aiqbench analysis
This arXiv preprint proposes a neural model that predicts phoneme-level duration distributions from text, aimed at improving alignment in text-to-speech systems. The reported gains are in alignment accuracy and in robustness of zero-shot TTS when the prompt audio and input audio don't match. For readers tracking speech generation, the practical question is whether these results hold on established benchmarks against stronger baselines, which the abstract alone doesn't settle.
Adaptive Duration Model for Text Speech Alignment
arXiv:2507.22612v3 Announce Type: replace-cross Abstract: Speech-to-text alignment is a critical component of neural text to speech (TTS) models. Autoregressive TTS models typically use an attention mechanism to learn these alignments on-line, while non-autoregressive end to end TTS models rely on durations extracted from external sources. In this paper, we propose a novel duration prediction framework that can give promising phoneme-level duration distribution with given text. In our experiments, the proposed duration model has more precise prediction and adaptation ability to conditions, compared to previous baseline models. Specifically, it makes a considerable improvement on phoneme-level alignment accuracy and makes the performance of zero-shot TTS models more robust to the mismatch between prompt audio and input audio.
Abstract:Speech-to-text alignment is a critical component of neural text to speech (TTS) models. Autoregressive TTS models typically use an attention mechanism to learn these alignments on-line, while non-autoregressive end to end TTS models rely on durations extracted from external sources. In this paper, we propose a novel duration prediction framework that can give promising phoneme-level duration distribution with given text. In our experiments, the proposed duration model has more precise prediction and adaptation ability to conditions, compared to previous baseline models. Specifically, it makes a considerable improvement on phoneme-level alignment accuracy and makes the performance of zero-shot TTS models more robust to the mismatch between prompt audio and input audio.
Sound (cs.SD); Artificial Intelligence (cs.AI); Audio and Speech Processing (eess.AS)
arXiv:2507.22612 [cs.SD]
(or arXiv:2507.22612v3 [cs.SD] for this version)
https://doi.org/10.48550/arXiv.2507.22612 Focus to learn more arXiv-issued DOI via DataCite
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Full text under the publisher's open licence · 209 words
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