by tuskbyte
Open source · 148 downloads · 0 likes
Le modèle *Nepali Male v1* est un système de synthèse vocale (TTS) basé sur l'architecture VITS, spécialisé dans la génération de voix masculine en népalais. Il transforme un texte en une parole naturelle et expressive, en intégrant des mécanismes d'apprentissage variational et adversarial pour améliorer la qualité du rendu. Grâce à un prédicteur de durée stochastique, il peut produire des intonations et des rythmes variés à partir d'un même texte, offrant ainsi une grande flexibilité d'expression. Ce modèle est particulièrement adapté à la création de contenus audio en népalais, comme des livres audio, des assistants vocaux ou des applications éducatives. Sa capacité à générer des voix réalistes et dynamiques le distingue des solutions TTS traditionnelles, tout en restant accessible via des bibliothèques comme Hugging Face Transformers.
Nepali language
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.
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("procit001/nepali_male_v1")
tokenizer = AutoTokenizer.from_pretrained("procit001/nepali_male_v1")
text = "म पनि जान्छु है त अहिले लाई"
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)
The model is licensed as atulpokharel.
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