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Asynchronous versus Synchronous

Frank:

You'd be surprised how many "hi-end" DACs have a wordclock output that will spend it's lifetime just sitting there collecting dust. 99% of "hi-end" guys simply use SPDIF over TOSLINK or COAX in "Asynchronous mode".

The Pros (and some hi-fi guys) know that asynchronous works - but it can really be prone to jitter. Typical "receivers" are CS8416, CS8420, and CS8421 which use what is called a "phase locked loop" (PLL) to deal with the fact that the clocked data coming is was clocked with a DIFFERENT clock that the clock than the one used for the receiver (and dac at the receiver end). The clocks don't have to be out by very much to be "skewed" - and without the PLL the clocks would eventually drift out of sync. The apparent trouble with PLL's is that any jitter before the SPDIF receiver is a whole lot worse going OUT of the receiver into the DAC.

So what some folks do is follow what is done in professional environments that handle digital transfers all of the time - they do synchronous transfers in what is called "master" mode. In master mode one device is deemed the master and it's clock is used. ALL other devices in the signal chain are then slaved to the master and use that same clock. One clock. No skew. No PLL. No sync issues. Brings a tear to your eye doesn't it?

Now to do this with the Lynx, I imagine that you would need to make it the slave and set it up to use an external clock. Why? I think quite a few DACs have "wordclock" output, but not too many at all can be set up as a slave and have a "wordclock" input. So, if I was the owner of a Lynx 2B, I would find a DAC that has a wordclock output and slave it. Another reason (and perhaps a better one) to make the Lynx a slave and not a master is that it seems using a PC to generate audio clocks is a "foul word". No matter HOW good the card, it seems folks around here are nervous about using PC's for digital OR analog circuits because of the noise that plagues computer power busses. Other folks counter-argue that good filtering of the supply is possible. Myself, I think "PC noise ruins everything" is a very very crude model: why can PC's have PRISTINE noiseless video output via a PCI card that relies on scanning frequencies in the kilhoertz but be totally useless for *audio*? I think the reason why so many PCI cards suck for digital is because they use asychronous SPDIF (and worse yet optical SPDIF which requires an electrical-to-optical and then optical-to-electrical conversion). I think PC SPDIF is no better that consumer SPDIF and depends on the hardware and the ability of the circuit designer to deal with noise and get clean power to the transceiver chip.

So this is why I've been waiting and have not sprung for ANY high-buck PC solution yet. I had/have so much to learn. I want to do this right. The best way. The low-jitter way. And if that means getting devices that use synchronous / master-slave connections with separate word clock connections then no problem. It's going to cost more sure, but not if I had done it the "trial and error" audiophile way.

Digital transfers are engineered solutions. The nice thing about engineered solutions is that they can work for you the first time. This is because the first 9 times were done "virtually" (on paper) and thrown in the garbage can. In my case, the "engineering time" (90% of which is spent learning) is free because this is my hobby. So it makes sense to do "free engineering" first and buy equipment later.

I've only bought cheap soundcards in the meantime just to keep myself amused, and learn about asynchronous SPDIF connections, and how a PCI card that has a world-class DAC that specs like a NASA rocket can end up sounding like thin and bloated crap. This, in my opinion, is not because of "parasitic PC power supply noise", but because of a multi-channel analog section that was scaled down to fit into the size of a postage stamp to fit on a PCI card.

Cheers,
Presto


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