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アイテム
Speech Analysis Based on Modeling the Effective Voice Source(Speech Analysis, <Special Section> Statistical Modeling for Speech Processing)
https://sucra.repo.nii.ac.jp/records/12882
https://sucra.repo.nii.ac.jp/records/12882e93cfd1b-e4a9-496a-90f5-539a2d0aa2fa
名前 / ファイル | ライセンス | アクション |
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Item type | 学術雑誌論文 / Journal Article(1) | |||||
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公開日 | 2007-10-04 | |||||
タイトル | ||||||
タイトル | Speech Analysis Based on Modeling the Effective Voice Source(Speech Analysis, <Special Section> Statistical Modeling for Speech Processing) | |||||
言語 | ||||||
言語 | eng | |||||
キーワード | ||||||
主題Scheme | Other | |||||
主題 | glottal waveform | |||||
キーワード | ||||||
主題Scheme | Other | |||||
主題 | effective voice source | |||||
キーワード | ||||||
主題Scheme | Other | |||||
主題 | linear prediction | |||||
キーワード | ||||||
主題Scheme | Other | |||||
主題 | least square method | |||||
キーワード | ||||||
主題Scheme | Other | |||||
主題 | system identification | |||||
資源タイプ | ||||||
資源タイプ識別子 | http://purl.org/coar/resource_type/c_6501 | |||||
資源タイプ | journal article | |||||
著者 |
RAHMAN, M. Shahidur
× RAHMAN, M. Shahidur× 島村, 徹也 |
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著者 ローマ字 | ||||||
値 | RAHMAN, M. Shahidur | |||||
著者 ローマ字 | ||||||
値 | SHIMAMURA, Tetsuya | |||||
著者 所属 | ||||||
値 | 埼玉大学工学部情報工学科 | |||||
著者 所属 | ||||||
値 | 埼玉大学大学院理工学研究科数理電子情報部門情報システム工学領域 | |||||
著者 所属(別言語) | ||||||
値 | Department of Information and Computer Sciences, Saitama University | |||||
著者 所属(別言語) | ||||||
値 | Graduate School of Science and Engineering, Saitama University | |||||
書誌情報 |
IEICE transactions on information and systems 巻 E89-D, 号 3, p. 1107-1115, 発行日 2006 |
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年月次 | ||||||
値 | 2006-3 | |||||
出版者名 | ||||||
出版者 | 社団法人電子情報通信学会 | |||||
収録物識別子 | ||||||
収録物識別子タイプ | ISSN | |||||
収録物識別子 | 09168532 | |||||
抄録 | ||||||
内容記述タイプ | Abstract | |||||
内容記述 | A new system identification based method has been proposed for accurate estimation of vocal tract parameters. An often encountered problem in using the conventional linear prediction analysis is due to the harmonic structure of the excitation source of voiced speech. This harmonic characteristic is coupled with the estimation of autoregressive (AR) coefficients that results in difficulties in estimating the vocal tract filter. This paper models the effective voice source from the residual obtained through the covariance analysis in the first-pass which is then used as input to the second-pass least-square analysis. A better source-filter separation is thus achieved. The formant frequencies and corresponding bandwidths obtained using the proposed method for synthetic vowels are found to be accurate up to a factor of more than three (in percent) compared to the conventional method. Since the source characteristic is taken into account, local variations due to the positioning of analysis window are reduced significantly. The validity of the proposed method is also examined by inspecting the spectra obtained from natural vowel sounds uttered by high-pitched female speaker. | |||||
注記 | ||||||
内容記述タイプ | Other | |||||
内容記述 | copyright(c)2006 IEICE 許諾番号:07RB0174 http://www.ieice.org/jpn/trans_online/index.html |
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資源タイプ | ||||||
内容記述タイプ | Other | |||||
内容記述 | text | |||||
フォーマット | ||||||
内容記述タイプ | Other | |||||
内容記述 | application/pdf | |||||
アイテムID | ||||||
値 | A1003075 |