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.

Microphone-array hearing aids with binaural output. II. A two-microphone adaptive system

A desired signal corrupted by additive noise can often be recovered by an adaptive noise canceller using the least mean squares (LMS) algorithm. A major disadvantage of the LMS algorithm is its excess mean-squared error, or misadjustment, which increases linearly with the desired signal power, This leads to degrading performance when the desired signal exhibits large power fluctuations and is a serious problem in many speech processing applications. This work considers two modified LMS algorithms, the weighted sum and sum methods, designed to solve this problem by reducing the size of the steps in the weight update equation when the desired signal is strong. The weighted sum method is derived from an optimal method (also developed in this work), which is not generally applicable because it requires quantities unavailable in a practical system. The previously proposed, but ad hoc, sum method is analyzed and compared to the weighted sum method. Analysis of the two modified LMS algorithms indicates that either one provides substantial improvements in the presence of strong desired signals and similar performance in the presence of weak desired signals, relative to the unmodified LMS algorithm. Computer simulations with both uncorrelated Gaussian noise and speech signals confirm the results of the analysis and demonstrate the effectiveness of the modified algorithms. The modified LMS algorithms are particularly suited for signals (such as speech) that exhibit large fluctuations in short-time power levels.

Modified LMS algorithms for speech processing with an adaptive noise canceller

A system and method for processing an acoustic input signal and providing at least one output acoustic signal to a user of a hearing-aid system. The hearing-aid system includes first and second channels with one of the channels having an adaptive delay. The first channel includes a directional unit for receiving the acoustic input signal and providing a directional signal; a correlative unit for receiving the directional signal and providing a noise reduced signal by utilizing correlative measures for identifying a speech signal of interest in the directional signal; and, a compensator for receiving the noise reduced signal and providing a compensated signal for compensating for a hearing loss of the user.

Binaural adaptive hearing aid

Microphone-array hearing aids provide a promising solution to the problems encountered by hearing-impaired persons when listening to speech in the presence of background noise. This chapter first discusses implementation issues and performance metrics specific to the hearing-aid application. A review of previous work on microphone-array hearing aids includes systems with directional microphones, fixed beamformers, adaptive beamformers, physiologically-motivated processing, and binaural outputs. Recent simulation results of one promising adaptive beamforming system are presented. The performance of microphone-array hearing aids depends heavily on the acoustic environments in which they are used. Additional information about the level of reverberation, number of interferers, and relative levels of interferers encountered by hearing-aid users in everyday situations is required to quantify the benefit of microphone-array hearing aids and to select the optimal processing strategy.

Microphone-Array Hearing Aids

In recent years, the established link between the various human communication production domains has become more widely utilised in the field of speech processing. In this work, we build on previous work by the authors and present a novel two-stage audiovisual speech enhancement system, making use of audio-only beamforming, automatic lip tracking, and pre-processing with visually derived Wiener speech filtering. Initial results have demonstrated that this two-stage multimodal speech enhancement approach can produce positive results with noisy speech mixtures that conventional audio-only beamforming would struggle to cope with, such as in very noisy environments with a very low signal to noise ratio, and when the type of noise is difficult for audio-only beamforming to process.

Novel Two-Stage Audiovisual Speech Filtering in Noisy Environments

: A common complaint of hearing aid users is the difficulty encountered when listening to a talker in a noisy environment. Conventional hearing aids amplify all sounds without discriminating between the desired source (target) and background noises (jammers). These devices increase the overall sound levels, but do nothing to improve target-to-jammer ratio (TJR). Research on microphone array hearing aids is motivated by the lack of success of single-microphone systems, as well as the documented advantages of binaural hearing and multiple-element sensing systems. Array processing can be classified as either fixed (time invariant) or adaptive (time varying). Previous work on microphone array hearing aids has demonstrated that under certain conditions, adaptive arrays can provide significantly better performance than simpler fixed arrays. The benefit of adaptive systems is realized when the input TJR is low and when the signals arriving via direct paths are stronger than the reflections. This benefit is reduced or eliminated at high TJR or in strong reverberation. This work studies modified adaptive algorithms to improve performance at high TJR and in reverberation; it also provides complete specifications for the design of an adaptive microphone array hearing aid. In particular, two previously proposed ad hoc methods for controlling adaptation at high TJR are analyzed and evaluated. The results confirm the usefulness of these methods and provide gnidelines for selecting relevant parameters in anechoic and reverberant environments. In addition, an analysis of the specific causes of target cancellation in reverberation reveals that a simple set of parameter choices can solve this problem. Computer simulations of the complete system demonstrate its benefits in a variety of acoustic environments. Steady-state results show that the system provides very large improvements i

Julie Elise Greenberg

Papers

Improved design of microphone-array hearing aids