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.

[1]  Richard W. Harris,et al.  A variable step (VS) adaptive filter algorithm , 1986, IEEE Trans. Acoust. Speech Signal Process..

[2]  Earl R. Ferrara,et al.  Frequency-domain implementations of periodically time-varying filters , 1985, IEEE Trans. Acoust. Speech Signal Process..

[3]  Richard W. Christiansen,et al.  Noise reduction in speech using adaptive filtering I: Signal processing algorithms , 1982 .

[4]  M. Morf,et al.  Recursive least squares ladder estimation algorithms , 1981 .

[5]  M. Morf,et al.  Recursive least squares ladder estimation algorithms , 1981 .

[6]  T. Kailath,et al.  Fast, recursive-least-squares transversal filters for adaptive filtering , 1984 .

[7]  J. Mazo,et al.  On the design of gradient algorithms for digitally implemented adaptive filters , 1973 .

[8]  Steven F. Boll.  Adaptive noise cancelling in speech using the short-time transform , 1980, ICASSP.

[9]  B. Widrow,et al.  Adaptive filtering in the frequency domain , 1978, Proceedings of the IEEE.

[10]  Fuyun Ling,et al.  Numerically robust least-squares lattice-ladder algorithms with direct updating of the reflection coefficients , 1986, IEEE Trans. Acoust. Speech Signal Process..

[11]  A. Gray,et al.  Unconstrained frequency-domain adaptive filter , 1982 .

[12]  H. Kotera,et al.  Application of Fast Kalman Estimation to Adaptive Equalization , 1978 .

[13]  S. Mitra,et al.  A unified approach to time- and frequency-domain realization of FIR adaptive digital filters , 1983 .

[14]  Steven F. Boll.  A spectral subtraction algorithm for suppression of acoustic noise in speech , 1979, ICASSP.

[15]  A. Peterson,et al.  Transform domain LMS algorithm , 1983 .

[16]  H. W. Bode,et al.  A Simplified Derivation of Linear Least Square Smoothing and Prediction Theory , 1950, Proceedings of the IRE.

[17]  E H Satorius,et al.  A Least Squares Adaptive Lattice Equalizer Algorithm. , 1980 .

[18]  Richard W. Christiansen,et al.  Application of the LMS adaptive filter to improve speech communication in the presence of noise , 1982, ICASSP.

[19]  B. Widrow,et al.  Adaptive noise cancelling: Principles and applications , 1975 .

[20]  S. Kakutani.  Review: Nobert Wiener, Extrapolation, interpolation, and smoothing of stationary time series with engineering applications , 1950 .

[21]  F. K. Becker,et al.  Automatic equalization for digital communication , 1965 .

[22]  A.V. Oppenheim,et al.  Enhancement and bandwidth compression of noisy speech , 1979, Proceedings of the IEEE.

[23]  Jae S. Lim,et al.  Speech enhancement , 1986, ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing.