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Using a fast RLS adaptive algorithm for efficient speech processing

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dc.contributor.author Papaodysseus, C en
dc.contributor.author Roussopoulos, G en
dc.contributor.author Panagopoulos, A en
dc.date.accessioned 2014-03-01T01:23:19Z
dc.date.available 2014-03-01T01:23:19Z
dc.date.issued 2005 en
dc.identifier.issn 0378-4754 en
dc.identifier.uri https://dspace.lib.ntua.gr/xmlui/handle/123456789/16900
dc.subject Adaptive RLS filtering en
dc.subject Forward linear prediction en
dc.subject Speech coding en
dc.subject Speech processing en
dc.subject.classification Computer Science, Interdisciplinary Applications en
dc.subject.classification Computer Science, Software Engineering en
dc.subject.classification Mathematics, Applied en
dc.subject.other Adaptive filtering en
dc.subject.other Algorithms en
dc.subject.other Approximation theory en
dc.subject.other Computational complexity en
dc.subject.other Linear equations en
dc.subject.other Set theory en
dc.subject.other Speech processing en
dc.subject.other Adaptive recursive least square (RLS) filtering en
dc.subject.other Forward linear prediction (FLP) en
dc.subject.other Quantization error en
dc.subject.other Speech signals en
dc.subject.other Speech coding en
dc.title Using a fast RLS adaptive algorithm for efficient speech processing en
heal.type journalArticle en
heal.identifier.primary 10.1016/j.matcom.2004.10.007 en
heal.identifier.secondary http://dx.doi.org/10.1016/j.matcom.2004.10.007 en
heal.language English en
heal.publicationDate 2005 en
heal.abstract In this paper, a new method is presented that offers efficient computation of Linear Prediction Coefficients (LPC) via a new Recursive Least Squares (RLS) adaptive filtering algorithm. This method can be successfully used in speech coding and processing. The introduced algorithm is numerically robust, fast, parallelizable and has particularly good tracking properties. By means of this scheme, Linear Prediction Coefficients are obtained that offer an improvement in the reconstruction of the speech signal before coding, as compared to the signal obtained by various classical algorithm. An analogous improvement is observed in speech coding experiments too, while a subjective test confirms the improvement of the quality of synthesized speech. The overall processing time of the proposed method of speech coding is a bit greater, but comparable to the time the classical methods need. © 2004 IMACS. Published by Elsevier B.V All rights reserved. en
heal.publisher ELSEVIER SCIENCE BV en
heal.journalName Mathematics and Computers in Simulation en
dc.identifier.doi 10.1016/j.matcom.2004.10.007 en
dc.identifier.isi ISI:000229077000002 en
dc.identifier.volume 68 en
dc.identifier.issue 2 en
dc.identifier.spage 105 en
dc.identifier.epage 113 en


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