Basically, yes......
But the whole subject is really
MUCH deeper than it usually seems to be.
In general, data signals come packaged with their own clock.... which may be part of the same signal, or may come in on an entirely separate wire (which is often better).
Both types of S/PDIF signals (optical and electrical) are the sort of signal where the clock is part of the signal....
(you can say that the signal is "self clocking" or that "the clock can be recovered from the signal").
Regardless of how you phrase it, this means that you are relying on the source device to do the clocking for you, and it is possible for the clock to degrade along the way.
The original (isochronous) type of USB connection was the same way.
(Without some sort of clocking or re-clocking, the jitter performance of USB outputs is notoriously bad.)
Now, in a situation like that, you can either use the clock as-is, or use some sort of filter to smooth it out (old USB DACs often used a fancy sort of filter called a PLL - a phase-locked loop - to do this).
(A PLL acts sort of like an electronic version of a flywheel..... and, by doing so, smooths out the incoming clock signal - at least to a degree.)
Modern USB inputs are asynchronous.... which means that the receiving device provides the clock instead of the sending device.
This is specifically beneficial because we assume that the receiving device uses a pretty good quality clock.
And, since the clock is in the DAC, we don't have to worry about the signal getting messed up
BETWEEN that clock and the DAC either.
The subject of the ASRC (sample rate converter) is so interesting because the process of converting from one sample rate to another isn't especially beneficial.
The benefit of using an ASRC derives from the
WAY in which the conversion is performed.... in fact, the jitter removal is really not the purpose... but rather a really useful side effect.
The way an ASRC works is basically that you feed in a signal, and you feed in a clock, and what the chip does is "convert the incoming signal to 'fit' the clock you gave it".
Because
YOU get to provide the chip with the "new" clock, if you ensure that you provide it with a very good quality clock, the output signal will also be very good quality (low jitter).
The details of how the chip locks onto both the clock and the incoming signal, creates a new version of the signal to match the clock, and ignores the effects of any jitter on the original signal, are
VERY complex.
(Suffice it to say that the chip contains a very powerful fixed-purpose DSP chip and is doing a lot of work.)
The ASRC between the DSP stages in the XMC-1 actually serves several purposes, but one of them is to re-clock the audio signal passing through it - which works on all the audio.
(The XMC-1 also has an asynchronous USB input.)
Since most modern DACs have asynchronous USB inputs these days, USB is usually the preferred method for digital audio.
(You will still find older DACs that lack an asynch USB input; and a few esoteric modern models use the older method because they insist it sounds better... in spite of inferior measured performance.)
KeithL , I think I remember you also stating in an earlier (fairly recent post) that Optical S/PDIF uses the Transmitters Clock and doesn't have the ability to perform Asynchronous Sample Rate Conversion. I assume the same is true for Coax (copper) S/PDIF?
And I'm reading your note here as saying that ASRC is rare on HDMI (at least for processors). But you're explicitly saying that the XMC-1
does perform ASRC on its HDMI inputs.
So it sounds like the preferred method for Audio Data Transfer should be USB (if available)? Is that true for all/most USB implementations? Since USB transfers packets of Audio Data it seems like it would be a requirement ...
In any case, since the only digital inputs available on my old DMC-1 are S/PIDF, I'm looking forward to glorious new sound with the RMC-1 when it comes out!
Casey
