HALL-E: Hierarchical Neural Codec Language Model for Minute-Long Zero-Shot Text-to-Speech Synthesis
Yuto Nishimura, Takumi Hirose, Masanari Ohi, Hideki Nakayama, Nakamasa Inoue
摘要
Recently, Text-to-speech (TTS) models based on large language models (LLMs) that translate natural language text into sequences of discrete audio tokens have gained great research attention, with advances in neural audio codec (NAC) models using residual vector quantization (RVQ). However, long-form speech synthesis remains a significant challenge due to the high frame rate, which increases the length of audio tokens and makes it difficult for autoregressive language models to generate audio tokens for even a minute of speech. To address this challenge, this paper introduces two novel post-training approaches: 1) Multi-Resolution Requantization (MReQ) and 2) HALL-E. MReQ is a framework to reduce the frame rate of pretrained NAC models. Specifically, it incorporates multi-resolution residual vector quantization (MRVQ) module that hierarchically reorganizes discrete audio tokens through teacher-student distillation. HALL-E is an LLM-based TTS model designed to predict hierarchical tokens of MReQ. Specifically, it incorporates the technique of using MRVQ sub-modules and continues training from a pre-trained LLM-based TTS model. Furthermore, to promote TTS research, we create MinutesSpeech, a new benchmark dataset consisting of 40k hours of filtered speech data for training and evaluating speech synthesis ranging from 3s up to 180s. In experiments, we demonstrated the effectiveness of our approaches by applying our post-training framework to VALL-E. We achieved the frame rate down to as low as 8 Hz, enabling the stable minituelong speech synthesis in a single inference step. Audio samples, dataset, codes and pre-trained models are available at https://yutonishimura-v2.github.io/HALL-E_DEMO. 1 We refer to the number of audio tokens per second as the frame rate (Hz).
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引用它的顶会 Paper6
- Towards Controllable Speech Synthesis in the Era of Large Language Models: A Systematic SurveyTianxin Xie, Yan Rong, Pengfei Zhang, Wenwu Wang 等EMNLP 2025 · 被引用 10 次
- Hierarchical Semantic-Acoustic Modeling via Semi-Discrete Residual Representations for Expressive End-to-End Speech SynthesisYixuan Zhou, Guoyang Zeng, Xin Liu, Xiang Li 等ICLR 2026
- Emotional Face-to-SpeechJiaxin Ye, Boyuan Cao, Hongming ShanICML 2025
- Multi-band Frequency Reconstruction for Neural Psychoacoustic CodingDianwen Ng, Kun Zhou, Yi-Wen Chao, Zhiwei Xiong 等ICML 2025
- Speech Token Prediction via Compressed-to-fine Language Modeling for Speech GenerationWenrui Liu, Qian Chen, Wen Wang, Guanrou Yang 等ACM MM 2025
它引用的顶会 Paper13
- Language Models are Few-Shot LearnersTom B. Brown, Benjamin Mann, Nick Ryder, Melanie Subbiah 等NeurIPS 2020 · 被引用 64,255 次
- High-Fidelity Audio Compression with Improved RVQGANRithesh Kumar, Prem Seetharaman, Alejandro Luebs, Ishaan Kumar 等NeurIPS 2023 · 被引用 910 次
- Simple and Controllable Music GenerationJade Copet, Felix Kreuk, Itai Gat, Tal Remez 等NeurIPS 2023 · 被引用 843 次
- NaturalSpeech 2: Latent Diffusion Models are Natural and Zero-Shot Speech and Singing SynthesizersKai Shen, Zeqian Ju, Xu Tan, Eric Liu 等ICLR 2024 · 被引用 362 次
- GLM-130B: An Open Bilingual Pre-trained ModelAohan Zeng, Xiao Liu, Zhengxiao Du, Zihan Wang 等ICLR 2023 · 被引用 295 次
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