This work is aimed at developing a design for the use of a microphone array with binaural hearing aids. The goal of such a hearing aid is to provide both the spatial-filtering benefits of the array and the natural benefits to sound localization and speech intelligibility that accrue from binaural listening. The present study examines two types of designs for fixed-processing systems: one in which independent arrays provide outputs to the two ears, and another in which the binaural outputs are derived from a single array. For the latter, various methods are used to merge array processing with binaural listening. In one approach, filters are designed to satisfy a frequency-dependent trade between directionality and binaural cue fidelity. In another, the microphone signals are filtered into low- and high-frequency components with the lowpass signals providing binaural cues and the highpass signal being the single output of the array processor. Acoustic and behavioral measurements were made in an anechoic chamber and in a moderately reverberant room to evaluate example systems. Theoretical performance was calculated for model arrays mounted on an idealized spherical head. Results show that both single- and dual-array systems provided target-intelligibility enhancements (2-4 dB improvements in speech reception threshold) relative to binaural cardioid microphones. In addition, the binaural-output systems provided cues that assist in sound localization, with resulting performance depending directly upon the cue fidelity. Finally, the sphere-based calculations accurately reflected the major features of the actual head-mounted array results, both in terms of directional sensitivity and output binaural cues.

For pt.I see ibid., vol.5, no.6, p.529-42, 1997. This work is aimed at developing a design for the use of a microphone array with binaural hearing aids. The goal of such a hearing aid is to provide both the spatial-filtering benefits of the array and the natural benefits to sound localization ability and speech intelligibility that result from binaural listening. The present study examines a design in which two ear-level omnidirectional microphones constitute the array. The merging of array processing with binaural listening is accomplished by dividing the frequency spectrum, devoting the lowpass part to binaural processing and the highpass part to adaptive array processing. Acoustic and behavioral measurements were made in an anechoic chamber and in a moderately reverberant room to assess the trade-off between sound localization and speech reception as the cutoff frequency was varied. A lowpass/highpass cutoff frequency of 500 Hz provided an improvement of 40 percentage points in sentence intelligibility over unaided listening for normal-hearing listeners, while still allowing adequate localization performance. Comparison of this binaural adaptive system to traditional amplification configurations with normal-hearing listeners showed improvements in speech reception in noise in a mildly reverberant room of approximately 3 dB over simple binaural amplification and 5 dB over monaural amplification.

Five techniques for reducing acoustic feedback in hearing aids were investigated: an adaptive notch filter, three previously described methods for adaptive feedback cancellation, and a novel method for feedback cancellation with adaptation during quiet intervals. Through real-time implementations, these techniques were assessed for added stable gain and sound quality. Test results showed the novel system to provide the best overall performance.>

The optimal performance of a two-microphone adaptive-beamforming hearing aid was analyzed under sample conditions. Using a straight-ahead target and a single off-axis jammer, source-to-microphone impulse responses were recorded from head-mounted arrays in anechoic and reverberant environments. Optimal filters (equivalent to the fully converged beamformer's adaptive filter) and associated system performance were computed for various filter structures and input target-to-jammer ratios. Generally, performance is sensitive to variations in filter length and filter primary-channel delay only under conditions of minimal reverberation and low target-to-jammer ratio. With realistic amounts of either reverberation or target, there is little sensitivity to filter length or primary channel delay.>