Richard, Even if your RC filter was a brick wall at 0.5 MHz you would have to keep your sampling rate at a minimum of 1 MHz to prevent any aliasing. That is, after your 44.1 MHz ADC the best you could do is take 1 of every 44 samples in order to have an artifact free signal (and that's with ideal brick wall RC and ideal A/D). Typically the way people do this sort of thing is that they oversample with the A/D in order to relax the restriction on the RC filter. Since you won't have a brickwall at 0.5 MHz you sample faster than 1 MHz to make sure you don't have any aliasing. You then DIGITALLY filter since you can get pretty close to a brick wall filter with an IIR filter (or maybe a couple cascaded). At that point you can then downsample to your desired sampling rate. Brad "Richard Owlett" <rowlett@atlascomm.net> wrote in message news:1109luhj7342pe7@corp.supernews.com...>I read that band limiting the input to an A/D eliminates aliasing. I also >read that using a very high sampling rate makes that filter very simple. > > Suppose I: > 1. filter with a single stage low pass RC filter with 3db pint at .5 MHz > 2. sample at 44.1 MHz > 3. do simple sample rate conversion by saving every thousandth sample > > Would I have a nice artifact free 44.1 kHz digitized signal? > > The question came to mind when reading "Digital Dharma of Audio A/D > Converters" ( http://www.rane.com/note137.html ). Particularly section > quoted below. > > "Shannon studied Nyquist's work closely and came up with a deceptively > simple addition. He observed (and proved) that if you restrict the input > signal's bandwidth to less than one-half the sampling frequency then no > errors due to aliasing are possible. So bandlimiting your input to no more > than one-half the sampling frequency guarantees no aliasing. Cool ... only > it's not possible. > > In order to satisfy the Shannon limit (as it is called -- Harry gets a > "criteria" and Claude gets a "limit") you must have the proverbial > brick-wall, i.e., infinite-slope filter. Well, this isn't going to happen, > not in this universe. You cannot guarantee that there is absolutely no > signal (or noise) greater than the Nyquist frequency. Fortunately there is > a way around this problem. In fact, you go all the way around the problem > and look at it from another direction. > > If you cannot restrict the input bandwidth so aliasing does not occur, > then solve the problem another way: Increase the sampling frequency until > the aliasing products that do occur, do so at ultrasonic frequencies, and > are effectively dealt with by a simple single-pole filter. This is where > the term "oversampling" comes in. For full spectrum audio the minimum > sampling frequency must be 40 kHz, giving you a useable theoretical > bandwidth of 20 kHz -- the limit of normal human hearing. Sampling at > anything significantly higher than 40 kHz is termed oversampling. In just > a few years time, we have seen the audio industry go from the CD system > standard of 44.1 kHz, and the pro audio quasi-standard of 48 kHz, to > 8-times and 16-times oversampling frequencies of around 350 kHz and 700 > kHz respectively. With sampling frequencies this high, aliasing is no > longer an issue."