๐ Wav2Vec2-Large-XLSR-53-Slovene
This is a fine-tuned model based on facebook/wav2vec2-large-xlsr-53 for Slovene speech recognition, using the Common Voice dataset. Ensure your speech input is sampled at 16kHz when using this model.
๐ Metadata
Property |
Details |
Language |
Slovene |
Datasets |
common_voice |
Tags |
audio, automatic-speech-recognition, speech, xlsr-fine-tuning-week |
License |
apache-2.0 |
Model Name |
XLSR Wav2Vec2 Slovene |
Task |
Speech Recognition (automatic-speech-recognition) |
Dataset |
Common Voice sl (common_voice, args: sl) |
Test WER |
36.97 |
๐ Quick Start
When using this model, make sure that your speech input is sampled at 16kHz.
โจ Features
- Fine - tuned for Slovene language using the Common Voice dataset.
- Can be used directly without a language model.
๐ฆ Installation
No specific installation steps are provided in the original document, so this section is skipped.
๐ป Usage Examples
Basic Usage
import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
test_dataset = load_dataset("common_voice", "sl", split="test[:2%]")
processor = Wav2Vec2Processor.from_pretrained("mrshu/wav2vec2-large-xlsr-slovene")
model = Wav2Vec2ForCTC.from_pretrained("mrshu/wav2vec2-large-xlsr-slovene")
resampler = torchaudio.transforms.Resample(48_000, 16_000)
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = torchaudio.load(batch["path"])
batch["speech"] = resampler(speech_array).squeeze().numpy()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
predicted_ids = torch.argmax(logits, dim=-1)
print("Prediction:", processor.batch_decode(predicted_ids))
print("Reference:", test_dataset["sentence"][:2])
๐ง Evaluation
The model can be evaluated as follows on the Slovene test data of Common Voice.
import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re
test_dataset = load_dataset("common_voice", "sl", split="test")
wer = load_metric("wer")
processor = Wav2Vec2Processor.from_pretrained("mrshu/wav2vec2-large-xlsr-slovene")
model = Wav2Vec2ForCTC.from_pretrained("mrshu/wav2vec2-large-xlsr-slovene")
model.to("cuda")
chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"\โ\%\โ\โ\๏ฟฝ\ยซ\ยป\)\(\โ\'\โ\โ\โ]'
resampler = torchaudio.transforms.Resample(48_000, 16_000)
def speech_file_to_array_fn(batch):
batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
speech_array, sampling_rate = torchaudio.load(batch["path"])
batch["speech"] = resampler(speech_array).squeeze().numpy()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
def evaluate(batch):
inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
pred_ids = torch.argmax(logits, dim=-1)
batch["pred_strings"] = processor.batch_decode(pred_ids)
return batch
result = test_dataset.map(evaluate, batched=True, batch_size=8)
print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
Test Result: 36.97 %
๐ Training
The Common Voice train
, validation
datasets were used for training.
The script used for training can be found here
๐ License
This model is licensed under the apache - 2.0 license.