I'm not aware of the Elsea/Lyon methodologies, but wouldn't it just be [read-buffer]-> [delay/filter network]->[write buffer], then copy contents of write-buffer into read and begin again.

or even a live running tapping buffer.

let me rephrase. "tapping buffer" was somewhat wrong:

- a reverb - a feedback loop around the reverb - a simple delay in the feedbackloop which is as long as the material is

since this will run endless, you have to click the stop button yourself.

Hmmm...

You create two buffers. Loading a source audio to the first buffer. Playing this buffer (I mean, the sound should be audible in the room). In the same time you are recording the sound (mic->adc->[second buffer]). Now you are repeating the procedure, but playing second buffer and overwriting the content of the first buffer. You repeat the procedure again (each time playing the last recorded buffer). With each repetition, the acoustic parameters of the room affect some frequencies.

Staying "inside" computer and using filters... it seems to me to act in the opposite to Lucier's assumptions..

yes, leaving the reverb part out is very economic. just not very "room".

yes of course, i thought you are going to emulate all parts in software. :)

here's a thing:

uses buffir -- best to use a large signal vector size to alleviate cpu

here's the soundfile to use (put it in your search path):