Speech Enhancement using Equivalent Source Inverse Filtering -Based Microphone Array

• Main Author: 何克男
• Other Authors: 白明憲
• Format: Others
• Language: en_US
• Published: 2009
• Online Access: http://ndltd.ncl.edu.tw/handle/01925904156427701223

碩士 === 國立交通大學 === 機械工程系所 === 97 === New microphone array techniques are proposed in this paper for acoustic signal processing in telecommunication application. These endeavors are based on the central idea of Equivalent Source Inverse Filtering (ESIF). The single input multiple output equivalence source imaging (SIMO-ESI) algorithms are suggested to reconstruct the speech signal in a reverberant environment. Specifically, the system serves two purposed: dereverberation and noise reduction. It has promise in telecommunication application such as the automotive hands-free system, where noise-corrupted speech signal often needs to be enhanced. In order to further improve the noise reduction performance in spatial filtering and robustness against system uncertainties, the SIMO-ESIF algorithm is combined with an adaptive Generalized Side-lobe Canceller (GSC). The system is implemented on an NI-PXI platform and evaluated experimentally in car environment. As indicated by several performance measures in noise reduction and speech distortion, the proposed microphone array algorithm proved effective in reducing noise in human speech without significantly compromising the speech quality. The results of subjective tests were processed by using analysis of variance (ANOVA) to justify the statistic significance. A post-hoc test Fisher’s LSD was conducted to further assess the pairwise difference between the NR algorithms.