Clocking a sample and hold at audio rates -- Why?
2005-12-30 by Richard Brewster
Someone mentioned the ability to clock the sample & hold at audio rates. People have talked about this with the CGS Analog Shift Register too, and I have tried it. If the input is a sine wave and the clock and input are both audio, what you get is a chopped up waveform on the outputs. The clock is heavily mixed into the output. This is to be expected. I seem to recall that people thought it would be some sort of analog delay, where the output would be a replica of the input, but delayed by one, two, or three clock cycles. This does not seem correct to me, because of the clock feed-through. A true analog delay, such as the Blacet Time Machine, goes to great lengths to remove the clock from the output. This is done by using an ultrasonic clock, which requires many stages (1024 is typical) to produce a long signal delay. The clock signal is removed by a lowpass filter at 15Khz or so. I do not see how clocking a sample and hold at audio rates can begin to simulate this. As I said, I tried it with the CGS ASR. The result sounded a lot like what you get by using an audio frequency square wave on the VC input of a VCA. It is a ring-modulator-like product. Could be useful, but it isn't a delay. Anyone have other thoughts on why you would clock a sample and hold at audio frequencies? -Richard Brewster