Professor Baker - this is an interesting idea and I say bravo for thinking along such lines in response to a common, practical problem. I think such mucking around in the frequency domain is commonly done in generating/editing audio for certain effects, and in many recording and production circumstances.
The problem is with this idea applied to recording is that the acoustic environment around a horn (the room) and its interaction with the horn and the microphones is fundamentally NOT a linear system. There are all kinds of nonlinear coupling and effects in such a complex acoustic system. Probably to a weak first-order approximation this transfer-function could be made, but I believe your suggestion relies on clean, linear-superposition of the behind and front acoustic signals, which would not be very accurate in a real situation. Check out the literature on acoustic source-separation and you'll find this problem ubiquitous in similar tasks. just my .02 cents. this would not be hard to mock-up in MATLAB sometime and try it out. any other thoughts anyone? david - physics and horn performance student ____________________________________________________________ Nutrition Improve your career health. Click now to study nutrition! http://thirdpartyoffers.juno.com/TGL2131/c?cp=7J1fl7BlM62h3OKqnQEtXwAAJz2tEPjwKKiSDPsz4fU1aV0DAAYAAAAAAAAAAAAAAAAAAADNAAAAAAAAAAAAAAAAAAASQwAAAAA= _______________________________________________ post: [email protected] unsubscribe or set options at https://pegasus.memphis.edu/cgi-bin/mailman/options/horn/archive%40jab.org
