MODEL PROFILE

tts-1.6b-en_fr

kyutai · tts-1.6b-en_fr

See also the pre-print research paper, the project page, the Colab example, the GitHub repository, and the repository of voices. This is a model for streaming text-to-speech (TTS). Unlike offline text-to-speech, where the model needs the entire text to produce the audio, our model starts to output audio as soon as the first few words from the text have been given as input. This model is actually 1.8B parameters, not 1.6B as the name might suggest. The model architecture is a hierarchical Transformer that consumes tokenized text and generateds audio tokenized by Mimi, see the Moshi paper. The frame rate is 12.5 Hz and each audio frame is represented by 32 audio tokens, although you can use less tokens at inference time for faster generation. The backbone model is 1B parameters, and the depth transformer is 600M parameters and uses partial weight sharing similar to Hibiki. The audio is shifted by 16 steps (1.28 sec.) with respect to the text, and the model uses an acoustic/semantic delay of 2.

Speech synthesislicense:cc-by-4.0
Context windowNot disclosed
Input → OutputText → Audio
Model weightsNot verified
Profile updated2026/11/2 19:15:40
OVERVIEW

Model overview

See also the pre-print research paper, the project page, the Colab example, the GitHub repository, and the repository of voices. This is a model for streaming text-to-speech (TTS). Unlike offline text-to-speech, where the model needs the entire text to produce the audio, our model starts to output audio as soon as the first few words from the text have been given as input. This model is actually 1.8B parameters, not 1.6B as the name might suggest. The model architecture is a hierarchical Transformer that consumes tokenized text and generateds audio tokenized by Mimi, see the Moshi paper. The frame rate is 12.5 Hz and each audio frame is represented by 32 audio tokens, although you can use less tokens at inference time for faster generation. The backbone model is 1B parameters, and the depth transformer is 600M parameters and uses partial weight sharing similar to Hibiki. The audio is shifted by 16 steps (1.28 sec.) with respect to the text, and the model uses an acoustic/semantic delay of 2.

Key features

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PRICING & COST

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INDEPENDENT BENCHMARKS

Public benchmarks & run configuration

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Sources & traceability

许可证:cc-by-4.0。模型介绍与第三方评测均以原来源的具体版本和方法为准。