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Author:

Zhou, J. (Zhou, J..) | Bao, C. (Bao, C..) | Zhang, X. (Zhang, X..) | Xiong, W. (Xiong, W..) | Jia, M. (Jia, M..)

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EI Scopus SCIE

Abstract:

Aiming at solving the problems of the conventional minimum variance distortionless response (MVDR) beamformer in practical applications, such as the sensibility of the steering vector mismatch and beampattern distortion, a robust broadband MVDR beamforming method with low-latency by reconstructing covariance matrix is proposed and applied to speech enhancement with a linear microphone array in this paper. In this work, some important steps are optimized, and the main contribution is to consider the problem of correlation terms generated by the low latency. Firstly, the direction of arrival (DOA) is corrected and the steering vector is estimated based on the sparsity of the DOAs corresponding to the sound sources, which improves the ability of anti-mismatches in the steering vector. Secondly, the correlation terms between the sound sources and noise are estimated and eliminated by the Capon power within the eigen-subspace, and the indirect dominant method is used to eliminate the correlation terms between the sound sources, so that the covariance matrix is reconstructed to obtain a more robust MVDR beamformer. Thirdly, the problem of white noise amplification at low frequency bins is analyzed, and a white noise gain (WNG) modification method is proposed to obtain a compromise between the interference suppression and WNG. In the experiments, the TIMIT corpus is used to generate the multi-channel speech data set, and the performance of the proposed method is evaluated with different DOAs and input signal to interference plus noise ratios (SINRs). The experimental results show that the proposed method can effectively suppress the interferences and reduce the noise with strong robustness. © 2023 Elsevier Ltd

Keyword:

Direction of arrival Minimum variance distortionless response Microphone array Speech enhancement Beamformer

Author Community:

  • [ 1 ] [Zhou J.]Speech and Audio Signal Processing Laboratory, Faculty of Information Technology, Beijing University of Technology, Beijing, 100124, China
  • [ 2 ] [Bao C.]Speech and Audio Signal Processing Laboratory, Faculty of Information Technology, Beijing University of Technology, Beijing, 100124, China
  • [ 3 ] [Zhang X.]Speech and Audio Signal Processing Laboratory, Faculty of Information Technology, Beijing University of Technology, Beijing, 100124, China
  • [ 4 ] [Xiong W.]Speech and Audio Signal Processing Laboratory, Faculty of Information Technology, Beijing University of Technology, Beijing, 100124, China
  • [ 5 ] [Jia M.]Speech and Audio Signal Processing Laboratory, Faculty of Information Technology, Beijing University of Technology, Beijing, 100124, China

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Source :

Applied Acoustics

ISSN: 0003-682X

Year: 2023

Volume: 211

3 . 4 0 0

JCR@2022

ESI Discipline: PHYSICS;

ESI HC Threshold:17

Cited Count:

WoS CC Cited Count: 0

SCOPUS Cited Count: 6

ESI Highly Cited Papers on the List: 0 Unfold All

WanFang Cited Count:

Chinese Cited Count:

30 Days PV: 8

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