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Neural ATSM: Fully Neural Network-based Adaptive Time-Scale Modification Using Sentence-Specific Dynamic Control
| DC Field | Value | Language |
|---|---|---|
| dc.contributor.author | Lee, Jaeuk | - |
| dc.contributor.author | Jang, Sohee | - |
| dc.contributor.author | Chang, Joon-Hyuk | - |
| dc.date.accessioned | 2025-02-12T08:00:33Z | - |
| dc.date.available | 2025-02-12T08:00:33Z | - |
| dc.date.issued | 2024-09 | - |
| dc.identifier.issn | 1990-9772 | - |
| dc.identifier.uri | https://scholarworks.bwise.kr/hanyang/handle/2021.sw.hanyang/206471 | - |
| dc.description.abstract | Adaptive time-scale modification (ATSM) adaptively adjusts audio speed and improves upon previous systems by tailoring the scale for each phoneme in two steps: phoneme positioning via Montreal forced aligner (MFA) and reconstruction with adaptive speaking rate. However, ATSM's phoneme-specific rate is constant regardless of sentences, and MFA struggles with precise phoneme alignment in synthetic speech. Driven by this, we propose a fully neural networks-based ATSM (Neural ATSM) that dynamically controls each phoneme's speaking rate to vary from sentence to sentence. It predicts phoneme-level rates using a speaking rate predictor and flexibly modifies the scales to fit sentence context using Gaussian upsampling and attention mechanism, ensuring feature similarity with Soft-dynamic time warping (DTW) loss. We also integrate a variational autoencoder (VAE) and flow models for enhanced time-scaled signals. Experimental results show that Neural ATSM outperforms ATSM for real and synthesized speech. | - |
| dc.format.extent | 5 | - |
| dc.language | 영어 | - |
| dc.language.iso | ENG | - |
| dc.title | Neural ATSM: Fully Neural Network-based Adaptive Time-Scale Modification Using Sentence-Specific Dynamic Control | - |
| dc.type | Article | - |
| dc.identifier.doi | 10.21437/Interspeech.2024-2380 | - |
| dc.identifier.scopusid | 2-s2.0-85214811259 | - |
| dc.identifier.wosid | 001331850105003 | - |
| dc.identifier.bibliographicCitation | Proceedings of the Annual Conference of the International Speech Communication Association, INTERSPEECH, pp 4903 - 4907 | - |
| dc.citation.title | Proceedings of the Annual Conference of the International Speech Communication Association, INTERSPEECH | - |
| dc.citation.startPage | 4903 | - |
| dc.citation.endPage | 4907 | - |
| dc.type.docType | Proceedings Paper | - |
| dc.description.isOpenAccess | N | - |
| dc.description.journalRegisteredClass | scopus | - |
| dc.relation.journalResearchArea | Computer Science | - |
| dc.relation.journalWebOfScienceCategory | Computer Science, Artificial Intelligence | - |
| dc.subject.keywordPlus | Speech enhancement | - |
| dc.subject.keywordAuthor | Adaptive time-scale modification | - |
| dc.subject.keywordAuthor | attention mechanism | - |
| dc.subject.keywordAuthor | Gaussian upsampling | - |
| dc.subject.keywordAuthor | speaking rate predictor | - |
| dc.identifier.url | https://www.isca-archive.org/interspeech_2024/lee24m_interspeech.html | - |
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