File size: 1,587 Bytes
361f4f3 b118a51 361f4f3 |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 |
---
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: 0.496
---
# Wav2Vec2-Large-XLSR-53-Moroccan
Fine-tuned [othrif/wav2vec2-large-xlsr-moroccan](https://huggingface.co/othrif/wav2vec2-large-xlsr-moroccan) on 6 hours of labelled darija Audios
## Usage
The model can be used directly (without a language model) 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-v1', tokenizer=tokenizer)
model=Wav2Vec2ForCTC.from_pretrained('boumehdi/wav2vec2-large-xlsr-moroccan-darija-v1')
# 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)
```
## Evaluation
**Test Result**: 49.68
|