Why the preoccupation with jitter?

[Ulas]

Isn’t there more to digital audio than jitter reduction? And if you value jitter-free clocks so highly, why don’t you look at the jitter in the clocks that actually control the DAC chip instead of concentrating all your efforts on the XO.

For example, there is an asynchronous re-clocking circuit floating around the 'net that uses a low-jitter XO to generate the clocks that drive the SCLK and FSYNC pins of the CS8412 running in mode 3.That’s all well and good, except, like all asynchronous re-clocking efforts, the downside is occasional missed or duplicated samples. What’s more, that particular circuit uses an HC4040 to divide the XO clock rate. Have you looked at the spec sheet for the HC4040? It’s an asynchronous counter, sometimes called a ripple counter. Only the first stage is actually clocked by the input clock. Each of the following stages is clocked by the preceding stage. The Fairchild data sheet shows the interstage propagation delay to be not more than 25ns at 25C with VCC=4.5v. That means, with 5 stages between them, FSYNC will lag SCLK by as much as 125ns. That’s worse than the CS8412, which guarantees the SCLK/FSYNC delay to be not more than +/- 20ns. The 125ns max propagation delay would not be a problem if it were absolutely constant - but it’s not. Given the slow and unmatched, 10-15ns rise and fall time of HC logic and 5 clock stages in between SCLK and FSYNC there’s a large margin for jitter to be introduced. I’ll bet the reclocked FSYNC from the HC4040 has more jitter than the original FSYNC coming from the CS8412.

I know you all claim to hear improvements after your efforts to reduce jitter, but have you actually measured the jitter at the DAC chip? It could be that after you feed the pristine output of your low-jitter XO through a lot of shoddy digital circuitry, you end up with more jitter, not less. Jitter is like dither in that they both introduce a small amount of random error to the reconstructed audio signal and it has been shown that a small amount of dither actually helps with the retrieval of low-level information. A small amount of additional jitter will do the same thing.

Now, back to the original question: Isn’t there more to digital audio than jitter reduction? I think so. I just finished a very interesting new DAC design. It starts with a CS8412 and differential PCM1704s but adds several new wrinkles. One is sub-LSB dither; another is … (I think I’ll save that for another day. If all you want to talk about is jitter I don’t want to waste your time with extraneous concepts.)

[Lars Clausen]

Surely there is more to Digital Audio than Jitter.

It's just that at this point the DAC chips are made good enough, that you can barely hear difference. But there is great difference in the sound of different apparatus using the same DAC's. Why? Well assuming the data stream is the same, the only difference left can be the analog stage (which is very critical if you want to have a good sound performance) and ... yes: Jitter.

I agree completely after a clock recovery where the clock phase is depending on the data content of the SP/DIF signal, you can not claim to have a low jitter clock. This is why a low jitter clock should always be fitted in the DAC apparatus, and feed the DAC chips directly. And secondarily a clock signal should be fed back to the transport to make it ruin the same speed (synced operation).

All the best from Lars Clausen

[Lars Clausen]

By the way: You don't need to use the all expensive PCM1704's

Several new DAC chips offer as good or better performance (and sound...!) at a small fraction of the price for PCM1704.

Take a look on www.cirruslogic.com or www.akm.com . Both these firms have very very good DAC chips for almost no money.
CS4396 is a great standard converter, which i will recommend for CD use, but they have others that are even better, and also can convert DSD. Again, you may find that the DAC chip performance is not your bottleneck .. more likely jitter is!

[jwb]

I'll agree with Ulas on one point. There have been several DIY DAC designs that are certainly going to have repeated or dropped samples, just because no care is taken to ensure that the clock domains are synchronized in the DAC and transport. These same designs tend to voilate the setup and hold time requirements of the ICs involved. These are "cargo cult" designs, slapped together from promising-looking bits of other circuits.

But, that doesn't mean one shouldn't have a good clock in one's own DAC!

[tiroth]

I’ll bet the reclocked FSYNC from the HC4040 has more jitter than the original FSYNC coming from the CS8412.

Yes, but anyone who has thought about it will just use a syncronous counter.

[Cobra2]

But I agree with first post...there is more than the DAC that needs the correct timing.
I have mentioned (in another tread) that I EASY could hear the difference when a (tunable) clock was out of tune/frequency compared to RIGHT frequency.
And I have measured a lot of X-tals since, and few are "right".
So I guess this also influence the diffrerence many hear when upgrading clock-circuits.

ArneK

[Christer]

Although people may have different opinions on the practical
value of jitter reduction, at least this is something that can be
straightforwardly explained in theory. A timing error in the
digitial domain will translate into an amplitude error in the
analog domain. How much it matters in practice can be
discussed, but the effect is there according to the theory.

There is another aspect on the jitter problem that I have
been wondering about and that I cannot remember ever
seing discussed. While audiophiles have for quite some years
put a lot of effort into jitter reduction in their CDPs and there
are a lot of replacement clocks available, what is the situation
at the studio end, ie. in the AD part of the chain? Are the
AD converters used for recording designed for low jitter,
or is this a still neglected thing there, or has it perhaps been
known and taken care of since long ago? Anybody who knows?

[Lars Clausen]

It has been adressed through the last 30 years in studio environments. Every studio has a low noise 'house clock' to syncronize all digital devices. It runs directly on the sampling frequency that the studio is producing media for. In case of a CD, it would be 44.1 kHz.

One example can be seen here: http://www.iclock-net.de/e_iclock.htm

A studio owner talks it out here: http://www.discmakers.com/pse/jung.php

[Christer]

Thanks Lars,

that was interesting as well as comforing reading.

[Ola]

At least in Rotel CDP (991?) the introduction of dither (it's switchable in this machine) was much less heareble that making the clock better.