scispace - formally typeset
Journal ArticleDOI

A microphone array for hearing aids

Bernard Widrow
- 01 Jan 2001 - 
- Vol. 1, Iss: 2, pp 26-32
Reads0
Chats0
TLDR
In this article, the authors proposed a directional acoustic receiving system, which consists of two or more microphones mounted on a housing supported on the chest of a user by a conducting loop encircling the user's neck.
Abstract
A directional acoustic receiving system is a form of a necklace including an array of two or more microphones mounted on a housing supported on the chest of a user by a conducting loop encircling the user's neck. Signal processing electronics contained in the same housing receive and combine the microphone signals in such a manner as to provide an amplified output signal which emphasizes sounds of interest arriving in a direction forward of the user. The amplified output signal drives the supporting conducting loop to produce a representative magnetic field. An electroacoustic transducer including a magnetic field pick up coil for receiving the magnetic field is mounted in or on the user's ear and generates an acoustic signal representative of the sounds of interest. The microphone output signals are weighted (scaled) and combined to achieve desired spatial directivity responses. The weighting coefficients are determined by an optimization process. By bandpass filtering the weighted microphone signals, with a set of filters covering the audio frequency range, and summing the filtered signals, a receiving microphone array with a small aperture size is caused to have a directivity pattern that is essentially uniform over frequency in two or three dimensions. This method enables the design of highly-directive-hearing instruments which are comfortable, inconspicuous, and convenient to use. The array provides the user with a dramatic improvement in speech perception over existing hearing aid designs, particularly in the presence of background noise, reverberation, and feedback.

read more

Citations
More filters
Proceedings ArticleDOI

Speech segregation based on sound localization

TL;DR: A technique for speech segregation based on sound localization cues by observing that systematic changes of the interaural time differences and intensity differences occur as the energy ratio of the original signals is modified is explored.
Journal ArticleDOI

Blind estimation of reverberation time.

TL;DR: A method for estimating RT without prior knowledge of sound sources or room geometry is presented, and results obtained for simulated and real room data are in good agreement with the real RT values.
Journal ArticleDOI

Combinations of Adaptive Filters: Performance and convergence properties

TL;DR: Adaptive filters are at the core of many signal processing applications, ranging from acoustic noise supression to echo cancelation to array beamforming.
Book ChapterDOI

Adaptive Beamforming for Audio Signal Acquisition

TL;DR: From this, a robust generalized sidelobe canceller (GSC) results as an attractive solution for practical audio acquisition systems and the general theoretical framework leads to new insights for the GSC behavior in complex practical situations.
Journal ArticleDOI

Rate-Constrained Collaborative Noise Reduction for Wireless Hearing Aids

TL;DR: This work investigates the noise reduction capability of hearing instruments that may exchange data by means of a rate-constrained wireless link and thus benefit from the signals recorded at both ears of the user under two different coding strategies.
Related Papers (5)