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Applied speech processing = algorith...

Dey, Nilanjan, (1984-)

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Applied speech processing = algorithms and case studies /

紀錄類型:	書目-電子資源 : Monograph/item
正題名/作者:	Applied speech processing/ Nilanjan Dey.
其他題名:	algorithms and case studies /
作者:	Dey, Nilanjan,
出版者:	Amsterdam :Academic Press, : 2021.,
面頁冊數:	1 online resource.
內容註:	1. Comparative Analysis of Machine Learning Algorithms for Audio Event Classification using Mel Frequency Cepstral Coefficients 2. Closed-Set Speaker Identification System based on MFCC and PNCC features in Combination with Different Fusion Strategy 3. Kurtosis Data Selective Affine Projection Adaptive Filtering Algorithm for Speech Processing Applications 4. A novel approach towards phoneme-to-viseme mapping in Malayalam 5. Modified Least Mean Square Adaptive Noise Reduction Algorithm for Speech Enhancement 6. Recursive noise estimation-based Wiener filtering for monaural speech enhancement 7. Unsupervised Speech Enhancement based on Phase Aware Time-Frequency Mask Estimation.
標題:	Speech processing systems - Mathematical models. -
電子資源:	https://www.sciencedirect.com/science/book/9780128238981
ISBN:	9780128242131 (electronic bk.)

Applied speech processing = algorithms and case studies /
Dey, Nilanjan,1984-

Applied speech processingalgorithms and case studies /[electronic resource] :Nilanjan Dey. - Amsterdam :Academic Press,2021. - 1 online resource. - Primers in biomedical imaging devices and systems. - Primers in biomedical imaging devices and systems..

1. Comparative Analysis of Machine Learning Algorithms for Audio Event Classification using Mel Frequency Cepstral Coefficients 2. Closed-Set Speaker Identification System based on MFCC and PNCC features in Combination with Different Fusion Strategy 3. Kurtosis Data Selective Affine Projection Adaptive Filtering Algorithm for Speech Processing Applications 4. A novel approach towards phoneme-to-viseme mapping in Malayalam 5. Modified Least Mean Square Adaptive Noise Reduction Algorithm for Speech Enhancement 6. Recursive noise estimation-based Wiener filtering for monaural speech enhancement 7. Unsupervised Speech Enhancement based on Phase Aware Time-Frequency Mask Estimation.

ISBN: 9780128242131 (electronic bk.)

Nat. Bib. No.: GBC0H3702bnbSubjects--Topical Terms:

2137767
Speech processing systems
--Mathematical models.Index Terms--Genre/Form:

542853
Electronic books.

LC Class. No.: TK7882.S65

Dewey Class. No.: 006.454

Applied speech processing = algorithms and case studies /
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505 0 $a 1. Comparative Analysis of Machine Learning Algorithms for Audio Event Classification using Mel Frequency Cepstral Coefficients 2. Closed-Set Speaker Identification System based on MFCC and PNCC features in Combination with Different Fusion Strategy 3. Kurtosis Data Selective Affine Projection Adaptive Filtering Algorithm for Speech Processing Applications 4. A novel approach towards phoneme-to-viseme mapping in Malayalam 5. Modified Least Mean Square Adaptive Noise Reduction Algorithm for Speech Enhancement 6. Recursive noise estimation-based Wiener filtering for monaural speech enhancement 7. Unsupervised Speech Enhancement based on Phase Aware Time-Frequency Mask Estimation.
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650 0 $a Speech processing systems $v Case studies. $3 3614893
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