Phil,
I take on board what you are saying but unfortunately don't know enough
about the subject to say whether I agree or not. However there are 2 readily
observed phenomena (one purely objective and one arguably subjective) which
say there is something not right about the squeezebox's digital output.
1) The fact that my dac locks on with different qualities of lock when mp3
and flacs are played back. Suggesting that there is indeed some difference
in the spdif data stream between the two formats.

2) The 'subjective fact' that the squeezebox sounded worse than a cd
transport.

If jitter is not the cause of these observations they I'd ask you if you can
propose an alternate reason for them.

Secondly we aren't talking about reclocking the dac. We are talking about
reclocking the squeezebox - specifically taking the spdif output, stripping
the old clock data out and adding a more accurate one to the data stream. If
I understand correctly this is what both the trichord and tent xo3 boards
do.

As for not using a dac, well the analogue output stages and filters inside
the squeezebox are poor compared to that in a decent off board dac. Even
with the jitter / unknown problems effecting the spdif output it still
sounds better than the analogue outs from the squeezebox.
 
Cheers

Julian.

-----Original Message-----
From: Phil Karn [mailto:[EMAIL PROTECTED] 
Sent: 01 March 2005 12:38
To: Slim Devices Discussion
Subject: Re: [slim] Modifying squeezebox clock

Triode wrote:
> For some science, try the following:
> 
>
http://www.essex.ac.uk/ese/research/audio_lab/malcolmspubdocs/C41%20SPDIF%20
interface%20flawed.pdf 

Okay, I've read it. And I'm still not convinced.

Although his math looks fairly solid, he makes a lot of questionable 
assumptions that lead to very questionable conclusions. But he also 
makes some interesting observations.

The first observation is that significant (i.e., measurable, though not 
necessarily audible) jitter appears only when a composite S/P-DIF signal 
is severely bandpass filtered. Indeed, his entire paper is all about the 
jitter caused by such filtering. This surprised me, as I had initially 
assumed that people were complaining about the jitter from timebase 
oscillators.

That too-tight filtering can generate jitter is no surprise to me at 
all. I'm a digital communications engineer with experience in modem 
design, and we call this very well known and thoroughly understood 
phenomenon "inter-symbol interference". (See the Nyquist Sampling 
Theorem for the math details.) Intersymbol interference is something we 
take great pains to avoid, as it can, if severe, impair the 
bit-error-rate performance of the modem even at high signal-to-noise ratios.

But here, the author concedes that the jitter isn't so bad as to cause 
any bit errors. The *only* problem that remains has to do with jitter in 
the recovered clock stream.

By now it should be obvious that if the only significant source of 
jitter is bandpass filtering of the S/P-DIF channel, then there is 
absolutely no point in replacing the timebase oscillator in a DAC in an 
attempt to reduce it. The timebase simply isn't the problem; the 
narrowband channel is the problem. Even the cheapest and worst crystal 
oscillators have very low phase noise; when you buy a more expensive 
crystal, you're mainly buying improved frequency accuracy and long term 
frequency stability, not lower phase noise. If the incoming S/P-DIF 
signal has a lot of jitter due to tight filtering, then even an 
absolutely noise-free local oscillator would be forced, by the error 
feedback signal in the clock recovery PLL, to reproduce this jitter at 
its output.

This reinforces the comment I made yesterday that if you're truly 
concerned about jitter, then the very last thing you want to do is to 
attach an external DAC to your Squeezebox. Just use its internal DAC and 
you'll get an analog signal with virtually no jitter because there's no 
S/P-DIF link and no PLL anywhere in the path. Just a DAC clocked 
directly by a crystal, playing out audio data at its own rate. You can't 
get better than that.

But back to S/P-DIF. The obvious and preferred solution to the S/P-DIF 
jitter "problem" -- if it's even a real problem -- is to simply avoid 
transmitting S/P-DIF signals over bandwidth-constrained channels in the 
first place. While this may be difficult in some professional 
applications where you have to go long distances, e.g., hundreds of 
meters, it ought to be easy in most consumer applications where you're 
only going a meter or so. Especially if the link is optical, as it often is.

If that's not possible, then I agree with the recommendations in the 
paper: tighten the loop bandwidth of the clock recovery PLL in the 
receiver so while it will continue to track the incoming clock, it won't 
attempt to track the faster jitter components. (Another way to look at 
this is that the local reference oscillator won't be forced to follow 
the higher frequency components in the incoming jitter.) As the paper 
correctly points out, this may slow lock-up or prevent it entirely if 
there is an excessive frequency offset between the incoming and local 
reference clocks, but a two-step acquire/track PLL can fix this.

But I'm still totally unconvinced there even *is* a jitter "problem" in 
the first place. His analyzes tend to assume absolute worst case, and 
even so the effect of jitter tends to be right at the level of the 
quantization noise in a 16-bit system. Considering that the vast 
majority of real-world audio sources, even the very best ones, have a 
dynamic range considerably worse than 16 bits, most people would 
conclude that jitter is simply not a problem worth fixing with hundreds 
of dollars of fancy accessories. And if anyone believes otherwise, all 
they have to do is to prove it with carefully controlled studies. 
Glowing testimonials and anecdotes from audiophiles won't do.

--Phil





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