A new coding method for speech and audio signals
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 | 2019-08-05T16:05:04Z | |
dc.date.available | 2019-08-31T12:10:23Z | |
dc.date.available | 2019-08-05T16:05:04Z | |
dc.date.issued | 2005 | |
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 paper a new representation or modeling method of speech signals is introduced. The proposed method is based on the generation of the so-called Predefined Signature S={S R } and Envelope vector E={E K } Sets (PSEVS). These vector sets are speaker and language independent. In this method, once the speech signals are divided into frames with selected lengths, then each frame signal piece X i is reconstructed by means of the mathematical form of X i =C i E K S R . In this representation, C i is called the frame coefficient, S R and E K are the vectors properly assigned from the PSEVS respectively. It is shown that the proposed method provides fast reconstruction and substantial compression ratio with acceptable hearing quality. | en_US |
dc.description.version | Publisher's Version | en_US |
dc.identifier.citation | Güz, Ü., Gürkan, H. & Yarman, B. S. B. (2005). A new coding method for speech and audio signals. Paper presented at the 13th European Signal Processing Conference, EUSIPCO 2005, 361-364. | en_US |
dc.identifier.endpage | 364 | |
dc.identifier.isbn | 1604238216 | |
dc.identifier.isbn | 9781604238211 | |
dc.identifier.scopus | 2-s2.0-84863640292 | |
dc.identifier.scopusquality | N/A | |
dc.identifier.startpage | 361 | |
dc.identifier.uri | https://hdl.handle.net/11729/2021 | |
dc.indekslendigikaynak | Scopus | 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 | 13th European Signal Processing Conference, EUSIPCO 2005 | 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 | en_US |
dc.subject | Audio signals | en_US |
dc.subject | Audition | en_US |
dc.subject | Bit rate | en_US |
dc.subject | Coding method | en_US |
dc.subject | Coding methods | en_US |
dc.subject | Compression ratio | en_US |
dc.subject | Compression ratio (machinery) | en_US |
dc.subject | Data compression | en_US |
dc.subject | Electrocardiography | en_US |
dc.subject | Envelope vector sets | en_US |
dc.subject | Frame coefficient | en_US |
dc.subject | Hearing quality | en_US |
dc.subject | Least squares approximations | en_US |
dc.subject | Mathematical form | en_US |
dc.subject | Mathematical forms | en_US |
dc.subject | Modeling method | en_US |
dc.subject | Predefined signature sets | en_US |
dc.subject | Principal component analysis | en_US |
dc.subject | PSEVS | en_US |
dc.subject | Representation method | en_US |
dc.subject | Signal compression | en_US |
dc.subject | Signal processing | en_US |
dc.subject | Speech | en_US |
dc.subject | Speech coding | en_US |
dc.subject | Speech signals | en_US |
dc.subject | Vectors | en_US |
dc.title | A new coding method for speech and audio signals | en_US |
dc.type | Conference Object | en_US |
dspace.entity.type | Publication |
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