A low-complexity adaptive speech dereverberation approach based on multichannel linear prediction

The adaptive multichannel linear prediction (AMCLP) technique is one of the most effective methods for speech dereverberation in real-time application systems. However, the computational complexity of the associated AMCLP algorithms is very high. In this paper, we propose a low-complexity speech dereverberation approach based on AMCLP. The coefficient matrix of the multichannel dereverberation filter is decomposed into a linear combination of a set of low-dimensional sub-filter matrices and a set of low-dimensional sub-filter vectors through Kronecker product. According to these sub-filter models, we define two sets of signal models and cost functions, from which the adaptive speech dereverberation algorithm is established. This solution not only leads to effective dereverberation performance, but also reduces the computational cost, which is very conducive to realistic speech systems.

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Publication Details

Journal
NOISE-CON proceedings
Published
2024-10-04
DOI
https://doi.org/10.3397/in_2024_4138
Primary Topic
Speech and Audio Processing
Type
article
Field-Weighted Citation Impact
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article

A low-complexity adaptive speech dereverberation approach based on multichannel linear prediction

Yi Yu, Zhu Zhen, Hongsen He, Zhimin QIU
NOISE-CON proceedings
Speech and Audio Processing
article

A low-complexity adaptive speech dereverberation approach based on multichannel linear prediction

Yi Yu, Zhu Zhen, Hongsen He, Zhimin QIU
article en

Abstract

The adaptive multichannel linear prediction (AMCLP) technique is one of the most effective methods for speech dereverberation in real-time application systems. However, the computational complexity of the associated AMCLP algorithms is very high. In this paper, we propose a low-complexity speech dereverberation approach based on AMCLP. The coefficient matrix of the multichannel dereverberation filter is decomposed into a linear combination of a set of low-dimensional sub-filter matrices and a set of low-dimensional sub-filter vectors through Kronecker product. According to these sub-filter models, we define two sets of signal models and cost functions, from which the adaptive speech dereverberation algorithm is established. This solution not only leads to effective dereverberation performance, but also reduces the computational cost, which is very conducive to realistic speech systems.

NOISE-CON proceedingsVol. 270(3)
Southwest University of Science and Technology (CN)
Peace, Justice and strong institutions
Openalex Percentile: Top 81%
Speech and Audio Processing
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A low-complexity adaptive speech dereverberation approach based on multichannel linear prediction — Yi Yu, Zhu Zhen, et al. · NOISE-CON proceedings (2024) | TGRS Research Map | TGRS