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---
language: ary
metrics:
- wer
tags:
- audio
- automatic-speech-recognition
- speech
- xlsr-fine-tuning-week
license: apache-2.0
model-index:
- name: XLSR Wav2Vec2 Moroccan Arabic dialect by Boumehdi
results:
- task:
name: Speech Recognition
type: automatic-speech-recognition
metrics:
- name: Test WER
type: wer
value: 0.09
---
# Wav2Vec2-Large-XLSR-53-Moroccan-Darija
**wav2vec2-large-xlsr-53** fine-tuned on 120 hours of labeled Darija Audios
## Usage
The model can be used directly as follows:
```python
import librosa
import torch
from transformers import Wav2Vec2CTCTokenizer, Wav2Vec2ForCTC, Wav2Vec2Processor, TrainingArguments, Wav2Vec2FeatureExtractor, Trainer
tokenizer = Wav2Vec2CTCTokenizer("./vocab.json", unk_token="[UNK]", pad_token="[PAD]", word_delimiter_token="|")
processor = Wav2Vec2Processor.from_pretrained('boumehdi/wav2vec2-large-xlsr-moroccan-darija', tokenizer=tokenizer)
model=Wav2Vec2ForCTC.from_pretrained('boumehdi/wav2vec2-large-xlsr-moroccan-darija')
# load the audio data (use your own wav file here!)
input_audio, sr = librosa.load('file.wav', sr=16000)
# tokenize
input_values = processor(input_audio, return_tensors="pt", padding=True).input_values
# retrieve logits
logits = model(input_values).logits
tokens=torch.argmax(logits, axis=-1)
# decode using n-gram
transcription = tokenizer.batch_decode(tokens)
# print the output
print(transcription)
```
Here's the output: قالت ليا هاد السيد هادا ما كاينش بحالو
email: [email protected]
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