A digital hearing aid that compensates loudness for impaired listeners and its performance

Abstract

A method of designing a digital hearing aid for speech signal suitable for sensorineural hearing loss with a narrow audible range is proposed. First, the input signal is divided into short time blocks (8 ms) and its spectrum is analyzed using short-time FFT. Then the frequency-gain characteristic of the digital filter for the block is determined using loudness compensation functions that describe the relation between the loudness of normal listeners and that of impaired listeners, so that the input signal within the block is projected on to the dynamic range of the subject. This algorithm is realized using a digital signal processor (DSP). The results of monosyllabic intelligibility tests to evaluate the performance of the system for six sensorineural impaired listeners are also shown. [Work supported by Grant-in-Aid for Scientific Research from the Ministry of Education and the Ministry of Public Welfare.]