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Open Access

Low Delay Noise Reduction and Dereverberation for Hearing Aids

  • Heinrich W. Löllmann (EURASIP Member)1Email author and
  • Peter Vary1
EURASIP Journal on Advances in Signal Processing20092009:437807

https://doi.org/10.1155/2009/437807

Received: 11 December 2008

Accepted: 16 March 2009

Published: 28 April 2009

Abstract

A new system for single-channel speech enhancement is proposed which achieves a joint suppression of late reverberant speech and background noise with a low signal delay and low computational complexity. It is based on a generalized spectral subtraction rule which depends on the variances of the late reverberant speech and background noise. The calculation of the spectral variances of the late reverberant speech requires an estimate of the reverberation time (RT) which is accomplished by a maximum likelihood (ML) approach. The enhancement with this blind RT estimation achieves almost the same speech quality as by using the actual RT. In comparison to commonly used post-filters in hearing aids which only perform a noise reduction, a significantly better objective and subjective speech quality is achieved. The proposed system performs time-domain filtering with coefficients adapted in the non-uniform (Bark-scaled) frequency-domain. This allows to achieve a high speech quality with low signal delay which is important for speech enhancement in hearing aids or related applications such as hands-free communication systems.

Keywords

Noise ReductionSpeech QualitySpeech EnhancementSpectral SubtractionReverberation Time

Publisher note

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Authors’ Affiliations

(1)
Institute of Communication Systems and Data Processing, RWTH Aachen University, Aachen, Germany

Copyright

© H.W. Löllmann and P. Vary. 2009

This article is published under license to BioMed Central Ltd. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.

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