A proof-of-principle, digital signal processing system is described which c
an perform deconvolution of audio-bandwidth signals in real time, enabling
separation and precise measurement of pulses smeared by a given impulse res
ponse. The system operates by convolving a time-domain expression of an inv
erse filter with the original signal to generate a processed output. It inc
orporates a high-level user interface for the design of the inverse filter,
a communications system and a purpose-designed digital signal processing e
nvironment employing a Motorola DSP56002 device. The user interface is extr
emely versatile, allowing arbitrary inverse filters to be designed and exec
uted within seconds, using a modified frequency sampling method. Since the
inverse filters are realized using a symmetrical finite impulse response, n
o phase distortion is introduced into the processed signals. A special feat
ure of the design is the manner in which the software and hardware componen
ts have been organized as an intelligent system, obviating on the part of t
he user a detailed knowledge of filter design theory or any abilities in pr
ocessor architecture and assembly code programming. At the present time, th
e system is capable of deconvolving signals sampled up to 48 kHz. It is the
refore ideally suited for real-time audio enhancement, for example, in tele
phony, public address and long-range broadcast systems, and in compensating
for building or room acoustics. Recent advances in DSP technology will ena
ble the same system structure to be applied to signals sampled at frequenci
es ten times this rate and beyond. This will allow the real-time deconvolut
ion of low-frequency ultrasonic signals used in the inspection and imaging
of heterogeneous media.