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            Speech Enhancement via Bayesian Multi-solution Shrinker
            作者:Zongbo Xie Jiuchao Feng
            來源:本站原創
            更新時間:2014/1/14 10:35:00
            正文:


            (1-2. School of Electronic and Information Engineering, South China University of Technology, Guangzhou, 510641)


            Abstract: To effectively extract inherent information from measured speech signals, it is important to preprocess data to reduce noise. In this letter, we propose an algorithm---Bayesian multi-solution shrinker (BMS) for speech enhancement. The basic idea of BMS is to utilize empirical Bayesian model in the wavelet coefficients shrinkage step. Using speech data and calculating signal-to-noise ratio (SNR) and segmental signal-to-noise ratio (SSNR), we show that shown that the proposed approach outperforms the benchmark methods based on log-spectral amplitude (LSA), spectral subtraction and Stein's Unbiased Risk Estimate (SURE) wavelet denoising, respectively.

            Keywords: Speech enhancement; denoise; Bayesian multi-solution shrinker.

             

             


                                                                        REFERENCE
            (1)Boll S.F. Suppression of acoustic noise in speech using spectral subtraction. IEEE Trans. Acoustics Speech Signal Process., 1979, 27: 113-120.
            (2)I. Cohen, B. Berdugo. Speech enhancement for non-stationary noise environments. Signal Processing, 2001, 81: 2403-2418. 
            (3)Donoho D.L., Johnstone I.M. Ideal spatial adaptation by wavelet shrinkage. Biometrika, 1994, 81: 425–455.
            (4)Donoho D.L., Johnstone I.M. Ideal Spatial Adaptation by Wavelet Shrinkage. Department of Statistics, Stanford University, USA. April 1993.
            (5)Vidakovic B., Ruggeri F. BAMS method: Theory and simulations. The Indian Journal of Statistics, Series B, 2001, 63(2): 234-249. 
            (6)Vidakovic B. Statistical Modeling by Wavelets. Wiley, New York, 1999.
            (7)Yang R., Berger J. A catalog of noninformative priors. Discussion Paper, 97042, ISDS, Duke University, NC. 
            (8)Garofolo J., Lamel L., Fisher W., Fiscus J., Pallett D., Dahlgren N., et al. TIMIT Acoustic–Phonetic Continuous Speech Corpus: Linguistic Data Consortium, 1993.
            (9)Johnson M.T., Yuan X., Ren Y.. Speech signal enhancement through adaptive wavelet thresholding. Speech Communication, 2007, 49: 123-133.
            (10)The MathWorks Inc. Wavelet Toolbox. Matlab: The Language of the Technical Computing. Version 7.14, Release 2012a, 2012.

             

             

            作者簡介: 

                 謝宗伯,博士,華南理工大學副研究員,IEEE/IEICE會員,主要研究方向為信號處理與機器學習。先后主持國家和省部級項目多項,在國際高水平期刊發表論文多篇。

               馮久超,教授,博導,華南理工大學教授,IEEE會員,廣東省“珠江學者”特聘教授,主要研究方向為非線性系統理論。

             
             
               
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