johnsalvini

It looks like it might be Neotech NEVD-2001 silver digital cable.

Regarding clock inputs, I don't understand why a dac would have a clock input if it has to synch to the SPDIF (RCA, BNC, or AES) input. If that's the case, and it had a clock input, what would it be for? Both the Mytek Brooklyn & Manhattan dacs have clock inputs. And the dCS Vivaldi has 3 clock inputs. In the Vivaldi's case, it has 1 for multiples of 44.1, and another for multiples of 48. But if it takes it's clocking from the AES signal, they would all be redundant. Please explain. Thanks, John

Thanks Eloise, That thread had the discussion I was looking for. Too bad it ended with an expert stating some facts (which is good) but leaving out his opinion of the different interfaces and jitter reduction solutions (which I would really like to hear). The important thing to remember, i guess, is that the really smart guys, the ones designing the equipment, have to really study this stuff and their conclusions are mainfested in their final product. Its "interesting" to us cause we want the best sound for our systems, but its "essential" to them cause their success depends upon it. You never really know how good a solution is till you try it; and then only time will tell if a concensus will develop or a breakthrough will emerge. I'm new to computer audio (obviously...and still saving for the big day I can jump in), but I'm quite happy that so much energy is being put into its development and excited about what is...and will be...a wonderful thing.

Oops, hope I didn't cause any problems. The subject of this thread has been on my mind a lot lately and I was wondering if anyone has any further insights. Would there still be a delay or overshoot when fast forwarding & rewinding if the buffer were big enough to contain the whole disc? Like with Chris' CAPS server where there is a large SSD that can hold at least an hours worth of HD audio that is downloaded from a NAS, or streamed from internet, or from CD or DVD or whatever, and then play that back freshly clocked. If that would work, then it wouldn't be any worse than getting up to put in a new disc...except maybe a little longer. If that what it takes to bring the music playback as close to perfect as possible, then so be it. Thanks

How do you keep the first 27 letters of a post from being repeated. Thanks,

Also, It might also be a problem when you have digital active crossovers, or bass management, then you would still need your DAC to be the master and a clock "out" connection so delays could be applied as needed for processing. I don't know. maybe all that is done before the final D/A conversion.

Thanks Bob, That does make sense. It would be quite bothersome for movies. For music though, it would seem a price I would be quite willing to pay...I think. I like extra time between songs anyway to savor the last one and then be ready to appreciate the next one fresh. Anyway, too bad its not a practical approach. I'm still wondering though: Would it still have that characteristic (the delay or overshoot when fast forwarding & rewinding) if the buffer were big enough to contain the whole disc? Like with the Chris' CAPS server where there is a large SSD that can hold at least an hours worth of HD audio that is downloaded from a NAS, or streamed from internet, or from CD or DVD or whatever, and then play that back freshly clocked. If that would work, then it wouldn't be any worse than getting up to put in a new disc...except maybe a little longer. If that what it takes to bring the music playback as close to perfect as possible, then so be it.

After reading about computer audio on this site and manufacturers' sites for the last couple months, I've noticed a lot of talk about the interface (aes, spdif, i2s, usb, coax, etc.) but nothing about the idea that ps audio was using in their perfect wave dac to "bank" the audio data in a large onboard memory, and then clocking it at the d/a conversion. But they don't even talk about it that much. I'm just a woodworking guy so don't know much about the computer or electronics world, but from here that sounds like a "no brainer" way to eliminate the critical nature of syncing clocks and the interfaces between the source and dacs. So there must be a reason for it, but I haven't been able to find out what that might be. Can anybody explain? Thanks, John