ASLGSDMLMar 5, 2020

Statistical Context-Dependent Units Boundary Correction for Corpus-based Unit-Selection Text-to-Speech

arXiv:2003.02837v21 citations
Originality Incremental advance
AI Analysis

This work addresses speaker adaptation for TTS systems, offering a simpler alternative to more expensive methods, though it appears incremental as it builds on existing HMM segmentation.

The paper tackles the problem of improving segmentation accuracy in unit-selection text-to-speech systems by proposing a statistical model that corrects boundaries using context-dependent phonetic unit classes, reducing systematic errors from HMM-based segmentation compared to a reference alignment.

In this study, we present an innovative technique for speaker adaptation in order to improve the accuracy of segmentation with application to unit-selection Text-To-Speech (TTS) systems. Unlike conventional techniques for speaker adaptation, which attempt to improve the accuracy of the segmentation using acoustic models that are more robust in the face of the speaker's characteristics, we aim to use only context dependent characteristics extrapolated with linguistic analysis techniques. In simple terms, we use the intuitive idea that context dependent information is tightly correlated with the related acoustic waveform. We propose a statistical model, which predicts correcting values to reduce the systematic error produced by a state-of-the-art Hidden Markov Model (HMM) based speech segmentation. Our approach consists of two phases: (1) identifying context-dependent phonetic unit classes (for instance, the class which identifies vowels as being the nucleus of monosyllabic words); and (2) building a regression model that associates the mean error value made by the ASR during the segmentation of a single speaker corpus to each class. The success of the approach is evaluated by comparing the corrected boundaries of units and the state-of-the-art HHM segmentation against a reference alignment, which is supposed to be the optimal solution. In conclusion, our work supplies a first analysis of a model sensitive to speaker-dependent characteristics, robust to defective and noisy information, and a very simple implementation which could be utilized as an alternative to either more expensive speaker-adaptation systems or of numerous manual correction sessions.

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