On Fri, Dec 30, 2005 at 12:45:48PM +1000, Adam Nielsen wrote:
> I've always wondered about this. Fair enough that two cards would play
> back sound at slightly different speeds, but why can't you just drop a
> sample or two every few milliseconds or so, to keep the sound roughly in
> sync?
How do you know how much to slew the playback by?
Say I have two sound cards in my box. Each sound card has an
independent crystal that generates a clock for timekeeping. The
crystals are rated to resonate at some fixed frequency, but there will
always be a margain of error. Even two "identical" parts from the
same manufacturer will produce different errors in their clock.
At first, the issue isn't a big deal, since the difference may be
small. But as time ticks by, many small differences begin to add up
to bigger and bigger difference. This type of error is called drift.
In addition, a crystal is a physical process, and so occasionally
there are abberant ticks. This causes sudden loss of time tracking.
This type of error is called jitter.
The two together are a deadly combination. You need some way of
mitigating both to keep stuff in sync. One way is to use some
out-of-band data to slave one card to the other card's timesource.
Lots of pro gear does this with word clock, spdif links, and the like.
You can solve the problem for your PC's internal clock with NTP.
Moreover, if you're interested in timekeeping issues like this, the
NTP research papers are an amazing source of a huge amount of
information on this kind of stuff.
--
Ross Vandegrift
[EMAIL PROTECTED]
"The good Christian should beware of mathematicians, and all those who
make empty prophecies. The danger already exists that the mathematicians
have made a covenant with the devil to darken the spirit and to confine
man in the bonds of Hell."
--St. Augustine, De Genesi ad Litteram, Book II, xviii, 37
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