Wisconsin Alumni Research Foundation

Nicolet Instrument Corporation

Malcolm J Williamson

Kurt E Hecox

Kenneth L Cummins

Diane K Bustamante

Acoustic feedback in digital signal processing hearing aids is suppressed by using signal processing techniques in the digital processor. A first processing technique causes the data to the main signal processing path in the digital signal processor to be delayed by varying amounts over time, preferably in a periodic manner, to disrupt the buildup of feedback resonances. In a second technique, a digital filter receives the input data and has its coefficients adjusted so that the output of the filter is substantially an optimal estimate of the current input sample based on past input samples. The output of the filter is then subtracted from the input signal data to provide difference signal data which substantially cancels out the resonant frequencies. In a third technique, the acoustic feedback path from the output to the input of the hearing aid is modeled in the digital signal processor as a delay and a linear filter. The output of the main signal processing path in the digital signal processor is delayed and the delayed data passed through the linear filter, with the output of the filter then being substracted from the input signal data to provide difference signal data which is provided to the main signal processing path. The coefficients of the digital filter in the feedback path are adjusted so that the signal passed through the feedback filter substantially corresponds to of the acoustic feedback signal to thereby cancel the same.

Feedback suppression in digital signal processing hearing aids

A hearing aid system utilizes digital signal processing to correct for the hearing deficit of a particular user and to maximize the intelligibility of the desired audio signal relative to noise. An analog signal from a microphone is converted to digital data which is operated on by a digital signal processor, with the output of the digital signal processor being converted back to an analog signal which is amplified and provided to the user. The digital signal processor includes a time varying spectral filter having filter coefficients which can be varied on a quasi-real time basis to spectrally shape the signal to match the hearing deficit of the user and to accommodate ambient signal and noise levels. The coefficients of the spectral filter are determined by estimating the energy in several frequency bands within the frequency range of the input signal, and using those energy estimates to calculate desired gains for the frequency bands and corresponding spectral filter coefficients. The spectral energy analysis may be carried out using pairs of high pass and low pass digital filters in cascade relation, with the output of each low pass filter being provided to the next pair of high pass and low pass filters. The rate at which output data is provided from the filters in each pair may be reduced from the sample rate of input data by one half for succeeding pairs of filters in the cascade to thereby reduce the computation time required.

Adaptive, programmable signal processing and filtering for hearing aids

A hearing aid system utilizing digital signal processing is programmable to fit the hearing deficit of a particular use and adaptive to the sound environment to maximize the intelligibility of the desired audio signal relative to noise. An analog signal picked from a microphone is amplified, filtered and converted to digital data. A digital signal processor preferably performs spectral shaping on the data to match the user's preference and performs a non-linear adaptive amplification function on the digital data. The amplification gain function may include several piecewise linear sections, including a first section providing expansion up to a first knee point, a second section providing linear amplification from the first knee point to a second knee point, and a third section providing compression for signals above the second knee to reduce the effort of over range signals and minimize loudness discomfort to the user. An estimate of the level of background noise is made as a function of the energy envelope of the input signal data, with the noise estimate then being used to adjust the position of the first knee up or down or to change the expansion ratio to reduce the noise component of the amplified signal supplied to the user. The digital signal processor includes a programmable read only memory which contains the desired spectral shaping characteristics and non-linear amplification characteristics suited to the user.

Growing community of inventors

Madison, WI, United States of America

Malcolm J Williamson