Abstract: 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 (d1(t), d2(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 (w1, w2). The filter coefficients (w1, w2) are calculated by a stochastic gradient procedure and updated in real time while minimizing a quadratic cost function consisting of cross-correlation terms.

Abstract: A circuit for adaptive suppression of acoustic feedback forms part of a digital hearing aid, comprising a microphone (1), subtracter (3), hearing correcting means (4), receiver (6), delay element (9), filter (10), updating unit (11), lattice decorrelators (12, 13) and control unit (14). The transmission path is modeled with the feedback characteristic (7) and an adder (8). First decorrelator (12) decorrelates the echo-compensated input signal (en) and second decorrelator (13) decorrelates the delayed output signal (xn) by using coefficients (kn) from first decorrelator (12). The coefficients (kn) of the two filters (12, 13) are calculated by adaptive decorrelation of the echo-compensated input signal (en). This permit maximum convergence rates for minimum distortions. Updating of the filter coefficients mainly takes place where the greatest amplifications occur in the hearing correcting means (4). The fed-back signal components are continuously removed from the input signal.

Abstract: With the method acoustic signals, e.g. in hearing aids, are processed in loudness-controlled manner in such a way that the loudness subjectively received by the hearing impaired person again always corresponds to the loudness received by listeners with normal hearing. Signal processing takes place without Fourier transformation and without subdivision of the signal into subband signals in iterative manner and completely in the time domain. This eliminates the disadvantage of unacceptably long signal delay times of known methods and permits a practical use. The apparatus for performing the method contains a processing stage (4) for the iterative calculation of a loudness-characteristic control quantity (&psgr;) and a correcting filter stage (7) controlled in time-dependent manner therewith.

Abstract: The circuit for the adaptive suppression of noise is a component part of a digital hearing aid, consisting 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 the adaptive suppression of noise can be implemented with the indicated circuit. The two microphones (1, 2), dependent on their spatial arrangement or their directional characteristics and dependent on the location of the acoustic signal sources, provide two differing electric signals (d1(t), d2(t)), which are digitalized in the two AD-converters (3, 4) and pre-processed together with the two fixed compensation filters (5, 6). Following subsequently are the two filters (17, 18) arranged symmetrically crosswise in forward direction with the adaptive filter coefficients (w1, w2).