Model reference · open weights
vits-ljs is an open-weight audio or speech model from kakao-enterprise. vits-ljs (FP32) weighs 73 MB; the smallest configuration that runs it is RTX 3060 12 GB.
For self-hosting, compare this build's requirements below with the GPU you plan to rent.
Summary of the kakao-enterprise/vits-ljs model card, 2026-10-01
What it is
| Released by | kakao-enterprise |
|---|---|
| Released | 2023-08-31 |
| Parameters | 36M |
| VRAM | 73 MB for the weights |
What it runs on
| Card | One stream | Memory |
|---|---|---|
| RTX 3060 12 GB | fits | 11.6 GB |
| RTX 4060 Ti 16 GB | fits | 15.4 GB |
| RTX 3090 24 GB | fits | 23.4 GB |
| RTX 4090 24 GB | fits | 23.4 GB |
| RTX 5090 32 GB | fits | 31.0 GB |
| L40S 48 GB | fits | 44.0 GB |
| A100 80 GB | fits | 78.2 GB |
| H100 80 GB | fits | 78.1 GB |
| RTX PRO 6000 Blackwell 96 GB | fits | 93.8 GB |
| DGX Spark (GB10) 128 GB unified | fits | 107 GB |
| H200 141 GB | fits | 138 GB |
| B200 180 GB | fits | 176 GB |
From the model card
VITS 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. This repository contains the weights for the official VITS checkpoint trained on the LJ Speech dataset.
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.
There are two variants of the VITS model: one is trained on the LJ Speech dataset, and the other is trained on the VCTK dataset. LJ Speech dataset consists of 13,100 short audio clips of a single speaker with a total length of approximately 24 hours. The VCTK dataset consists of approximately 44,000 short audio clips uttered by 109 native English speakers with various accents. The total length of the audio clips is approximately 44 hours.
| Checkpoint | Train Hours | Speakers |
|---|---|---|
| vits-ljs | 24 | 1 |
| vits-vctk | 44 | 109 |
VITS 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("kakao-enterprise/vits-ljs")
tokenizer = AutoTokenizer.from_pretrained("kakao-enterprise/vits-ljs")
text = "Hey, it's Hugging Face on the phone"
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)
Or displayed in a Jupyter Notebook / Google Colab:
from IPython.display import Audio
Audio(output, rate=model.config.sampling_rate)
This model was developed by Jaehyeon Kim et al. from Kakao Enterprise. If you use the model, consider citing the VITS paper:
@inproceedings{kim2021conditional,
title={"Conditional Variational Autoencoder with Adversarial Learning for End-to-end Text-to-speech"},
author={Kim, Jaehyeon and Kong, Jungil and Son, Juhee},
booktitle={International Conference on Machine Learning},
pages={5530--5540},
year={2021},
organization={PMLR}
}
The model is licensed as MIT.
Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.