AW: Analog to digital conversion - sample rate

["Rainer Thelonius Balthasar Straschill" <rs@moinlabs.de>]

Well said, Bob. Within the scope of this forum, there's really nothing
to add.

        Rainer

-----Ursprüngliche Nachricht-----
Von: Bob Amstadt [mailto:bob@amstadt.com] 
Gesendet: Samstag, 17. Dezember 2005 19:50
An: Loopers-Delight@loopers-delight.com
Betreff: Analog to digital conversion - sample rate


Hi everyone,

I've been trying to stay out of the conversation, but I do want to clear
up 
a few things.  I would like to point out that this topic is the subject
of 
an entire college.  So, it is very difficult to simplify it down to just
a 
couple of paragraphs.  Let me hit the highlights.

A/D converters don't see the signal as a set of sine waves.
Mathematically 
we look at signals as a set of sine waves because it allows us create a 
system of mathematics that does a very good job of describing filters
both 
analog and digital.  You could create a system of mathematics based of 
different frequency square waves, but sine waves result in much simpler 
equations.

The Nyquist rate is a theoretical concept that results from the
theoretical 
mathematics and it indicates to us the maximum frequency that can be 
represented after sampling a signal.  As has been stated, it is
necessary 
to filter a signal before sampling to avoid significant frequency
content 
about the Nyquist rate.  The topic of filters is a huge one.

Higher sampling rates are better, but twice the sampling rate doesn't
mean 
that your sound will be twice as good.  It is a very interesting topic
and 
for those of your interested in it, I highly recommend that you take 
courses or do some experimentation.  It is a fun topic to explore.