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dc.contributor.authorWu, WRen_US
dc.contributor.authorChen, PCen_US
dc.contributor.authorChang, HTen_US
dc.contributor.authorKuo, CHen_US
dc.date.accessioned2014-12-08T15:27:17Z-
dc.date.available2014-12-08T15:27:17Z-
dc.date.issued1998en_US
dc.identifier.isbn0-7803-4325-5en_US
dc.identifier.urihttp://hdl.handle.net/11536/19525-
dc.description.abstractKalman Filtering is an effective speech enhancement technique, in which speech and noise signals are usually modeled as autoregressive (AR) processes and represented in the state-space domain. Since AR coefficients identification and Kalman filtering require extensive computations, practical implementation of this approach is difficult. This paper proposes a simple and practical scheme that overcomes these problems. Speech signals are first decomposed into subbands. Subband speech signals are then modeled as low-order AR processes, such that low-order Kalman filters can be applied. Enhanced fullband speech signals are finally obtained by combining the enhanced subband speech signals. Using a frame-based algorithm, autocorrelation functions of subband speech are calculated and the Yuler-Walker equations are solved to obtain the AR parameters. Simulation results show that Kalman filtering in the subband domain not only greatly reduces the computational complexity but also achieves better performance compared to that in the fullband domain.en_US
dc.language.isoen_USen_US
dc.titleFrame-based subband Kalman filtering for speech enhancementen_US
dc.typeProceedings Paperen_US
dc.identifier.journalICSP '98: 1998 FOURTH INTERNATIONAL CONFERENCE ON SIGNAL PROCESSING, PROCEEDINGS, VOLS I AND IIen_US
dc.citation.spage682en_US
dc.citation.epage685en_US
dc.contributor.department電信工程研究所zh_TW
dc.contributor.departmentInstitute of Communications Engineeringen_US
dc.identifier.wosnumberWOS:000081007500165-
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