đ Wav2Vec2-Large-XLSR-53-Mongolian
This model is fine - tuned from facebook/wav2vec2-large-xlsr-53 on Mongolian using the Common Voice dataset. When using this model, ensure that your speech input is sampled at 16kHz.
Model Information
Property |
Details |
Language |
Mongolian |
Datasets |
Common Voice |
Metrics |
WER (Word Error Rate) |
Tags |
Audio, Automatic Speech Recognition, Speech, XLSR - Fine - Tuning - Week |
License |
Apache 2.0 |
Model Name |
Mongolian XLSR Wav2Vec2 Large 53 by Anton Lozhkov |
Task |
Speech Recognition (Automatic Speech Recognition) |
Dataset in Evaluation |
Common Voice mn |
Test WER |
38.53 |
đ Quick Start
⨠Features
This model is a fine - tuned version of facebook/wav2vec2-large-xlsr-53
on the Mongolian language, enabling high - quality automatic speech recognition for Mongolian.
đĻ Installation
No specific installation steps are provided in the original document.
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", "mn", split="test[:2%]")
processor = Wav2Vec2Processor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-mongolian")
model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-mongolian")
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
The following code shows how to evaluate the model on the Mongolian 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/mn.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-mongolian")
model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-mongolian")
model.to("cuda")
cv_test = pd.read_csv("cv-corpus-6.1-2020-12-11/mn/test.tsv", sep='\t')
clips_path = "cv-corpus-6.1-2020-12-11/mn/clips/"
def clean_sentence(sent):
sent = sent.lower()
sent = "".join(ch if ch.isalpha() 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: 38.53 %
đ Documentation
Training
The Common Voice train
and validation
datasets were used for training.
đ License
This model is licensed under the Apache 2.0 license.