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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

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