Well said, Bob. Within the scope of this forum, there's really nothing to add. Rainer -----Ursprüngliche Nachricht----- Von: Bob Amstadt [mailto:firstname.lastname@example.org] 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.