| 摘要 |
The circuit for adaptive suppression of noise is a component part of a digital-hearing aid, consists of two microphones ( 1, 2 ), two AD-converters ( 3, 4 ), two compensating filters ( 5, 6 ), two retarding elements ( 7, 8 ), two subtractors ( 9, 10 ), a processing unit ( 11 ), a DA-converter ( 13 ), an earphone ( 15 ) as well as the two filters ( 17, 18 ). The method for adaptive suppression of noise can be implemented with the indicated circuit. The two microphones ( 1, 2 ), provide two differing electric signals (d<SUB>1</SUB>(t), d<SUB>2</SUB>(t)), which are digitalized in the two AD-converters ( 3, 4 ) and pre-processed together with the two fixed compensation filters ( 5, 6 ). Downstream the compensation filters are arranged the two filters ( 17, 18 ) symmetrically crosswise in a forward direction and having adaptive filter coefficients (w<SUB>1</SUB>, w<SUB>2</SUB>). The filter coefficients (w<SUB>1</SUB>, w<SUB>2</SUB>) are calculated by a stochastic gradient procedure and updated in real time while minimizing a quadratic cost function consisting of cross-correlation terms. As a result of this, spectral differences of the input signals are selectively amplified. With a suitable positioning of the microphones ( 1, 2 ) or selection of the directional characteristics, the signal to noise ratio of output signals (s<SUB>1</SUB>, s<SUB>2</SUB>) compared to that of the individual microphone signals (d<SUB>1</SUB>(t), d<SUB>2</SUB>(t)) can be significantly increased. Preferably, one of the two improved output signals (s<SUB>1</SUB>, s<SUB>2</SUB>) within one of the processing units ( 11, 12 ) is subjected to the usual processing specific to hearing aids, sent to one of the DA-converters ( 13, 14 ) and acoustically output once again through one of the earphones ( 15, 16 ). Four additional cross-over element filters ( 19-22 ) carry out a signal-dependent transformation of the input and output signals (y<SUB>1</SUB>, y<SUB>2</SUB>; s<SUB>1</SUB>, s<SUB>2</SUB>), and solely the transformed signals are utilized for the updating of the filter coefficients (w<SUB>1</SUB>, w<SUB>2</SUB>). This makes possible a rapidly reacting, and nonetheless calculation-efficient updating of the filter coefficients (w<SUB>1</SUB>, w<SUB>2</SUB>), and in contrast to other methods only causes minimal audible distortions.
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