摘要
在分析经典自适应滤波算法的基础之上提出了一种有效的消除噪声且减小语音失真的语音增强方法.为了进一步提高语音增强效果,根据人耳掩蔽效应把语音信号中关键频率段的阈值作为自适应滤波算法的动态系数,根据此系数估算误差函数.在Matlab上对此算法输入不同信噪比的信号,仿真结果表明:输出的信噪比明显高于传统滤波算法,且残留噪声较小.
Based on the analysis of classical adaptive filtering algorithm,an effective speech enhancement algorithm is proposed,which could reduce noise and distortion effectively. According to the auditory masking properties,this algorithm uses masking thresholds of the key frequency segments as dynamic coefficients,to further improve the effect of speech enhancement and estimate error function dynamically. Signals with different SNR are input to simulate this proposed algrithm on Matlab. The high SNR and low residual noise of the output signals show that the algrithm is better than traditional filtering algrithms.
出处
《南京信息工程大学学报（自然科学版）》
CAS
2010年第6期529-532,共4页
Journal of Nanjing University of Information Science & Technology（Natural Science Edition）
关键词
语音增强
听觉掩蔽效应
自适应滤波
speech enhancement
auditory masking properties
adaptive filtering
