glass optical connection to be noticeably cleaner and better sounding. I do not know what would account for this - perhaps the difference is due to the even better isolation (between SB3 and DSP) when using the glass optical connection, or the lack of coax-cable impedance issues, etc. But I'm only guessing here. I hope you will please post your impressions too, after you have a chance to try it out.

Hi New Buyer

  That is really interesting, I am guessing you tried the glass cable after reading my post?  So you definitely prefer it, correct?  From a reviewer I think this could be easily be the problem "First, I’d like to see 75Ω BNC connectors offered—perhaps as an option—on theS/PDIF input and output."   Everything I read tells me impedance matching in digital domain is crucial, I wonder if they have changed that?

UPDATE after post, I am beginning to think my DAC designer below was aware that most CDPs did not have have a 75ohm out, so just recommended glass (never the plastic) to make sure his clients skipped the issue.  What I am saying is I guess maybe glass is better at handling impedance mismatches, which might account as you said.  I' be willing to bet an old crummy RCA plug, that's how confident I am after 4 billion hrs. on the net reading audio crap!        

Anyway the guy who designed my DAC always encouraged his clients to use a glass cable with his DAC, I had a strong bias against them and never did and was quite happy w my HDVX.  A year or so later a guy told me he bought a 25-30 glass cable that he used and could not tell a difference between his HDVX and the glass.  Sure enough I bought one and compared the 2.    I had to really strain to hear a diff.  The only diff is the HDVX had ever so slightly extended? / slightly enhanced treble loudness-read not enough diff.  So having both connected to my DAC I can switch.  Honestly I could not tell you which one I am using at present. So 30. vs 150.up 30. works for me.

Now the posts I've studied with the experts on them say a digital coax should never be shorter than 1.5 meter minimum.  and then of course some of those same experts also say glass cable sucks.  My coax (have to check to be sure) and my glass cable are both 1m.  So the next time I need more of either I will hit the 1.5 min at the very least just to CMA.  I can not test every single thing.  One designer did say many more times than not there is usually an impedance mismatch from CDPs to external DACs and it just makes them sound bad period, it was a MAJOR issue for this guy.  Now I forgot is your SB modded did Wayne put a 75ohm digital out on the SB?   Mine is still stock, FWIW.

Now I forgot if you have the HDVX stereovox for coax, but the replacement VX2 supposedly stomps the old one?  so where that leaves me with glass and coax not sure, but I had every intention of trying glass from SB-DSP-coax-DAC.  and now every variation between all 3 pieces. You can demo the VX2, there is a link here somewhere.  This is not the first time I have read of glass being better than coax for people I'm not entirely surprised.  But I am excited cause I think good glass is far cheaper than good coax.


evidently GW labs does its clock info diff than Monarchy

"......The manufacturer also notes that the DSP generates its own preciseclock, and does not use the recoveredclock from the CD or DVD transport. This advantage is inherent in asynchro-nous sampling rate converter chips, but won’t necessarily apply to any product using the CS8420.

Cirrus Logic offers the option of either using the recovered clock or deriving a new output clock from a stable crystal oscillator. GW Labs has wiselychosen the second option."

fianally after this long corrected post I really think these differences between the cables are all related to impedance mismatch issues.   SB w out 75 ohm>> non 75 ohm DSP non 75 ohm>> DAC (mine does have 75 ohm there), but the rest of my picture looks out of whack.

calling all experts on impedance

[1000a]

reason for longer coax cables:  http://www.positive-feedback.com/Issue14/spdif.htm

and unfortunitly 75 ohm RCAs do not insure 75 ohm 1- cause they are never truely 75 ohm (not even in their nature) and 2-cause most times keeping all the associated linking inside DACs and inside the CDP or transports is almost never ever done at the 75 ohm needed figure to complete the entire path at 75ohm anyway, so it seems mismatched impedance really seems to be the norm for the great majority of us.  an honest modder could probably make this happen, but at what cost vs what improvement?

this is interesting cause in analogue box connections impedance mismatching also is the reason for bad sound (realtive of course) hence all the excitment about the Promitheus TVC and others cause if I have this right they help to correct impedance or get matchs much closer = better sound for most.  Could impedance issues be be the primary curse of seperate components.

Here we go back to circuit city, give me that all in one jobby but it has to be Mark Levinson do you carry his stuff?