A novel noise robust and low bit rate speech coding algorithm
dc.authorid | 0000-0002-7008-4778 | |
dc.authorid | 0000-0002-4597-0954 | |
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 | 2015-07-14T23:48:10Z | |
dc.date.available | 2015-07-14T23:48:10Z | |
dc.date.issued | 2009 | |
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 noise robust and variable length frame based speech modeling method is introduced. This method consists of three major steps which includes noise removal algorithm, coding and encoding algorithms, respectively. Coding and encoding parts are developed based on SYMPES (SYsteMatic Procedure for Predefined Envelope and Signature sequence sets). These sets have been developed in two types which represent voiced and unvoiced parts of the speech signals separately in order to obtain more efficient coding strategy and higher compression ratio while preserving the perceptual quality of the speech signals. As an extension of our previous works our new framework is not only consider the coding of the clean speech signals but also noisy speech signals. The new noise robust module suppresses the noise and delivers the clean speech signal to the newly designed modeling part. The modeling part promises higher compression ratios by switching to the more appropriate type of predefined sets take into account the voiced and unvoiced frames. | en_US |
dc.description.version | Publisher's Version | en_US |
dc.identifier.citation | Güz, Ü., Gürkan, H. & Yarman, B. S. B. (2009). A novel noise robust and low bit rate speech coding algorithm. Paper presented at the 2009 24th International Symposium on Computer and Information Sciences, 471-474. doi:10.1109/ISCIS.2009.5291876 | en_US |
dc.identifier.doi | 10.1109/ISCIS.2009.5291876 | |
dc.identifier.endpage | 474 | |
dc.identifier.isbn | 9781424450213 | |
dc.identifier.isbn | 9781424450237 | |
dc.identifier.scopus | 2-s2.0-73949092707 | |
dc.identifier.scopusquality | N/A | |
dc.identifier.startpage | 471 | |
dc.identifier.uri | https://hdl.handle.net/11729/624 | |
dc.identifier.uri | http://dx.doi.org/10.1109/ISCIS.2009.5291876 | |
dc.identifier.wos | WOS:000275024200083 | |
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 | 0000-0002-7008-4778 | |
dc.institutionauthorid | 0000-0002-4597-0954 | |
dc.language.iso | en | en_US |
dc.peerreviewed | Yes | en_US |
dc.publicationstatus | Published | en_US |
dc.publisher | IEEE | en_US |
dc.relation.ispartof | 2009 24th International Symposium on Computer and Information Sciences | en_US |
dc.relation.publicationcategory | Konferans Öğesi - Uluslararası - Kurum Öğretim Elemanı | en_US |
dc.rights | info:eu-repo/semantics/closedAccess | en_US |
dc.subject | Speech processing | en_US |
dc.subject | Signal reconstruction | en_US |
dc.subject | Signal representations | en_US |
dc.subject | Speech coding | en_US |
dc.subject | Speech synthesis | en_US |
dc.subject | Sympes | en_US |
dc.subject | Low bit rate speech coding algorithm | en_US |
dc.subject | Noise removal algorithm | en_US |
dc.subject | Speech signals | en_US |
dc.subject | Bit rate | en_US |
dc.subject | Databases | en_US |
dc.subject | Encoding | en_US |
dc.subject | Noise robustness | en_US |
dc.subject | Signal to noise ratio | en_US |
dc.subject | Speech enhancement | en_US |
dc.subject | Working environment noise | en_US |
dc.subject | Compression ratio (machinery) | en_US |
dc.subject | Encoding (symbols) | en_US |
dc.subject | Information science | en_US |
dc.subject | Signal analysis | en_US |
dc.subject | Clean speech | en_US |
dc.subject | Coding strategy | en_US |
dc.subject | Compression ratios | en_US |
dc.subject | Encoding algorithms | en_US |
dc.subject | Low bit-rate speech coding | en_US |
dc.subject | Noisy speech signals | en_US |
dc.subject | Perceptual quality | en_US |
dc.subject | Predefined sets | en_US |
dc.subject | Signature sequence sets | en_US |
dc.subject | Speech modeling | en_US |
dc.subject | Variable length | en_US |
dc.title | A novel noise robust and low bit rate speech coding algorithm | en_US |
dc.type | Conference Object | en_US |
dspace.entity.type | Publication |
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