4aEA6. DSP algorithms for hearing aids.

Session: Thursday Morning, December 5

Time: 9:15


Author: Julie E. Greenberg
Location: MIT Res. Lab. of Electron., Rm. 36-761, Cambridge, MA 02139
Author: Patrick M. Zurek
Location: MIT Res. Lab. of Electron., Rm. 36-761, Cambridge, MA 02139
Author: William M. Rabinowitz
Location: MIT Res. Lab. of Electron., Rm. 36-761, Cambridge, MA 02139

Abstract:

Advances in semiconductor technology are making digital signal-processing (DSP) hearing aids a viable alternative to conventional analog devices. Some of the most promising DSP algorithms for hearing aids are found in the areas of (1) microphone-array processing for interference reduction, (2) feedback reduction for increased gain margin and stability, and (3) automatic gain control for improved audibility and comfort. Typical common elements of such algorithms are a tapped-delay line, which is used to store a sequence of input values sampled over time, and a time-varying response, which is computed based on the input values. For example, adaptive filters are required by feedback reduction techniques that estimate the feedback path, as well as by adaptive microphone arrays, which attenuate spatially distinct interference sources. Additional benefits in DSP hearing aids are expected from their ability to use sophisticated fitting methods, particularly when the fitting procedure includes selection of algorithm-specific parameter values as well as the frequency gain characteristic. Finally, DSP hardware platforms designed for the hearing aid application provide a powerful research tool, facilitating field trials designed to evaluate algorithms and determine optimal parameter settings. In particular, such devices allow side-by-side comparison of multiple algorithms, including both DSP algorithms and those that could ultimately be implemented using analog components. [Work supported by NIH.]


ASA 132nd meeting - Hawaii, December 1996