A novel fast algorithm for speech and audio coding
dc.authorid | 0000-0002-7008-4778 | |
dc.authorid | 0000-0002-4597-0954 | |
dc.authorid | 0000-0002-7008-4778 | en_US |
dc.authorid | 0000-0002-4597-0954 | en_US |
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 | 2015-07-14T23:48:12Z | |
dc.date.available | 2015-07-14T23:48:12Z | |
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 these centroids. By using this approach, the drawbacks by means of the size of the sets, speed of the reconstruction process (computational complexity) which arise in the 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.version | Publisher's Version | en_US |
dc.identifier.citation | Güz, Ü., Gürkan, H. & Yarman, B. S. B. (2007). A novel fast algorithm for speech and audio coding. Paper presented at the 2007 IEEE International Symposium on Circuits and Systems, 4020-4023. doi:10.1109/ISCAS.2007.378800 | en_US |
dc.identifier.doi | 10.1109/ISCAS.2007.378800 | |
dc.identifier.endpage | 4023 | |
dc.identifier.isbn | 9781424409204 | |
dc.identifier.isbn | 1424409217 | |
dc.identifier.isbn | 1424409209 | |
dc.identifier.issn | 0271-4302 | en_US |
dc.identifier.issn | 2158-1525 | en_US |
dc.identifier.scopus | 2-s2.0-34548812574 | en_US |
dc.identifier.scopusquality | N/A | en_US |
dc.identifier.startpage | 4020 | |
dc.identifier.uri | https://hdl.handle.net/11729/645 | |
dc.identifier.uri | http://dx.doi.org/10.1109/ISCAS.2007.378800 | |
dc.identifier.wos | WOS:000251608405092 | en_US |
dc.identifier.wosquality | N/A | en_US |
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.language.iso | en | en_US |
dc.peerreviewed | Yes | en_US |
dc.publicationstatus | Published | en_US |
dc.publisher | IEEE | en_US |
dc.relation.ispartof | 2007 IEEE International Symposium on Circuits and Systems | 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 | Auditory system | en_US |
dc.subject | Bandwidth | en_US |
dc.subject | Bit rate | en_US |
dc.subject | Clustering algorithms | en_US |
dc.subject | Computational complexity | en_US |
dc.subject | Pulse modulation | en_US |
dc.subject | Signal to noise ratio | en_US |
dc.subject | Speech coding | en_US |
dc.subject | Speech synthesis | en_US |
dc.subject | K-Means clustering algorithm | en_US |
dc.subject | Mathematical modeling | en_US |
dc.subject | Predefined envelope sequence | en_US |
dc.subject | Predefined signature sequence | en_US |
dc.subject | Signal representation | en_US |
dc.subject | Signals reconstruction | en_US |
dc.subject | Approximation theory | en_US |
dc.subject | Statistical analysis | en_US |
dc.subject | Algorithms | en_US |
dc.subject | Mathematical models | en_US |
dc.subject | Set theory | en_US |
dc.subject | Predefined Signature Sequence (PSS) | en_US |
dc.title | A novel fast algorithm for speech and audio coding | en_US |
dc.type | Conference Object | en_US |
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