ZMM-TTS: Zero-Shot Multilingual and Multispeaker Speech Synthesis Conditioned on Self-Supervised Discrete Speech Representations

IF 4.1 2区 计算机科学 Q1 ACOUSTICS IEEE/ACM Transactions on Audio, Speech, and Language Processing Pub Date : 2024-09-06 DOI:10.1109/TASLP.2024.3451951
Cheng Gong;Xin Wang;Erica Cooper;Dan Wells;Longbiao Wang;Jianwu Dang;Korin Richmond;Junichi Yamagishi
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Abstract

Neural text-to-speech (TTS) has achieved human-like synthetic speech for single-speaker, single-language synthesis. Multilingual TTS systems are limited to resource-rich languages due to the lack of large paired text and studio-quality audio data. TTS systems are typically built using a single speaker's voice, but there is growing interest in developing systems that can synthesize voices for new speakers using only a few seconds of their speech. This paper presents ZMM-TTS, a multilingual and multispeaker framework utilizing quantized latent speech representations from a large-scale, pre-trained, self-supervised model. Our paper combines text-based and speech-based self-supervised learning models for multilingual speech synthesis. Our proposed model has zero-shot generalization ability not only for unseen speakers but also for unseen languages. We have conducted comprehensive subjective and objective evaluations through a series of experiments. Our model has proven effective in terms of speech naturalness and similarity for both seen and unseen speakers in six high-resource languages. We also tested the efficiency of our method on two hypothetically low-resource languages. The results are promising, indicating that our proposed approach can synthesize audio that is intelligible and has a high degree of similarity to the target speaker's voice, even without any training data for the new, unseen language.
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ZMM-TTS:以自监督离散语音表征为条件的零镜头多语言和多发言人语音合成
神经文本到语音(TTS)已在单人、单语种合成中实现了类人合成语音。由于缺乏大量配对文本和录音室质量的音频数据,多语言 TTS 系统仅限于资源丰富的语言。TTS 系统通常使用单个说话者的语音来构建,但人们对开发只使用几秒钟说话者语音就能为新说话者合成语音的系统越来越感兴趣。本文介绍了 ZMM-TTS,这是一种多语言和多发言人框架,它利用来自大规模预训练自监督模型的量化潜在语音表示。我们的论文结合了基于文本和基于语音的自监督学习模型,用于多语言语音合成。我们提出的模型不仅对未见过的说话人,而且对未见过的语言都具有零点泛化能力。我们通过一系列实验进行了全面的主观和客观评估。事实证明,我们的模型对六种高资源语言中见过和没见过的说话人都能有效地提高语音自然度和相似度。我们还在两种假设的低资源语言上测试了我们方法的效率。结果很有希望,表明我们提出的方法即使在没有任何新的、未见过的语言的训练数据的情况下,也能合成可理解的、与目标说话人的声音高度相似的音频。
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来源期刊
IEEE/ACM Transactions on Audio, Speech, and Language Processing
IEEE/ACM Transactions on Audio, Speech, and Language Processing ACOUSTICS-ENGINEERING, ELECTRICAL & ELECTRONIC
CiteScore
11.30
自引率
11.10%
发文量
217
期刊介绍: The IEEE/ACM Transactions on Audio, Speech, and Language Processing covers audio, speech and language processing and the sciences that support them. In audio processing: transducers, room acoustics, active sound control, human audition, analysis/synthesis/coding of music, and consumer audio. In speech processing: areas such as speech analysis, synthesis, coding, speech and speaker recognition, speech production and perception, and speech enhancement. In language processing: speech and text analysis, understanding, generation, dialog management, translation, summarization, question answering and document indexing and retrieval, as well as general language modeling.
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