Model reference · open weights

mms-tts-hin

Audio facebook Text→speech 1 build Non-commercial 119k dl/mo

mms-tts-hin is an open-weight audio or speech model from Meta. mms-tts-hin (FP32) weighs 73 MB; the smallest configuration that runs it is RTX 3060 12 GB.

  • mms-tts-hin is a 36M parameter text-to-speech model developed by Meta for the Hindi language.
  • It utilizes the VITS architecture to synthesize speech waveforms from text inputs and is part of the Massively Multilingual Speech project.
  • The model is available under the CC-BY-NC 4.0 license and can be used with the Transformers library version 4.33 or later.

Summary of the facebook/mms-tts-hin model card, 2026-10-01

What it is

Released byMeta
Released2023-09-01
Parameters36M
VRAM73 MB for the weights

What it runs on

Memory and cards for mms-tts-hin (FP32)

73 MBweights, file size
1.6 GBruntime overhead
CardOne streamMemory
RTX 3060 12 GBfits11.6 GB
RTX 4060 Ti 16 GBfits15.4 GB
RTX 3090 24 GBfits23.4 GB
RTX 4090 24 GBfits23.4 GB
RTX 5090 32 GBfits31.0 GB
L40S 48 GBfits44.0 GB
A100 80 GBfits78.2 GB
H100 80 GBfits78.1 GB
RTX PRO 6000 Blackwell 96 GBfits93.8 GB
DGX Spark (GB10) 128 GB unifiedfits107 GB
H200 141 GBfits138 GB
B200 180 GBfits176 GB

From the model card

What Meta says about mms-tts-hin

Read the model card

This repository contains the Hindi (hin) language text-to-speech (TTS) model checkpoint.

This model is part of Facebook's Massively Multilingual Speech project, aiming to provide speech technology across a diverse range of languages. You can find more details about the supported languages and their ISO 639-3 codes in the MMS Language Coverage Overview, and see all MMS-TTS checkpoints on the Hugging Face Hub: facebook/mms-tts.

MMS-TTS is available in the 🤗 Transformers library from version 4.33 onwards.

Model Details

VITS (Variational Inference with adversarial learning for end-to-end Text-to-Speech) is an end-to-end speech synthesis model that predicts a speech waveform conditional on an input text sequence. It is a conditional variational autoencoder (VAE) comprised of a posterior encoder, decoder, and conditional prior.

A set of spectrogram-based acoustic features are predicted by the flow-based module, which is formed of a Transformer-based text encoder and multiple coupling layers. The spectrogram is decoded using a stack of transposed convolutional layers, much in the same style as the HiFi-GAN vocoder. Motivated by the one-to-many nature of the TTS problem, where the same text input can be spoken in multiple ways, the model also includes a stochastic duration predictor, which allows the model to synthesise speech with different rhythms from the same input text.

The model is trained end-to-end with a combination of losses derived from variational lower bound and adversarial training. To improve the expressiveness of the model, normalizing flows are applied to the conditional prior distribution. During inference, the text encodings are up-sampled based on the duration prediction module, and then mapped into the waveform using a cascade of the flow module and HiFi-GAN decoder. Due to the stochastic nature of the duration predictor, the model is non-deterministic, and thus requires a fixed seed to generate the same speech waveform.

For the MMS project, a separate VITS checkpoint is trained on each langauge.

Usage

MMS-TTS is available in the 🤗 Transformers library from version 4.33 onwards. To use this checkpoint, first install the latest version of the library:

pip install --upgrade transformers accelerate

Then, run inference with the following code-snippet:

from transformers import VitsModel, AutoTokenizer
import torch

model = VitsModel.from_pretrained("facebook/mms-tts-hin")
tokenizer = AutoTokenizer.from_pretrained("facebook/mms-tts-hin")

text = "some example text in the Hindi language"
inputs = tokenizer(text, return_tensors="pt")

with torch.no_grad():
    output = model(**inputs).waveform

The resulting waveform can be saved as a .wav file:

import scipy

scipy.io.wavfile.write("techno.wav", rate=model.config.sampling_rate, data=output.float().numpy())

Or displayed in a Jupyter Notebook / Google Colab:

from IPython.display import Audio

Audio(output.numpy(), rate=model.config.sampling_rate)

BibTex citation

This model was developed by Vineel Pratap et al. from Meta AI. If you use the model, consider citing the MMS paper:

@article{pratap2023mms,
    title={Scaling Speech Technology to 1,000+ Languages},
    author={Vineel Pratap and Andros Tjandra and Bowen Shi and Paden Tomasello and Arun Babu and Sayani Kundu and Ali Elkahky and Zhaoheng Ni and Apoorv Vyas and Maryam Fazel-Zarandi and Alexei Baevski and Yossi Adi and Xiaohui Zhang and Wei-Ning Hsu and Alexis Conneau and Michael Auli},
    journal={arXiv},
    year={2023}
}

License

The model is licensed as CC-BY-NC 4.0.

Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.

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