Downloads · 30 days
19
3% of all-time downloads
namphungdn134/whisper-base-vi
whisper-base-vi is a automatic speech recognition model from namphungdn134. Use it when you need speech turned into text. It is set up for transformers. The card lists the license as mit.
should probably proofread and complete it, then remove this comment. --
Downloads · 30 days
19
3% of all-time downloads
All-time downloads
598
Public
Parameters
72.6M
291 MB on disk
Likes
2
Public
Click a slice to open those files.
.safetensors290 MB · 99%
From the Hugging Face model README
This is a fine-tuned version of openai/whisper-base model on 100 hours of Vietnamese speech data. The model aims to improve transcription accuracy and robustness for Vietnamese automatic speech recognition (ASR) tasks, especially in real-world scenarios.
Evaluation was performed on a held-out test set with diverse regional accents and speaking styles.
The Whisper Base model is a transformer-based sequence-to-sequence model designed for automatic speech recognition and translation tasks. It has been trained on over 680,000 hours of labeled audio data in multiple languages. The fine-tuned version of this model focuses on the Vietnamese language, aiming to improve transcription accuracy and handling of local dialects.
This model works with the WhisperProcessor to pre-process audio inputs into log-Mel spectrograms and decode them into text.
To use the fine-tuned model, you can go to: https://github.com/namphung134/np-asr-vietnamese
Or you can follow these steps:
Install the required dependencies:
# Install required libraries
!pip install transformers torch librosa soundfile --quiet
# Import necessary libraries
import torch
import librosa
import soundfile as sf
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
print("Environment setup completed!")
Use the model for inference:
import torch
import librosa
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
device = "cuda" if torch.cuda.is_available() else "cpu"
print(f"Using device: {device}")
# Load processor and model
model_id = "namphungdn134/whisper-base-vi"
print(f"Loading model from: {model_id}")
processor = AutoProcessor.from_pretrained(model_id)
model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id).to(device)
# config language and task
forced_decoder_ids = processor.get_decoder_prompt_ids(language="vi", task="transcribe")
model.config.forced_decoder_ids = forced_decoder_ids
print(f"Forced decoder IDs for Vietnamese: {forced_decoder_ids}")
# Preprocess
audio_path = "example.wav"
print(f"Loading audio from: {audio_path}")
audio, sr = librosa.load(audio_path, sr=16000)
input_features = processor(audio, sampling_rate=16000, return_tensors="pt").input_features.to(device)
print(f"Input features shape: {input_features.shape}")
# Generate
print("Generating transcription...")
with torch.no_grad():
predicted_ids = model.generate(input_features, forced_decoder_ids=forced_decoder_ids)
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
print("📝 Transcription:", transcription)
# Debug: Print token to check
print("Predicted IDs:", predicted_ids[0].tolist())
This model is licensed under the MIT License.
If you use this model in your research or application, please cite the original Whisper model and this fine-tuning work as follows:
@article{Whisper2021,
title={Whisper: A Multilingual Speech Recognition Model},
author={OpenAI},
year={2021},
journal={arXiv:2202.12064},
url={https://arxiv.org/abs/2202.12064}
}
@misc{title={Whisper Base Vi V1.1 - Nam Phung},
author={Nam Phùng},
organization={DUT},
year={2025},
url={https://huggingface.co/namphungdn134/whisper-base-vi}
}
For questions, collaborations, or suggestions, feel free to reach out via [[email protected]].