Using a fast RLS adaptive algorithm for efficient speech processing

Papaodysseus, C; Roussopoulos, G; Panagopoulos, A

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. &COPY; 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