SPEECH ENHANCEMENT USING AN ADAPTIVE WIENER FILTERING APPROACH

被引:34
作者
Abd El-Fattah, M. A. [1 ]
Dessouky, M. I. [1 ]
Diab, S. M. [1 ]
Abd El-Samie, F. E. [1 ]
机构
[1] Menoufia Univ, Fac Elect Engn, Dept Elect & Elect Commun, Menoufia, Egypt
来源
PROGRESS IN ELECTROMAGNETICS RESEARCH M | 2008年 / 4卷
关键词
D O I
10.2528/PIERM08061206
中图分类号
TM [电工技术]; TN [电子技术、通信技术];
学科分类号
0808 ; 0809 ;
摘要
This paper proposes the application of the Wiener filter in an adaptive manner in speech enhancement. The proposed adaptive Wiener filter depends on the adaptation of the filter transfer function from sample to sample based on the speech signal statistics (mean and variance). The adaptive Wiener filter is implemented in time domain rather than in frequency domain to accommodate for the varying nature of the speech signal. The proposed method is compared to the traditional Wiener filter and the spectral subtraction methods and the results reveal its superiority.
引用
收藏
页码:167 / 184
页数:18
相关论文
共 11 条
  • [1] Berouti M., 1979, ICASSP 79. 1979 IEEE International Conference on Acoustics, Speech and Signal Processing, P208
  • [2] SUPPRESSION OF ACOUSTIC NOISE IN SPEECH USING SPECTRAL SUBTRACTION
    BOLL, SF
    [J]. IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING, 1979, 27 (02): : 113 - 120
  • [3] Deller J. R., 1997, DISCRETE TIME PROCES
  • [4] Ephraim Y., 1993, ICASSP-93. 1993 IEEE International Conference on Acoustics, Speech, and Signal Processing (Cat. No.92CH3252-4), P355, DOI 10.1109/ICASSP.1993.319311
  • [5] EPHRAIM Y, 1995, INT CONF ACOUST SPEE, P804, DOI 10.1109/ICASSP.1995.479816
  • [6] Haykin S. S., 2008, ADAPTIVE FILTER THEO
  • [7] HU Y, 2002, ACOUST SPEECH SIG PR, P573
  • [8] Lim J. S., 1978, IEEE T ACOUST SPEECH, VASSP-26
  • [9] ENHANCEMENT AND BANDWIDTH COMPRESSION OF NOISY SPEECH
    LIM, JS
    OPPENHEIM, AV
    [J]. PROCEEDINGS OF THE IEEE, 1979, 67 (12) : 1586 - 1604
  • [10] Signal/noise KLT based approach for enhancing speech degraded by colored noise
    Mittal, U
    Phamdo, N
    [J]. IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, 2000, 8 (02): : 159 - 167