Computer Engineering and Applications ›› 2012, Vol. 48 ›› Issue (7): 142-145.

• 数据库、信号与信息处理 • Previous Articles     Next Articles

Adaptive speech enhancement based on spectral subtraction and LMS

JIANG Zhancai1,2, SUN Yan3, WANG Defang1   

  1. 1.Department of Physics, Qinghai Normal University, Xining 810008, China
    2.Tibetan Information Processing Center, Qinghai Normal University, Xining 810008, China
    3.School of Computer Science and Technology, Qinghai Nationalities University, Xining 810007, China
  • Received:1900-01-01 Revised:1900-01-01 Online:2012-03-01 Published:2012-03-01

基于谱减和LMS的自适应语音增强

姜占才1,2,孙 燕3,王得芳1   

  1. 1.青海师范大学 物理系,西宁 810008
    2.青海师范大学 藏文信息处理中心,西宁 810008
    3.青海民族大学 计算机科学与技术学院,西宁 810007

Abstract: FIR adaptive filtering algorithm can improve the adaptive speed based on spectral subtraction and LMS of the short-term non-stationary or wide range signals for speech enhancement in condition broadband noise. This algorithm detects an expected speech signal of derived from short-term noise with spectral subtraction, this expected signal is used as reference signal of the output filter. The method improves performance of filter and chooses differences as the error signal between the referenced signal and the output filter. The filters of reconstruction are revised directly by filter corrective value based on LMS of weigh. This algorithm adopts the way of normalization LMS and symbolization LMS and blocking LMS in the steepest descent adjusting the weights, and simplifies step selection of converge weight, and reduces the computation of corrective weight. The simulation experiment shows that the algorithm can effectively enhance speech of all kinds of SNR, and it is superior to spectral subtraction. The result shows that the enhancement method is satisfaction in the 3 dB.

摘要: 针对宽带噪声背景下的语音增强问题,将短时语音视为非平稳或宽平稳信号,基于谱减法和自适应滤波的最小均方(LMS)算法,提出了一种FIR型自适应滤波算法(SSLMS):用减谱法由短时噪声观测语音估计期望信号,作为滤波器输出信号的参考信号;用滤波器的输出与参考信号的差值为误差信号,用LMS算法求得滤波器权系数修正量,并修正滤波器。权系数最速下降调整中,采用了归一化LMS、符号LMS、块LMS技术,以简化保证权系数收敛的步长选择、减少权系数修正的运算量,从而提高自适应速度。对不同的语音在各种信噪比下仿真实验,并与改进的谱减法比较,结果表明,该法增强效果优于谱减法;在信噪比为3 dB时该法的增强效果仍然令人满意。