Abstract: 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.

Abstract: 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.

Abstract: 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.