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dc.contributor.authorGomez, Randyja
dc.contributor.authorKawahara, Tatsuyaja
dc.contributor.alternative河原, 達也ja
dc.date.accessioned2010-10-19T02:09:26Z-
dc.date.available2010-10-19T02:09:26Z-
dc.date.issued2010-09-
dc.identifier.issn1558-7916ja
dc.identifier.urihttp://hdl.handle.net/2433/128840-
dc.description.abstractAutomatic speech recognition (ASR) in reverberant environments is a challenging task. Most dereverberation techniques address this problem through signal processing and enhances the reverberant waveform independent from the speech recognizer. In this paper, we propose a novel scheme to perform dereverberation in relation with the likelihood of the back-end ASR system. Our proposed approach effectively selects the dereverberation parameters, in the form of multiband scale factors, so that they improve the likelihood of the acoustic model. Then, the acoustic model is retrained using the optimal parameters. During the recognition phase, we implement additional optimization of the parameters. By using Gaussian mixture model (GMM), the process for selecting the scale factors become efficient. Moreover, we remove the dependency of the adopted dereverberation technique on the room impulse response (RIR) measurement, by using an artificial RIR generator and selecting based on the acoustic likelihood. Experimental results show significant improvement in recognition performance with the proposed method over the conventional approach.ja
dc.format.mimetypeapplication/pdfja
dc.language.isoeng-
dc.publisherIEEEja
dc.rights© 2010 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other users, including reprinting/ republishing this material for advertising or promotional purposes, creating new collective works for resale or redistribution to servers or lists, or reuse of any copyrighted components of this work in other works.ja
dc.titleRobust Speech Recognition Based on Dereverberation Parameter Optimization Using Acoustic Model Likelihoodja
dc.type.niitypeJournal Articleja
dc.identifier.ncidAA12103538ja
dc.identifier.jtitleIEEE Transactions on Audio, Speech, and Language Processingja
dc.identifier.volume18ja
dc.identifier.issue7ja
dc.identifier.spage1708ja
dc.identifier.epage1716ja
dc.relation.doi10.1109/TASL.2010.2052610ja
dc.textversionpublisherja
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