đ Wav2Vec2-Large-XLSR-53-Sorbian
This model is a fine - tuned version of facebook/wav2vec2-large-xlsr-53 in Sorbian, leveraging the Common Voice. Ensure your speech input is sampled at 16kHz when using this model.
đĻ Information Table
Property |
Details |
Language |
Sorbian (hsb) |
Datasets |
Common Voice |
Tags |
audio, automatic - speech - recognition, speech, xlsr - fine - tuning - week |
License |
apache - 2.0 |
Model Name |
XLSR Wav2Vec2 Sorbian by Adam Montgomerie |
Task |
Speech Recognition (automatic - speech - recognition) |
Dataset |
Common Voice hsb |
Test WER |
41.74 |
đ Quick Start
Fine - tuned facebook/wav2vec2-large-xlsr-53 in Sorbian using the Common Voice. When using this model, make sure that your speech input is sampled at 16kHz.
đģ Usage Examples
Basic Usage
import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
test_dataset = load_dataset("common_voice", "hsb", split="test[:2%]")
processor = Wav2Vec2Processor.from_pretrained("iarfmoose/wav2vec2-large-xlsr-sorbian")
model = Wav2Vec2ForCTC.from_pretrained("iarfmoose/wav2vec2-large-xlsr-sorbian")
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])
đ§ Evaluation
The model can be evaluated as follows on the Sorbian 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", "hsb", split="test")
wer = load_metric("wer")
processor = Wav2Vec2Processor.from_pretrained("iarfmoose/wav2vec2-large-xlsr-sorbian")
model = Wav2Vec2ForCTC.from_pretrained("iarfmoose/wav2vec2-large-xlsr-sorbian")
model.to("cuda")
chars_to_ignore_regex = '[\\\\\\\\\\\\\\\\,\\\\\\\\\\\\\\\\?\\\\\\\\\\\\\\\\.\\\\\\\\\\\\\\\\!\\\\\\\\\\\\\\\\-\\\\\\\\\\\\\\\\;\\\\\\\\\\\\\\\\:\\\\\\\\\\\\\\\\"\\\\\\\\\\\\\\\\â\\\\\\\\\\\\\\\\%\\\\\\\\\\\\\\\\â\\\\\\\\\\\\\\\\â\\\\\\\\\\\\\\\\īŋŊ\\\\\\\\\\\\\\\\â\\\\\\\\\\\\\\\\â\\\\\\\\\\\\\\\\ÂŦ\\\\\\\\\\\\\\\\â
]'
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: 41.74 %
đ Training
The Common Voice train
, validation
datasets were used for training.
The script used for training can be found here
A notebook of the evaluation script can be found here
đ License
This project is licensed under the Apache 2.0 license.