A two-stage method for single-channel speech enhancement

被引:2
|
作者
Hamid, ME [1 ]
Fukabayashi, T
机构
[1] Shizuoka Univ, Grad Sch Elect Sci & Technol, Hamamatsu, Shizuoka 4328561, Japan
[2] Shizuoka Univ, Fac Engn, Hamamatsu, Shizuoka 4328561, Japan
关键词
enhancement of speech; single-channel; autocorrelalion function; degree of noise; subtraction in time domain; blind source separation;
D O I
10.1093/ietfec/e89-a.4.1058
中图分类号
TP3 [计算技术、计算机技术];
学科分类号
0812 ;
摘要
A time domain (TD) speech enhancement technique to improve SNR in noise-contaminated speech is proposed. Additional supplementary scheme is applied to estimate the degree of noise of noisy speech. This is estimated from a function, which is previously prepared as the function of the parameter of the degree of noise. The function is obtained by least square (LS) method using the given degree of noise and the estimated parameter of the degree of noise. This parameter is obtained from the autocorrelation function (ACF) on frame-by-frame basis. This estimator almost accurately estimates the degree of noise and it is useful to reduce noise. The proposed method is based on two-stage processing. In the first stage, subtraction in time domain (STD), which is equivalent to ordinary spectral subtraction (SS), is carried out. In the result, the noise is reduced to a certain level. Further reduction of noise and by-product noise residual is carried out in the second stage, where blind source separation (BSS) technique is applied in time domain. Because the method is a single-channel speech enhancement, the other signal is generated by taking the noise characteristics into consideration in order to apply BSS. The generated signal plays a very important role in BSS. This paper presents an adaptive algorithm for separating sources in convolutive mixtures modeled by finite impulse response (FIR) filters. The coefficients of the FIR filter are estimated from the decorrelation of two mixtures. Here we are recovering only one signal of interest, in particular the voice of primary speaker free from interfering noises. In the experiment, the different levels of noise are added to the clean speech signal and the improvement of SNR at each stage is investigated. The noise types considered initially in this study consist of the synthesized white and color noise with SNR set from 0 to 30 dB. The proposed method is also tested with other real-world noises. The results show that the satisfactory SNR improvement is attained in the two-stage processing.
引用
收藏
页码:1058 / 1068
页数:11
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