[time-nuts] Clocks for Audio gear

[Attila Kinali]

On Wed, 09 May 2012 14:25:34 -0700
Hal Murray <hmurray at megapathdsl.net> wrote:

> Back to technical stuff...
> 
> As a practical matter, is clock jitter or phase noise from a typical low cost 
> crystal and decent board layout a significant problem in audio gear?  How 
> hard is it to measure?

Depends on how accurate it should be ;-)

But for audio purposes, the crystal itself has low enough noise. The
Problem is the oscillator circuit and the power supply. That's where
most designs mess up.

 
> Video geeks solved their clock distribution problems by using frame buffers.  
> Is there a similar trick for audio?  Is there a need for it?

Video is a lot less sensitive to jitter. An A/V desync starts to be
noticable from 10ms upwards for most people. Trained people notice
it even below 5ms. I guess that 1ms is beyond what anyone can notice.
But imagine you've a 1ms gap in your audio... People will scream at
the poor quality.

> Part of the problem is that they are doing magic down conversion in the ADC.  
> (I can't think of the term.)

Direct downconversion using sampling :-)

>  Suppose you have a 100 MHz signal with a 1 MHz 
> bandwidth.  You don't have to sample at 200 MHz.  You can sample at 2 MHz and 
> your signal will alias down.  It's turning what is normally a bug into a 
> feature.  

It's not a bug. It's a feature of the sampling process itself.
Think of sampling as multiplication of an comp of dirac pulses
with your input signal. Now remember that mixing is a multiplication
of two signals....And the idea doing that is actually quite old,
but it wasn't feasible with main stream components until a few years back.


			Attila Kinali

-- 
The trouble with you, Shev, is you don't say anything until you've saved
up a whole truckload of damned heavy brick arguments and then you dump
them all out and never look at the bleeding body mangled beneath the heap
		-- Tirin, The Dispossessed, U. Le Guin