A new algorithm for high speed speech and audio coding
dc.authorid | 0000-0002-4597-0954 | |
dc.authorid | 0000-0002-7008-4778 | |
dc.authorid | 0000-0003-1562-5524 | |
dc.contributor.author | Güz, Ümit | en_US |
dc.contributor.author | Gürkan, Hakan | en_US |
dc.contributor.author | Yarman, Bekir Sıddık Binboğa | en_US |
dc.date.accessioned | 2019-08-31T12:10:23Z | |
dc.date.accessioned | 2019-08-05T16:03:06Z | |
dc.date.available | 2019-08-31T12:10:23Z | |
dc.date.available | 2019-08-05T16:03:06Z | |
dc.date.issued | 2007 | |
dc.department | Işık Üniversitesi, Mühendislik Fakültesi, Elektrik-Elektronik Mühendisliği Bölümü | en_US |
dc.department | Işık University, Faculty of Engineering, Department of Electrical-Electronics Engineering | en_US |
dc.description.abstract | In this work, a new mathematical modeling approach is proposed for the representation of the speech and audio signals. This approach is based on the generation of the so called Predefined Signature Sequence (PSS) and Predefined Envelope Sequence (PES) Sets. After the generation process of the PSS and PES sets, they are clustered by effective k-means clustering algorithm and the PSS and PES are redefined by using the centroids of the clusters. By using this approach, the drawbacks such as the size of the sets, speed of the reconstruction process (computational complexity) which arise in our proposed methods previously are highly eliminated. In spite of these improvements, the initial results proved that, the quality of the reconstructed signals remains within the limitations of the acceptable hearing quality. | en_US |
dc.description.sponsorship | European Circuit Society (ECS) | en_US |
dc.description.sponsorship | IEEE Circuits and Systems Society | en_US |
dc.description.sponsorship | Institute of Electrical and Electronics Engineering (IEEE) | en_US |
dc.description.sponsorship | Ministerio de Educacion y Ciencia | en_US |
dc.description.sponsorship | Universidad de Sevilla | en_US |
dc.description.version | Publisher's Version | en_US |
dc.identifier.citation | Güz, Ü., Gürkan, H. & Yarman, B. S. B. (2007). A new algorithm for high speed speech and audio coding. Paper presented at the 2007 European Conference on Circuit Theory and Design, 180-183. doi:10.1109/ECCTD.2007.4529566 | en_US |
dc.identifier.doi | 10.1109/ECCTD.2007.4529566 | |
dc.identifier.endpage | 183 | |
dc.identifier.isbn | 9781424413416 | |
dc.identifier.isbn | 9781424413423 | |
dc.identifier.isbn | 1424413427 | |
dc.identifier.scopus | 2-s2.0-49749111485 | |
dc.identifier.scopusquality | N/A | |
dc.identifier.startpage | 180 | |
dc.identifier.uri | https://hdl.handle.net/11729/1806 | |
dc.identifier.uri | https://dx.doi.org/10.1109/ECCTD.2007.4529566 | |
dc.identifier.wos | WOS:000258708400046 | |
dc.identifier.wosquality | N/A | |
dc.indekslendigikaynak | Web of Science | en_US |
dc.indekslendigikaynak | Scopus | en_US |
dc.indekslendigikaynak | Conference Proceedings Citation Index – Science (CPCI-S) | en_US |
dc.institutionauthor | Güz, Ümit | en_US |
dc.institutionauthor | Gürkan, Hakan | en_US |
dc.institutionauthorid | 0000-0002-4597-0954 | |
dc.institutionauthorid | 0000-0002-7008-4778 | |
dc.language.iso | en | en_US |
dc.peerreviewed | Yes | en_US |
dc.publicationstatus | Published | en_US |
dc.publisher | IEEE | en_US |
dc.relation.ispartof | 2007 European Conference on Circuit Theory and Design | en_US |
dc.relation.publicationcategory | Konferans Öğesi - Uluslararası - Kurum Öğretim Elemanı | en_US |
dc.rights | info:eu-repo/semantics/closedAccess | en_US |
dc.subject | Audio coding | en_US |
dc.subject | Audio signal representation | en_US |
dc.subject | Audio signal | en_US |
dc.subject | Audition | en_US |
dc.subject | Auditory system | en_US |
dc.subject | Bandwidth | en_US |
dc.subject | Bit rate | en_US |
dc.subject | Circuit theory | en_US |
dc.subject | Clustering algorithms | en_US |
dc.subject | Computational complexity | en_US |
dc.subject | Data compression | en_US |
dc.subject | Electrocardiography | en_US |
dc.subject | Generation process | en_US |
dc.subject | Hearing quality | en_US |
dc.subject | High speed | en_US |
dc.subject | K-means clustering | en_US |
dc.subject | K-means clustering algorithm | en_US |
dc.subject | Mathematical modeling | en_US |
dc.subject | New mathematical model | en_US |
dc.subject | Predefined envelope sequence sets | en_US |
dc.subject | Predefined signature sequence sets | en_US |
dc.subject | Pulse modulation | en_US |
dc.subject | Reconstruction process | en_US |
dc.subject | Signal compression | en_US |
dc.subject | Signal reconstruction | en_US |
dc.subject | Signal to noise ratio | en_US |
dc.subject | Signature sequences | en_US |
dc.subject | Speech coding | en_US |
dc.subject | Speech signal representation | en_US |
dc.subject | Speech synthesis | en_US |
dc.subject | Speech and audio coding | en_US |
dc.title | A new algorithm for high speed speech and audio coding | en_US |
dc.type | Conference Object | en_US |
dspace.entity.type | Publication |
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