🚀 go-pugleaf

The old style electronic delays used two open-reel tape recorders mounted
side by side so that a large tape path could be made from one to the other.
The left-most machine was set to record while the righthand machine was
in the play mode.  The distance between the two machines was such that it
took x number of seconds for the tape to get from the record head on the
recorder to the playback head on the player.
     Modern audio delays use digital electronics and, therefore, are much
neater in their operation than the old electromechanical systems.  They 
have an analog to digital converter as the encoder anda digital to analog
converter as the decoder.  Of course, the encoder and decoder have low-pass
filters to satisfy the Niquist requirements of the system.  Between the
encoder and decoder is a large memory buffer which has enough storage to
hold a given number of seconds of audio.  The length of the delay is limited
by the number of bytes in the buffer.  At a sampling rate of 20,000 samples
per second, the highest audio frequency would be 10Khz and a 7-second delay
would need about 140Kbytes.  Of course, there are compression tricks such as
delta PCM to reduce the amount of memory required, but such delays use 
quite a lot of memory.  The higher the fidelity of the audio needed, the
faster the sampling rate and the more memory needed.  The delay is achieved
by reading each location in the buffer and sending that to the output, then
writing a new sample to that same location.  As the system goes round and 
round, the output is always x seconds behind the input.

Martin McCormick WB5AGZ   Stillwater, OK
O.S.U. Computer Center Data Communications Group