Importance of clocking

The issue with S/PDIF or AES "reclockers" is that you have two clocks arguing over what should the absolute clock rate be.

The DAC is forced to take one of two approaches: Abandon it’s internal clock or attempt to keep it’s internal metronome and "fix" upstream deviations from it’s own mechanism.

How does the DAC know? What do you mean by 'abandon it's internal clock'? Are you talking about the external clock input of a DAC substituting its internal one?   Don't you agree that the better (more accurate with less jitter/noise) the incoming data on AES/SPDIF, the easier the job of the DAC's internal clock?

@greg_f   There are two different ways of dealing with S/Pdif.  One, very common way, is to adjust D/A clock to average frequency of incoming S/Pdif stream while another is to completely ignore S/Pdif clock, strip the data and send it to D/A converter at different rate (Asynchronous Rate Converter).  

In first case data is often oversampled being sent to D/A converter in exact multiples of received frequency, while in the case of Asynchronous Rate Converter, D/A conversion frequency and incoming signal frequency are independent.  My Benchmark DAC3 D/A converter runs always at 210.9kHz - no matter what signal frequency is (it was 110kHz in DAC1).

CDP output data can be buffered and sent at exact intervals since both buffer clock and motor speed come from the same source (quartz crystal), but receiving D/A converter has to be somehow synchronized, otherwise samples may be lost.  In older devices it was Phase Lock Loop (PLL) analog circuit adjusting Voltage Controlled Oscillator (VCO) - D/A conversion clock.  These days everything is digital, so PLL circuit is also likely digital.  Either way, in this scheme D/A conversion rate is not constant, but adjusted to average frequency of incoming S/Pdif stream.

@kijanki  Thanks for the explanation. My DAC is an older Weiss DAC202 that uses ARC. It has extensive fairly advanced PLLs (it uses a wider range digital PLL (I don't know its noise/jitter spectrum) and its output is fed to an further analog PLL to reduce the jitter of the first PLL. I believe both PLLs are implemented on a single DICE IC which was state of the art back then. I used to have a datasheet but I must have lost it during a move so I uncertain of the exact operation. The DAC uses older ESS converters and I think they always run at 1.536MHz. To complicate the clocking scheme further, the AES output of my source feeds a Trinnov ST2 Hifi (digital room correction device) which also uses a DICE generated clock and its AES output then goes to the DAC.  I need to refresh my memory!

My issue is that my DAC doesn't have a USB i/f and therefore I need a streamer with either an AES/SPDIF output or an additional DDC to convert the USB.

Your thoughts please?

@greg_f I remember that Weiss DACs had Firewire. I had Firewire HDs with my old Mac computer and used them with the newer generation with an adapter. Advantage of Firewire was that it had, being extension bus, sustained bandwidth. Peripheral buses like USB don’t have that running under protocol. This sustained bandwidth made it very popular in Audio/Video industry. The great feature of Firewire - Direct Memory Access (DMA) killed it. Once somebody made pocket Linux to run on the phone it made it dangerous. Hackers could connect it to portal in large company and bypass (or even obtain password).

You can try it. Other than that you have only choice of coax or Toslink. I use Toslink because of my DAC’s very strong jitter suppression to avoid electrical noise injection and possible ground loops. On the negative side Toslink drivers have slow edges making it noise susceptible (noise on the edge modifies threshold point affecting timing). Toslink works better when system is clean of electrical noise. Faster edges of coax are less sensitive to noise but create another problem - reflections from characteristic impedance boundary. You need to use impedance matched cables possibly 1.5m or 2m long (to make first reflection miss original edge) or very short cable, Rule of thumb is that transmission line effects (reflections) start when propagation time one way is longer than 1/8 of transition time. For typical 25ns transition 1/8 is about 3ns making it 0.6m cable (assuming 5ns/m). Since it includes all wiring inside on both ends I wouldn’t go further than a foot. The shorter the better. As for the conversion from USB - that might be good, especially if it contains electrical isolation.

@kijanki Thank you for the response. I enjoy reading your replies since you seem to understand electronics better than most on this site. Sometimes it is difficult to have a technical conversation with people who have little knowledge even about the basic concepts but they call themselves experts. I find audio a strange field... Your analysis on transmission line is spot on. For spdif I rather have proper 75 Ohms BNC terminated cable than with RCA plugs, I had an argument with somebody who was saying it is only audio, there is nothing high frequency so no need to worry about reflections, what he forgot is that reflections are caused by the rise/fall time of the edges and not the waveform frequency or data rate. Anyway, I diverted :)

Yes, yes, I much rather use Firewire than USB, in fact I still use a PC (with very little optimazation) with a PCIe Firewire card into the Weiss. But Firewire is getting old and new device of course don’t support it. I also use the Weiss AES i/f which is not too bad, almost as good as the Firewire. As I am interested in Qobuz, my next purchase is going to be a streamer, the way I see it is either fully optimize my server (both hardware and software), use the Firewire card and hopefully find some software that will support it, or buy a relatively decent streamer like Lumin/Innuos/Aurender and possibly a DDC as USB look like being the most common i/f these days. Unfortunately where I currently live there is no way to demo anything so I have to buy by what I read on forums and reviews... not the best way.