Application of adaptive digital signal processing to speech enhancement for the hearing impaired.
A major complaint of individuals with normal hearing and hearing impairments is a reduced ability to understand speech in a noisy environment. This paper describes the concept of adaptive noise cancelling for removing noise from corrupted speech signals. Application of adaptive digital signal processing has long been known and is described from a historical as well as technical perspective. The Widrow-Hoff LMS (least mean square) algorithm developed in 1959 forms the introduction to modern adaptive signal processing. This method uses a "primary" input which consists of the desired speech signal corrupted with noise and a second "reference" signal which is used to estimate the primary noise signal. By subtracting the adaptively filtered estimate of the noise, the desired speech signal is obtained. Recent developments in the field as they relate to noise cancellation are described. These developments include more computationally efficient algorithms as well as algorithms that exhibit improved learning performance. A second method for removing noise from speech, for use when no independent reference for the noise exists, is referred to as single channel noise suppression. Both adaptive and spectral subtraction techniques have been applied to this problem--often with the result of decreased speech intelligibility. Current techniques applied to this problem are described, including signal processing techniques that offer promise in the noise suppression application.
Electrical Engineering Department, Brigham Young University, Provo, Utah 84602.
D M
DM
Chabries
R W
RW
Christiansen
R H
RH
Brey
M S
MS
Robinette
R W
RW
Harris
D000162:Acoustics; D000465:Algorithms; D003638:Deafness / Q000534:rehabilitation*; D006310:Hearing Aids; D006801:Humans; D008838:Microcomputers; D012815:Signal Processing, Computer-Assisted; D012984:Software; D013067:Speech Perception