andrespm/w2v-bert-2.0-multi-gl-es-v1.0

W2v-BERT 2.0 speech encoder fine-tuned to Galician and Spanish

Fine-tuned version of the Conformer-based W2v-BERT 2.0 speech encoder as described in Section 3.2.1 of the paper, which is at the core of our Seamless models.

This model was pre-trained on 4.5M hours of unlabeled audio data covering more than 143 languages. It requires finetuning to be used for downstream tasks such as Automatic Speech Recognition (ASR), or Audio Classification.

Model Name	#params	checkpoint
W2v-BERT 2.0	600M	checkpoint

This model and its training are supported by 🤗 Transformers, more on it in the docs.

🤗 Transformers usage

This is a bare checkpoint without any modeling head, and thus requires finetuning to be used for downstream tasks such as ASR. You can however use it to extract audio embeddings from the top layer with this code snippet:

from transformers import AutoFeatureExtractor, Wav2Vec2BertModel
import torch
from datasets import load_dataset

dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation")
dataset = dataset.sort("id")
sampling_rate = dataset.features["audio"].sampling_rate

processor = AutoProcessor.from_pretrained("andrespm/w2v-bert-2.0-multi-gl-es-v1.0")
model = Wav2Vec2BertModel.from_pretrained("andrespm/w2v-bert-2.0-multi-gl-es-v1.0")

# audio file is decoded on the fly
inputs = processor(dataset[0]["audio"]["array"], sampling_rate=sampling_rate, return_tensors="pt")
with torch.no_grad():
    outputs = model(**inputs)

To learn more about the model use, refer to the following resources:

• its docs
• a blog post showing how to fine-tune it on Mongolian ASR
• a training script example