A new method to represent speech signals via predefined signature and envelope sequences
dc.authorid | 0000-0002-4597-0954 | |
dc.authorid | 0000-0002-7008-4778 | |
dc.authorid | 0000-0002-4597-0954 | en_US |
dc.authorid | 0000-0002-7008-4778 | 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 | 2015-01-15T23:00:50Z | |
dc.date.available | 2015-01-15T23:00:50Z | |
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 | A novel systematic procedure referred to as "SYMPES" to model speech signals is introduced. The structure of SYMPES is based on the creation of the so-called predefined "signature S = {S(R)(n)} and envelope E = {E(K) (n)}" sets. These sets are speaker and language independent. Once the speech signals are divided into frames with selected lengths, then each frame sequence X(i)( n) is reconstructed by means of the mathematical form X(i)( n) = C(i)E(K) (n) S(R)(n). In this representation, C(i) is called the gain factor, S(R)(n) and E(K) (n) are properly assigned from the predefined signature and envelope sets, respectively. Examples are given to exhibit the implementation of SYMPES. It is shown that for the same compression ratio or better, SYMPES yields considerably better speech quality over the commercially available coders such as G. 726 (ADPCM) at 16 kbps and voice excited LPC-10E (FS1015) at 2.4 kbps. | en_US |
dc.description.version | Publisher's Version | en_US |
dc.identifier.citation | Güz, Ü., Gürkan, H. & Yarman, B. S. B. (2007). A new method to represent speech signals via predefined signature and envelope sequences. Eurasip Journal on Advances in Signal Processing, 2007, 1-17. doi:10.1155/2007/56382 | en_US |
dc.identifier.doi | 10.1155/2007/56382 | |
dc.identifier.endpage | 17 | |
dc.identifier.issn | 1110-8657 | en_US |
dc.identifier.issn | 1687-6172 | en_US |
dc.identifier.scopus | 2-s2.0-33846254213 | en_US |
dc.identifier.scopusquality | Q2 | en_US |
dc.identifier.startpage | 1 | |
dc.identifier.uri | https://hdl.handle.net/11729/269 | |
dc.identifier.uri | http://dx.doi.org/10.1155/2007/56382 | |
dc.identifier.volume | 2007 | |
dc.identifier.wos | WOS:000245273100001 | en_US |
dc.identifier.wosquality | Q2 | |
dc.identifier.wosquality | Q2 | en_US |
dc.indekslendigikaynak | Web of Science | en_US |
dc.indekslendigikaynak | Scopus | en_US |
dc.indekslendigikaynak | Science Citation Index Expanded (SCI-EXPANDED) | 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 | Hindawi Publishing Corporation | en_US |
dc.relation.ispartof | EURASIP Journal on Advances in Signal Processing | en_US |
dc.relation.publicationcategory | Makale - Uluslararası Hakemli Dergi - Kurum Öğretim Elemanı | en_US |
dc.rights | info:eu-repo/semantics/openAccess | en_US |
dc.subject | Electrocardiography | en_US |
dc.subject | Data compression | en_US |
dc.subject | Signal compression | en_US |
dc.subject | Code converters | en_US |
dc.subject | Image compression | en_US |
dc.subject | Mathematical models | en_US |
dc.subject | Numerical methods | en_US |
dc.subject | Compression ratio | en_US |
dc.subject | Gain factors | en_US |
dc.subject | Speech quality | en_US |
dc.subject | Speech signals | en_US |
dc.subject | Speech processing | en_US |
dc.title | A new method to represent speech signals via predefined signature and envelope sequences | en_US |
dc.type | Article | en_US |
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