The University of Tokyo · Computer Science
Professor Takaaki Saeki's research lab specializes in self-supervised and semi-supervised learning for speech and text processing, with a strong focus on multilingual and low-resource speech technologies. The lab develops advanced neural speech synthesis (TTS) and speech restoration systems that reduce reliance on high-quality paired data, enabling applications in low-resource and zero-shot multilingual settings. Key research directions include joint speech-text pretraining, degradation-robust TTS, and self-supervised restoration using real-world degraded speech. The lab aims to make high-quality speech technologies accessible to thousands of languages by leveraging text-only data and transfer learning.
Figures are computed from collected data and may differ slightly.
We present the UTokyo-SaruLab mean opinion score (MOS) prediction system submitted to VoiceMOS Challenge 2022.The challenge is to predict the MOS values of speech samples collected from previous Blizzard Challenges and Voice Conversion Challenges for two tracks: a main track for in-domain prediction and an out-of-domain (OOD) track for which there is less labeled data from different listening tests.Our system is based on ensemble learning of strong and weak learners.Strong learners incorporate s
While neural text-to-speech (TTS) has achieved human-like natural synthetic speech, multilingual TTS systems are limited to resource-rich languages due to the need for paired text and studio-quality audio data. This paper proposes a method for zero-shot multilingual TTS using text-only data for the target language. The use of text-only data allows the development of TTS systems for low-resource languages for which only textual resources are available, making TTS accessible to thousands of langua
This paper proposes Virtuoso, a massively multilingual speech–text joint semi-supervised learning framework for text-to-speech synthesis (TTS) models. Existing multilingual TTS typically supports tens of languages, which are a small fraction of the thousands of languages in the world. One difficulty to scale multilingual TTS to hundreds of languages is collecting high-quality speech–text paired data in low-resource languages. This study extends Maestro, a speech–text joint pretraining framework
Neural text-to-speech (TTS) systems have made significant progress in generating natural synthetic speech. However, neural TTS requires large amounts of paired training data, which limits its applicability to a small number of resource-rich languages. Previous work on low-resource TTS has addressed the data hungriness based on transfer learning from a multilingual model to low-resource languages, but it still relies heavily on the availability of paired data for the target languages. In this pap
Most text-to-speech (TTS) methods use high-quality speech corpora recorded in a well-designed environment, incurring a high cost for data collection.To solve this problem, existing noise-robust TTS methods are intended to use noisy speech corpora as training data.However, they only address either timeinvariant or time-variant noises.We propose a degradationrobust TTS method, which can be trained on speech corpora that contain both additive noises and environmental distortions.It jointly represen
We present a self-supervised speech restoration method without paired speech corpora.Because the previous general speech restoration method uses artificial paired data created by applying various distortions to high-quality speech corpora, it cannot sufficiently represent acoustic distortions of real data, limiting the applicability.Our model consists of analysis, synthesis, and channel modules that simulate the recording process of degraded speech and is trained with real degraded speech data i
Incremental text-to-speech (TTS) synthesis generates utterances in small linguistic units for the sake of real-time and low-latency applications. We previously proposed an incremental TTS method that leverages a large pre-trained language model to take unobserved future context into account without waiting for the subsequent segment. Although this method achieves comparable speech quality to that of a method that waits for the future context, it entails a huge amount of processing for sampling f
Collecting high-quality studio recordings of audio is challenging, which limits the language coverage of text-to-speech (TTS) systems. This paper proposes a framework for scaling a multilingual TTS model to 100+ languages using found data without supervision. The proposed framework combines speech-text encoder pretraining with unsupervised training using untranscribed speech and unspoken text data sources, thereby leveraging massively multilingual joint speech and text representation learning. W
Restoring high-quality speech from degraded historical recordings is crucial for the preservation of cultural and endangered linguistic resources. A key challenge in this task is the scarcity of paired training data that replicate the original acoustic conditions of the historical audio. While previous approaches have used pseudo paired data generated by applying various distortions to clean speech corpora, their limitations stem from the inability to authentically simulate the acoustic variatio
This paper proposes two high-fidelity and computationally efficient neural voice conversion (VC) methods based on a direct waveform modification using spectral differentials. The conventional spectral-differential VC method with a minimum-phase filter achieves high-quality conversion for narrow-band (16 kHz-sampled) VC but requires heavy computational cost in filtering. This is because the minimum phase obtained using a fixed lifter of the Hilbert transform often results in a long-tap filter. Fu
In this paper, we propose computationally efficient and high-quality methods for statistical voice conversion (VC) with direct waveform modification based on spectral differentials. The conventional method with a minimum-phase filter achieves high-quality conversion but requires heavy computation in filtering. This is because the minimum phase using a fixed lifter of the Hilbert transform often results in a long-tap filter. One of our methods is a data-driven method for lifter training. Since th
Cooperative relaying (CR) is a promising technique to provide spatial diversity by combining multiple signals from source and relay stations. In the present paper, the impact and use of the asymmetric property in bi-directional CR under asymmetric traffic conditions are discussed assuming that CR involves one communication pair and one relay station in a time division duplex (TDD) system. The asymmetric property means that the average communication quality differs for each transmission direction
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