Added an use example
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README.md
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The efforts resulting with this model were coordinated by Nikola Ljubešić, the rough manual data alignment was performed by Ivo-Pavao Jazbec, the method for fine automatic data alignment from [Plüss et al.](https://arxiv.org/abs/2010.02810) was applied by Vuk Batanović and Lenka Bajčetić, while the final modelling was performed by Peter Rupnik.
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Initial evaluation on partially noisy data showed the model to achieve a word error rate of 13.68% and a character error rate of 4.56%.
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The efforts resulting with this model were coordinated by Nikola Ljubešić, the rough manual data alignment was performed by Ivo-Pavao Jazbec, the method for fine automatic data alignment from [Plüss et al.](https://arxiv.org/abs/2010.02810) was applied by Vuk Batanović and Lenka Bajčetić, while the final modelling was performed by Peter Rupnik.
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Initial evaluation on partially noisy data showed the model to achieve a word error rate of 13.68% and a character error rate of 4.56%.
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## Usage in `transformers`
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```python
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from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
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from datasets import Audio
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import soundfile as sf
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import torch
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import os
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# load model and tokenizer
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processor = Wav2Vec2Processor.from_pretrained(
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"classla/wav2vec2-xls-r-sabor-hr")
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model = Wav2Vec2ForCTC.from_pretrained("classla/wav2vec2-xls-r-sabor-hr")
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# download the example wav files:
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os.system("curl https://huggingface.co/classla/wav2vec2-xls-r-sabor-hr/raw/main/00020570a.flac.wav")
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# read the wav file as datasets.Audio object
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audio = Audio(sampling_rate=16000).decode_example("00020570a.flac.wav")
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# remove the raw wav file
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os.system("rm 00020570a.flac.wav")
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# tokenize
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input_values = processor(
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audio["array"], return_tensors="pt", padding=True,
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sampling_rate=16000).input_values
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# retrieve logits
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logits = model(input_values).logits
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# take argmax and decode
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.batch_decode(predicted_ids)
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# transcription: ['veliki broj poslovnih subjekata posluje sa minusom velik dio']
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```
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