摘要
提出一种基于GSC的语音增强算法,该算法应用了DFT调制子带滤波器组将语音信号分解到子带进行自适应滤波,从而获得更好的增强效果以及更低的运量复杂度。同时,将范数约束自适应滤波(NCAF)算法应用于自适应噪声对消器(ANC)以降低语音的失真度。为了进一步去除增强后语音中的残留噪声,算法使用改进的Wiener后置滤波器。仿真结果表明,相对于基于全带GSC的麦克风阵列语音增强算法以及传统Wiener后置滤波算法,采用本文所用算法具有更高的输出分段信噪比。
A kind of speech enhancement algorithm based on GSC is put forward in this report. This algorithm applies DTF modulated sub-band filter banks to decompose speech signal into sub-band in order to do the adaptive filtering which can realize better enhancement and lower computational complexity.Meanwhile,the norm-constrained adaptive filters （NCAF） is applied in adaptive noise canceller （ANC） to decrease voice distortion. In order to remove the residual noise in the enhanced speech signal, the improved Wiener post-filter is used. The simulation result indicates that compared with the microphone array speech enhancement algorithm based on full-band GSC and traditional Wiener post-filtering algorithm,the algorithm introduced in this report can require higher output segment signal-noise ratio.
出处
《电子设计工程》
2013年第5期173-175,179,共4页
Electronic Design Engineering
