đ Wav2Vec2-Large-XLSR-53-Ukrainian
This is a fine - tuned model based on facebook/wav2vec2-large-xlsr-53 on the Ukrainian language, utilizing the Common Voice dataset. When using this model, ensure that your speech input is sampled at 16kHz.
đ Quick Start
This model is fine - tuned on Ukrainian using the Common Voice dataset. Make sure your speech input is sampled at 16kHz.
⨠Features
- Fine - tuned on Ukrainian with the Common Voice dataset.
- Can be used for automatic speech recognition tasks.
đĻ Installation
No specific installation steps are provided in the original document, so this section is skipped.
đģ Usage Examples
Basic Usage
The model can be used directly (without a language model) as follows:
import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
test_dataset = load_dataset("common_voice", "uk", split="test[:2%]")
processor = Wav2Vec2Processor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-ukrainian")
model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-ukrainian")
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])
Advanced Usage
No advanced usage examples are provided in the original document, so this part is not added.
đ Documentation
Evaluation
The model can be evaluated as follows on the Ukrainian test data of Common Voice.
import torch
import torchaudio
import urllib.request
import tarfile
import pandas as pd
from tqdm.auto import tqdm
from datasets import load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
data_url = "https://voice-prod-bundler-ee1969a6ce8178826482b88e843c335139bd3fb4.s3.amazonaws.com/cv-corpus-6.1-2020-12-11/uk.tar.gz"
filestream = urllib.request.urlopen(data_url)
data_file = tarfile.open(fileobj=filestream, mode="r|gz")
data_file.extractall()
wer = load_metric("wer")
processor = Wav2Vec2Processor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-ukrainian")
model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-ukrainian")
model.to("cuda")
cv_test = pd.read_csv("cv-corpus-6.1-2020-12-11/uk/test.tsv", sep='\t')
clips_path = "cv-corpus-6.1-2020-12-11/uk/clips/"
def clean_sentence(sent):
sent = sent.lower()
sent = sent.replace("â", "'")
sent = "".join(ch if ch.isalpha() or ch == "'" else " " for ch in sent)
sent = " ".join(sent.split())
return sent
targets = []
preds = []
for i, row in tqdm(cv_test.iterrows(), total=cv_test.shape[0]):
row["sentence"] = clean_sentence(row["sentence"])
speech_array, sampling_rate = torchaudio.load(clips_path + row["path"])
resampler = torchaudio.transforms.Resample(sampling_rate, 16_000)
row["speech"] = resampler(speech_array).squeeze().numpy()
inputs = processor(row["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)
targets.append(row["sentence"])
preds.append(processor.batch_decode(pred_ids)[0])
print("WER: {:2f}".format(100 * wer.compute(predictions=preds, references=targets)))
Test Result: 32.29 %
Training
The Common Voice train
and validation
datasets were used for training.
đ§ Technical Details
No specific technical details are provided in the original document, so this section is skipped.
đ License
This model is licensed under the apache - 2.0
license.
Information Table
Property |
Details |
Model Type |
Wav2Vec2 - Large - XLSR - 53 - Ukrainian |
Training Data |
Common Voice train and validation datasets |
Metrics |
Word Error Rate (WER) |
Test WER |
32.29% |
License |
apache - 2.0 |