Multi-microphone sub-band adaptive signal processing for improvement of hearing aid performance: primarily results using normal hearing volunteers

摘要

A system for the binaural pre-processing of speech signals for input to a standard linear hearing aid has been proposed. The work is based on that of Toner and Campbell (1993) which applied the least mean squares (LMS) algorithm in sub-bands to speech signals from various acoustic environments and signal to noise ratios (SNR). The method attempts to take advantage of the multiple inputs to perform noise cancellation. The use of sub-bands enables a diverse processing mechanism to be employed, where the wide-band signal is split into smaller sub-bands, which can subsequently be processed according to their signal characteristics. The results of a series of intelligibility tests are presented from experiments in which acoustic speech and noise data, generated in a simulated room was tested on normal hearing volunteers.