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HALL-E: Hierarchical Neural Codec Language Model for Minute-Long Zero-Shot Text-to-Speech Synthesis

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arxiv 2410.04380 v1 pith:YKSBUN43 submitted 2024-10-06 eess.AS cs.SD

classification eess.AScs.SD
keywords audiomodelsspeechtokenshall-elanguagesynthesisframe
verification ladder T0 review T1 audit T2 compute T3 formal
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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 pre-trained 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 minitue-long 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/.

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Cited by 4 Pith papers

Reviewed papers in the Pith corpus that reference this work. Sorted by Pith novelty score. Full citation record

  1. Speech Token Prediction via Compressed-to-fine Language Modeling for Speech Generation

    eess.AS 2025-05 conditional novelty 6.0 of 10

    Compressed-to-fine language modeling improves speech token prediction by retaining prompt and local tokens while compressing long-range token spans into compact summaries.

  2. VoiceStar: Robust Zero-Shot Autoregressive TTS with Duration Control and Extrapolation

    eess.AS 2025-05 conditional novelty 6.0 of 10

    VoiceStar uses a progress-based rotary position embedding and mixed prompt training to give zero-shot voice cloning precise duration control and much longer output than training clips.

  3. Multimodal Large Language Model-Enabled Video Translation: A Role-Oriented Survey

    cs.CV 2026-04 unverdicted novelty 5.0 of 10

    The paper offers the first focused review of MLLM-based video translation organized by a three-role taxonomy of Semantic Reasoner, Expressive Performer, and Visual Synthesizer, plus open challenges.

  4. CLEAR: Continuous Latent Autoregressive Modeling for High-quality and Low-latency Speech Synthesis

    eess.AS 2025-08 conditional novelty 5.0 of 10

    CLEAR is a zero-shot TTS model that autoregressively predicts compact continuous audio latents with a per-token rectified flow head, reaching 1.88% WER on LibriSpeech Subset-B with an RTF of 0.29 and a 96 ms streaming delay.

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