VALL-E R: Robust and Efficient Zero-Shot Text-to-Speech Synthesis via Monotonic Alignment

Fuente: arXiv
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Autores principales: Han, Bing, Zhou, Long, Liu, Shujie, Chen, Sanyuan, Meng, Lingwei, Qian, Yanming, Liu, Yanqing, Zhao, Sheng, Li, Jinyu, Wei, Furu
Formato: Preprint
Publicado: 2024
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author Han, Bing
Zhou, Long
Liu, Shujie
Chen, Sanyuan
Meng, Lingwei
Qian, Yanming
Liu, Yanqing
Zhao, Sheng
Li, Jinyu
Wei, Furu
author_facet Han, Bing
Zhou, Long
Liu, Shujie
Chen, Sanyuan
Meng, Lingwei
Qian, Yanming
Liu, Yanqing
Zhao, Sheng
Li, Jinyu
Wei, Furu
contents With the help of discrete neural audio codecs, large language models (LLM) have increasingly been recognized as a promising methodology for zero-shot Text-to-Speech (TTS) synthesis. However, sampling based decoding strategies bring astonishing diversity to generation, but also pose robustness issues such as typos, omissions and repetition. In addition, the high sampling rate of audio also brings huge computational overhead to the inference process of autoregression. To address these issues, we propose VALL-E R, a robust and efficient zero-shot TTS system, building upon the foundation of VALL-E. Specifically, we introduce a phoneme monotonic alignment strategy to strengthen the connection between phonemes and acoustic sequence, ensuring a more precise alignment by constraining the acoustic tokens to match their associated phonemes. Furthermore, we employ a codec-merging approach to downsample the discrete codes in shallow quantization layer, thereby accelerating the decoding speed while preserving the high quality of speech output. Benefiting from these strategies, VALL-E R obtains controllablity over phonemes and demonstrates its strong robustness by approaching the WER of ground truth. In addition, it requires fewer autoregressive steps, with over 60% time reduction during inference. This research has the potential to be applied to meaningful projects, including the creation of speech for those affected by aphasia. Audio samples will be available at: https://aka.ms/valler.
format Preprint
id arxiv_https___arxiv_org_abs_2406_07855
institution arXiv
publishDate 2024
record_format arxiv
spellingShingle VALL-E R: Robust and Efficient Zero-Shot Text-to-Speech Synthesis via Monotonic Alignment
Han, Bing
Zhou, Long
Liu, Shujie
Chen, Sanyuan
Meng, Lingwei
Qian, Yanming
Liu, Yanqing
Zhao, Sheng
Li, Jinyu
Wei, Furu
Computation and Language
Sound
Audio and Speech Processing
With the help of discrete neural audio codecs, large language models (LLM) have increasingly been recognized as a promising methodology for zero-shot Text-to-Speech (TTS) synthesis. However, sampling based decoding strategies bring astonishing diversity to generation, but also pose robustness issues such as typos, omissions and repetition. In addition, the high sampling rate of audio also brings huge computational overhead to the inference process of autoregression. To address these issues, we propose VALL-E R, a robust and efficient zero-shot TTS system, building upon the foundation of VALL-E. Specifically, we introduce a phoneme monotonic alignment strategy to strengthen the connection between phonemes and acoustic sequence, ensuring a more precise alignment by constraining the acoustic tokens to match their associated phonemes. Furthermore, we employ a codec-merging approach to downsample the discrete codes in shallow quantization layer, thereby accelerating the decoding speed while preserving the high quality of speech output. Benefiting from these strategies, VALL-E R obtains controllablity over phonemes and demonstrates its strong robustness by approaching the WER of ground truth. In addition, it requires fewer autoregressive steps, with over 60% time reduction during inference. This research has the potential to be applied to meaningful projects, including the creation of speech for those affected by aphasia. Audio samples will be available at: https://aka.ms/valler.
title VALL-E R: Robust and Efficient Zero-Shot Text-to-Speech Synthesis via Monotonic Alignment
topic Computation and Language
Sound
Audio and Speech Processing
url https://arxiv.org/abs/2406.07855