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khanusa/nd_asr_wav2vec2
nd_asr_wav2vec2 is a automatic speech recognition model from khanusa. Use it when you need speech turned into text. The card lists the license as apache-2.0.
This is a fine-tuned wav2vec2 model for Vietnamese Automatic Speech Recognition (ASR), based on nguyenvulebinh/wav2vec2-base-vi.
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Updated Jul 23, 2025
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From the Hugging Face model README
This is a fine-tuned wav2vec2 model for Vietnamese Automatic Speech Recognition (ASR), based on nguyenvulebinh/wav2vec2-base-vi.
import torch
import librosa
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
# Load model and processor
processor = Wav2Vec2Processor.from_pretrained("khanusa/nd_asr_wav2vec2")
model = Wav2Vec2ForCTC.from_pretrained("khanusa/nd_asr_wav2vec2")
# Load and preprocess audio
audio, sr = librosa.load("path_to_your_audio.wav", sr=16000)
# Tokenize and predict
inputs = processor(audio, sampling_rate=16000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values).logits
# Decode predictions
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(predicted_ids)[0]
print(transcription)
17.36 hours custom Vietnamese speech dataset
| Metric | Value |
|---|---|
| WER | 0.2123 |
Note: Please update the WER value with your actual evaluation results.
This model was fine-tuned from an English base model on a specific Vietnamese speech dataset and may not generalize well to:
@article{baevski2020wav2vec,
title={wav2vec 2.0: A framework for self-supervised learning of speech representations},
author={Baevski, Alexei and Zhou, Yuhao and Mohamed, Abdelrahman and Auli, Michael},
journal={Advances in neural information processing systems},
volume={33},
pages={12449--12460},
year={2020}
}
This model is released under the Apache 2.0 License.