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README.md
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- WER: 0.257040856802856
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- CER: 0.07025001801282513
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##
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## Training procedure
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- WER: 0.257040856802856
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- CER: 0.07025001801282513
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## Installation
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Install the following libraries on top of HuggingFace Transformers for the supports of language model.
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```
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pip install pyctcdecode
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pip install https://github.com/kpu/kenlm/archive/master.zip
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```
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## Usage
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**Approach 1:** Using HuggingFace's pipeline, this will cover everything end-to-end from raw audio input to text output.
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```python
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from transformers import pipeline
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# Load the model
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pipe = pipeline(model="vitouphy/wav2vec2-xls-r-300m-khmer")
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# Process raw audio
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output = pipe("sound_file.wav", chunk_length_s=10, stride_length_s=(4, 2))
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```
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**Approach 2:** More custom way to predict phonemes.
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```python
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from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
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import librosa
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import torch
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# load model and processor
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processor = Wav2Vec2Processor.from_pretrained("vitouphy/wav2vec2-xls-r-300m-khmer")
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model = Wav2Vec2ForCTC.from_pretrained("vitouphy/wav2vec2-xls-r-300m-khmer")
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# Read and process the input
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speech_array, sampling_rate = librosa.load("sound_file.wav", sr=16_000)
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inputs = processor(speech_array, sampling_rate=16_000, return_tensors="pt", padding=True)
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with torch.no_grad():
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logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
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predicted_ids = torch.argmax(logits, axis=-1)
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predicted_sentences = processor.batch_decode(predicted_ids)
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print(predicted_sentences)
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```
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## Intended uses & limitations
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The data used for this model is only around 4 hours of recordings.
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- We split into 80/10/10. Hence, the training hour is 3.2 hours, which is very very small.
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- Yet, its performance is not too bad. Quite interesting for such small dataset, actually. You can try it out.
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- Its limitation is:
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- Rare characters, e.g. ឬស្សី ឪឡឹក
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- Speech needs to be clear and articulate.
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- More data to cover more vocabulary and character may help improve this system.
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## Training procedure
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