Wav2vec2 Large Xlsr Egyptian

🚀 Wav2Vec2-Large-XLSR-53-Egyptian-Arabic

This model is a fine - tuned version of facebook/wav2vec2-large-xlsr-53 for Egyptian Arabic, leveraging the arabicspeech.org MGB - 3. Ensure your speech input is sampled at 16kHz when using this model.

🚀 Quick Start

This model is a fine - tuned version of facebook/wav2vec2-large-xlsr-53 for Egyptian Arabic, using the arabicspeech.org MGB - 3. When using this model, make sure your speech input is sampled at 16kHz.

✨ Features

• Language: Egyptian Arabic
• Based on: facebook/wav2vec2-large-xlsr-53
• Dataset: arabicspeech.org MGB - 3

📦 Installation

No specific installation steps are provided in the original document.

💻 Usage Examples

Basic Usage

import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor

test_dataset = load_dataset("common_voice", "ar", split="test[:2%]")

processor = Wav2Vec2Processor.from_pretrained("othrif/wav2vec2-large-xlsr-egyptian")
model = Wav2Vec2ForCTC.from_pretrained("othrif/wav2vec2-large-xlsr-egyptian")

resampler = torchaudio.transforms.Resample(48_000, 16_000)

# Preprocessing the datasets.
# We need to read the audio files as arrays
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

import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re

test_dataset = load_dataset("common_voice", "ar", split="test") 
wer = load_metric("wer")

processor = Wav2Vec2Processor.from_pretrained("othrif/wav2vec2-large-xlsr-egyptian") 
model = Wav2Vec2ForCTC.from_pretrained("othrif/wav2vec2-large-xlsr-egyptian")
model.to("cuda")

chars_to_ignore_regex = '[\؛\—\_get\«\»\ـ\ـ\,\?\.\!\-\;\:\"\“\%\‘\”\�\#\،\☭,\؟]'
resampler = torchaudio.transforms.Resample(48_000, 16_000)

# Preprocessing the datasets.
# We need to read the audio files as arrays
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)

# Preprocessing the datasets.
# We need to read the audio files as arrays
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"])))

📚 Documentation

Test Result

The model achieved a Word Error Rate (WER) of 55.2 on the test dataset.

Training

The Common Voice train and validation datasets were used for training. The training script can be found here.

🔧 Technical Details

The model is based on the fine - tuning of facebook/wav2vec2-large-xlsr-53 on Egyptian Arabic data from arabicspeech.org MGB - 3.

📄 License

This model is licensed under the Apache 2.0 license.

📊 Model Information