期刊名称:Signal & Image Processing : An International Journal (SIPIJ)
印刷版ISSN:2229-3922
电子版ISSN:0976-710X
出版年度:2012
卷号:3
期号:3
页码:39
出版社:Academy & Industry Research Collaboration Center (AIRCC)
摘要:This paper describes the development of an adaptive noise cancellation algorithm for effective recognitionof speech signal and also to improve SNR for an adaptive step size input. An adaptive filter with Fast BlockLeast Mean square Algorithm is designed for noise free audio (speech/music) signals. The signal inputused is a audio speech signal which could be in the form of a recorded voice. The filter used is adaptivefilter and the algorithm used is Fast Block LMS algorithm. A Gaussian noise is added to this input signaland given as a input to the Fast Block LMS. The algorithm is implemented in Matlab and was tested fornoise cancellation in speech signals. A Simulink model is designed which results in a noise free audiospeech signal at the output. The FBLMS algorithm is computationally efficient in noise cancellation. Thenoise level in speech signal can be 1) mild, 2) moderate, 3) severe. The SNR is estimated by varying theadaptive step size.