Publications-Detail

Speech Dereverberation for Hearing Aids with a Binaural Data Link

Authors:
Löllmann, H. W.Jeub, M.Vary, P.
Book Title:
Diskussionssitzung "Binaurales Hören mit Hörgeräten und Cochlearimplantaten" im Rahmen der 3 Ländertagung der ÖGMP, DGMP und SGSMP
Organization:
DGMP
Date:
Sep. 2011
Note:

www.medphyswien2011.org

Language:
English

Abstract

Persons with hearing aids often experience a substantial loss of listening comfort or speech intelligibility, if they communicate with other persons in noisy or reverberant environments.
Therefore, speech enhancement algorithms are a common feature of modern hearing aids. A
recent trend is thereby to establish a binaural data link between both hearing aid devices in
order to improve the enhancement of acoustically distorted speech signals. While most
algorithms for binaural speech enhancement aim for the enhancement of noisy speech, the
enhancement of reverberant speech is still in an early stage.
In this contribution, we present a two-stage approach for binaural speech dereverberation in
hearing aids, which achieves a suppression of early and late reverberant speech. An important
feature is that the overall binaural processing does not affect the most important binaural cues,
i.e., the interaural time difference (ITD) and the interaural level difference (ILD). This is
important for speech enhancement in hearing aids to preserve the ability for source
localization in the azimuth plane.
The first stage of the algorithm is based on a spectral subtraction where the weights depend on
the spectral variance of the late reverberant speech. The computation of this spectral variance
in turn requires an estimate of the reverberation time. This is accomplished by an efficient
algorithm, which is based on a maximum likelihood (ML) estimation. The output of the first
stage is further enhanced by a two-channel Wiener filter in a second step. This filter is derived
by a coherence model which takes the shadowing effects of the head into account.
The presented algorithms works blindly, i.e., all needed quantities are estimated from the
reverberant speech so that no a priori information about the acoustical environment is
required. Experiments have shown that the proposed system achieves a significant reduction
of early and late reverberation.

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