Circuit and method for the adaptive suppression of noise

Electrical audio signal processing systems and devices – Noise or distortion suppression

Reexamination Certificate

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C381S071110

Reexamination Certificate

active

06928171

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. 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 (s1, s2) compared to that of the individual microphone signals (d1(t), d2(t)) can be significantly increased. Preferably, one of the two improved output signals (s1, s2) 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 (y1, y2; s1, s2), and solely the transformed signals are utilized for the updating of the filter coefficients (w1, w2). This makes possible a rapidly reacting, and nonetheless calculation-efficient updating of the filter coefficients (w1, w2), and in contrast to other methods only causes minimal audible distortions.

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