From: lasner@watsun.cc.columbia.edu (Charles Lasner)
Subject: Re: oversampling (was Re: using green marker on CDs)
Reply-To: lasner@watsun.cc.columbia.edu (Charles Lasner)
Organization: Columbia University
Date: Thu, 1 Oct 1992 09:03:08 GMT

In article <HOD.92Sep28161508@cucbs.chem.columbia.edu> hod@cucbs.chem.columbia.edu (Hod Greeley) writes:

>This is a tangential question, but if the error rate is so low, what
>purpose does oversampling serve?  Is this more industry hype?  If,
>say, an 8x pass over the data is required to achieve this level of
>accuracy, that would suggest to me that the error rate must be much
>higher than you claim.  Am I missing something?  If the rate of
>uncorrectable errors is as low as you suggest, it would seem
>superfluous.  Is oversampling not what I think it is?
>
>Hod

I know of two forms of oversampling; someone else will have to explain the
applicability of these known methods to CD's; I don't claim to be an expert
on the thing at this level:

In the world of NRZ data such as RS-232 communications as we all use, the
bits occur at a nominal rate (2400, etc.) but are actually oversampled, 
typically 16x that rate.  The idea is that the bit change may or may not
occur, so we always want to determine the center-time best suited to
looking for a potential change.  If adjacent bits don't change, we run the
risk of getting out of time sync with them, especially if they jitter for
any reason.  Of course, once a transition occurs, we now have a reference
point.  Oversampling therefore gives us the opportunity to continually
adapt to any jitter by constantly correcting the predicted event time for
where the next transition ought to occur, thus limiting the effect of the
jitter.  I would assume that some similar effect applies for CD's basic
bit stream to overcome any motor jitter of the drive, or the original
motor that created the CD data (which in the worst case are additive effects).

Once you have a stream of sample bitstrings established, the basic clock
is that there is a 16-bit update 44.1 K times/second.  But you can define
a higher-order function that is happening Nx that rate, say 16x the rate.

The input to that function is the available data, but as a consequence of
its basic higher-frequency rate, the rate of change of the output of the
device is frequency-limited.  A digital system induces step-wise changes
which have a raucous audio spectrum of components that have to be filtered
out by a filter that is also desired-frequency linear and also hopefully
phase-linear in that pass-band as well.  But such "well-behaved" filters
for the frequencies desired, isn't all that great a filter at removing the
higher-frequency "mess" caused by the inherent switching of the sampling
method.  Thus, anything that frequency-limits the change in output means
that post-filtering need not be as severe.  An alternate view is that
oversampling allows a built-in digital filter as part of the overall
removal process for these undesired high-frequencies.  In any case, the
signal can no longer jump at 44.1 KHz generating all this noise, because
even such a jump is now converted to a lot of little "steps" at the higher
rate.  Thus the junk is at more of an even higher frequency, which is
easier to get rid of, and also at a lower-amplitude at the same time.

Units that oversample definitely do the latter.  Someone else can post if
there is anything applicable to the ideas in the former.  I assume that
both could be done, etc.

cjl