I had to go back and reread what the original question surrounding this was.
And to be clear, jitter in regards to networking - I'm not really clear on, besides a basic understanding of transport protocols, once the IT bar gets raised beyond basic networks, I spread Chee-zee Poofs around my desk and club the IT guy that gets closest and force him to do my bidding.
Digital audio I've been working with for a while. From the
stone age to
now.
Quote:
What a DAC does is it takes this digital rendering of a sound wave and it reassembles the analog sound wave. To do this properly it needs to have a clock sync so it can establish the correct interval of the samples.
Absolutely correct.
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Think of it like a plotter. The timing is the rate at which the paper is moving under the pen. If the paper isn't moving at a constant and determined speed, you are not going to reproduce a proper graph. It will either end up drawing the wave over a longer duration of time than the orignal or a shorter duration.
Which is why digital audio is reframed every frame to the internal oscillator. Again, the part I'm unclear on is how the jitter is supposed to let the stream run long or short with regards to time. If you could shed some light on that - the actual practicality of what occurs to disrupt the waveform - do you suggest that some bits pass other bits on the wire (or in the fibreoptic bundle) to arrive at the receiver before others? That some travel slower than others and cause a traffic jam on the wire? That the sending unit somehow loses time and starts firing the info at a non-regular rhythm? Or that both are like drummers and each ones' oscillator is off-beat with the other, and one is receiving on the upbeat while the other is sending on the downbeat?
In home audio, there are sometimes issues with digital audio, my DVD player transport is sometimes slow in switching over chapter lines, and the data becomes corrupted. My receiver will switch from Dolby Digital or DTS ES 6.1 or whatever it's decoding and revert to the standard DSP for the DVD connection (Neo 6 Cinema in my case) when it loses any part of the data. This happens maybe once a movie and only happens with this one DVD player. It's a bit of a PoS (curse you Toshiba!)
And on the other side of things, something I go back to a lot. What do the pros use? Whatever the guy actually cutting the movie, TV show or sporting event is using is probably more than good enough for the home user. And what is that standard? AES/EBU, SPDIF's big brother and nearly identical except for voltage (AES3 is 3-10v, I think SPDIF is <1V), connector (balanced or unbalanced XLR or BNC vs "RCA"/whatever that little toslink connector is called) and some of the data transfer.
It's really a great system, well designed, flexible and well thought out by the Audio Engineering Society and European Broadcasting Union.
Bren R.
