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dc.contributor.authorKeser, Serkan
dc.contributor.authorGerek, Ömer Nezih
dc.contributor.authorSeke, Erol
dc.contributor.authorGillmezoğlu, Mehmet Bilginer
dc.date.accessioned2019-10-21T20:11:57Z
dc.date.available2019-10-21T20:11:57Z
dc.date.issued2017
dc.identifier.issn0167-6393
dc.identifier.issn1872-7182
dc.identifier.urihttps://dx.doi.org/10.1016/j.specom.2017.09.002
dc.identifier.urihttps://hdl.handle.net/11421/20364
dc.descriptionWOS: 000414819300005en_US
dc.description.abstractIn this study, two novel methods, which are based on Karhunen Loeve Transform (KLT) and Independent Component Analysis (ICA), are proposed for coding of speech signals. Instead of immediately dealing with eigenvalue magnitudes, the KLT- and ICA-based methods use eigenvectors of covariance matrices (or independent components for ICA) by geometrically grouping these vectors into fewer numbers of vectors. In this way, a data representation compaction is achieved. Further compression is achieved through discarding autocovariance eigenvectors corresponding to the small eigenvalues and applying vector quantization on the remaining eigenvectors. Additionally, this study proposes an iterative error refinement process, which uses the rest of the available bandwidth in order to transmit an efficient representation of the description error for better SNR. The overall process constitutes a new approach to efficient speech coding, with ICA being used in subspace speech coding for the first time. Constant bit rate (CBR) and variable bit rate (VBR) coding algorithms are employed with the proposed methods. TIMIT speech database is used in the experimental studies. Speech signals are synthesized at 2.4 kbps, 8 kbps, 12.2 kbps, 16 kbps, 16.4kbps and 19.85 kbps rates by using various frame lengths. The qualities of synthesized speech signals are compared to those of available speech codecs, i.e., LPC (2.4 kbps), G.728 (LD-CELP, 16 kbps), G.729A (CS-CELP, 8 kbps), EVS (16.4 kbps), AMR-NB (12.2 kbps) and AMR-WB (19.85 kbps)en_US
dc.language.isoengen_US
dc.publisherElsevier Science BVen_US
dc.relation.isversionof10.1016/j.specom.2017.09.002en_US
dc.rightsinfo:eu-repo/semantics/closedAccessen_US
dc.subjectIndependent Component Analysis (Ica)en_US
dc.subjectKarhunen Loeve Transform (Klt)en_US
dc.subjectSpeech Codecsen_US
dc.subjectSubspace Methodsen_US
dc.titleA subspace based progressive coding method for speech compressionen_US
dc.typearticleen_US
dc.relation.journalSpeech Communicationen_US
dc.contributor.departmentAnadolu Üniversitesi, Mühendislik Fakültesi, Elektrik ve Elektronik Mühendisliği Bölümüen_US
dc.identifier.volume94en_US
dc.identifier.startpage50en_US
dc.identifier.endpage61en_US
dc.relation.publicationcategoryMakale - Uluslararası Hakemli Dergi - Kurum Öğretim Elemanıen_US
dc.contributor.institutionauthorGerek, Ömer Nezih


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