This paper describes a comprehensive, integrated, microcomputer-based facility for research on hearing aid fitting, and for clinical use. The system first applies formal prescription rules to audiometric measures which have been obtained from an individual patient to predict which hearing aid amplification function is likely to provide the optimal benefit for a hearing impaired listener. It then designs and implements a filter in a digital-signal processing (DSP) board, which precisely implements the desired gain function. Speech intelligibility and listener preferences can then be measured to evaluate the prescribed hearing aid gain function using efficient, computer-based testing procedures. The system requires an lBM/AT compatible microcomputer (eg., Zenith Model 386 or 248, with math coprocessor and EGA or VGA graphics), an ARIEL DSPdigital signal processing board, plus filters and associated analog audio equipment, and a Mouse.The precise relation between a hearing-impaired listener's performance on basic audiometric tests and the amplification characteristics which would lead to the highest level of speech understanding for that listener remains an open question. A wide variety of procedures have been proposed to guide hearing aid prescription [I ,2,3,4,5,6,7]. These methods differ in both the form and manner in which audiometric data are obtained, the operational definition of the spectra and levels of conversational speech and the computational rules used to predict the "optimal" amplification characteristics for a give listener.

We describe a general-purpose, programmable system that provides high-quality, low-cost devices for experimentation in psychoacoustics and speech perception. The system is controlled by a host computer (e.g., an IBM PC), over a serial line. Through the use of a high-level, general-purpose experiment control program, the designed interconnection of devices can be specified logically, and the settings of the devices modified dynamically, during the experiment.While the stimulus and experimental-control requirements for speech and hearing research are demanding, they are relatively uniform across experiments and laboratories. One must first specify and control the spectra and levels of the presented stimuli, including the contribution of the overall frequency response of the complete electroacoustic environment to the level of the acoustic coupling to the subject's ear. To ensure that testing occurs under standardized conditions, equipment must be calibrated at the beginning of each experimental session. Then, one must be able to present the desired signals in the sequence and at the times specified.A relatively large number of special-purpose and/or expensive components are required for such research: specialized equipment is typically required to filter (antialias) signals for digital-to-analog conversion, to amplify and/or attenuate signals, and to mix signals from different sources. Typically, the preparation for an experiment involves the manual interconnection of the devices used during the experiment, with the associated delays, as well as opportunities for human error. Moreover, the interconnection of equipment with different grounding and cabling systems tends to introduce noise (e.g., 60 Hz) into the circuitry. 1 Also, when two or more experimental conditions requiring different configurations of equipment are used during a given testing session, such concerns are multiplied.In order to improve the efficiency with which experiments can be set up and controlled while reducing the possibilities for noise and error, researchers have adopted several approaches. The first is to use a connector panel,