DocumentCode
311409
Title
Multi-microphone sub-band adaptive signal processing for improvement of hearing aid performance: primarily results using normal hearing volunteers
Author
Shields, Paul W. ; Campbell, Douglas R.
Author_Institution
Dept. of Electron. Eng. & Phys., Paisley Univ., UK
Volume
1
fYear
1997
fDate
21-24 Apr 1997
Firstpage
415
Abstract
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
Keywords
acoustic noise; acoustic signal processing; acoustic transducer arrays; adaptive signal processing; array signal processing; direction-of-arrival estimation; handicapped aids; hearing aids; least mean squares methods; microphones; noise abatement; speech intelligibility; speech processing; LMS algorithm; SNR; acoustic environments; acoustic noise data; acoustic speech data; binaural preprocessing; experiments; hearing aid performance; intelligibility tests; least mean squares; linear hearing aid; multimicrophone subband adaptive signal processing; noise cancellation; normal hearing volunteers; signal characteristics; signal to noise ratio; simulated room; speech processing; speech signals; wideband signal; Acoustic testing; Adaptive signal processing; Auditory system; Least squares approximation; Noise cancellation; Signal processing; Signal processing algorithms; Signal to noise ratio; Speech enhancement; Wideband;
fLanguage
English
Publisher
ieee
Conference_Titel
Acoustics, Speech, and Signal Processing, 1997. ICASSP-97., 1997 IEEE International Conference on
Conference_Location
Munich
ISSN
1520-6149
Print_ISBN
0-8186-7919-0
Type
conf
DOI
10.1109/ICASSP.1997.599661
Filename
599661
Link To Document