A proof-of-principle, digital signal processing system is described which can perform deconvolution of audio-bandwidth signals in real time, enabling separation and precise measurement of pulses smeared by a given impulse response. The system operates by convolving a time-domain expression of an inverse filter with the original signal to generate a processed output. It incorporates a high-level user interface for the design of the inverse filter, a communications system and a purpose-designed digital signal processing environment employing a Motorola DSP56002 device. The user interface is extremely versatile, allowing arbitrary inverse filters to be designed and executed within seconds, using a modified frequency sampling method. Since the inverse filters are realized using a symmetrical finite impulse response, no phase distortion is introduced into the processed signals. A special feature of the design is the manner in which the software and hardware components have been organized as an intelligent system, obviating on the part of the user a detailed knowledge of filter design theory or any abilities in processor architecture and assembly code programming. At the present time, the system is capable of deconvolving signals sampled up to 48 kHz. It is therefore ideally suited for real-time audio enhancement, for example, in telephony, public address and long-range broadcast systems, and in compensating for building or room acoustics. Recent advances in DSP technology will enable the same system structure to be applied to signals sampled at frequencies ten times this rate and beyond. This will allow the real-time deconvolution of low-frequency ultrasonic signals used in the inspection and imaging of heterogeneous media.