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README.md
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---
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language: ary
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metrics:
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- wer
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tags:
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- audio
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- automatic-speech-recognition
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- speech
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- xlsr-fine-tuning-week
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license: apache-2.0
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model-index:
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- name: XLSR Wav2Vec2 Moroccan Arabic dialect by Boumehdi
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results:
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- task:
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name: Speech Recognition
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type: automatic-speech-recognition
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metrics:
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- name: Test WER
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type: wer
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value: 0.496
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---
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# Wav2Vec2-Large-XLSR-53-Moroccan
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Fine-tuned [othrif/wav2vec2-large-xlsr-moroccan](https://huggingface.co/othrif/wav2vec2-large-xlsr-moroccan) on 6 hours of labelled speech
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## Usage
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The model can be used directly (without a language model) as follows:
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```python
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import librosa
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import torch
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from transformers import Wav2Vec2CTCTokenizer, Wav2Vec2ForCTC, Wav2Vec2Processor, TrainingArguments, Wav2Vec2FeatureExtractor, Trainer
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tokenizer = Wav2Vec2CTCTokenizer("./vocab.json", unk_token="[UNK]", pad_token="[PAD]", word_delimiter_token="|")
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processor = Wav2Vec2Processor.from_pretrained('boumehdi/wav2vec2-large-xlsr-moroccan-darija-v1', tokenizer=tokenizer)
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model=Wav2Vec2ForCTC.from_pretrained('boumehdi/wav2vec2-large-xlsr-moroccan-darija-v1')
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# load the audio data (use your own wav file here!)
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input_audio, sr = librosa.load('file.wav', sr=16000)
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# tokenize
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input_values = processor(input_audio, return_tensors="pt", padding=True).input_values
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# retrieve logits
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logits = model(input_values).logits
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tokens=torch.argmax(logits, axis=-1)
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# decode using n-gram
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transcription = tokenizer.batch_decode(tokens)
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# print the output
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print(transcription)
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```
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## Evaluation
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**Test Result**: 49.68
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