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--- |
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license: cc-by-4.0 |
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language: |
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- gvc |
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metrics: |
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- cer |
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pipeline_tag: automatic-speech-recognition |
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datasets: |
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- ivangtorre/second_americas_nlp_2022 |
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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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- kotiria |
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- xlsr-fine-tuning |
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model-index: |
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- name: Wav2Vec2 XLSR 300M Kotiria Model by M Romero and Ivan G Torre |
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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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dataset: |
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name: Americas NLP 2022 Kotiria |
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type: second_americas_nlp_2022 |
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args: Kotiria |
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metrics: |
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- name: Test CER |
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type: cer |
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value: 36.00 |
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--- |
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This model was finetuned from a Wav2vec2.0 XLS-R model: 300M with the Kotiria train parition of the Americas NLP 2022 dataset. This challenge took place during NeurIPSS 2022. |
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## Example of 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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from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC |
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import torch |
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import torchaudio |
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# load model and processor |
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processor = Wav2Vec2Processor.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-kotiria") |
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model = Wav2Vec2ForCTC.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-kotiria") |
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# Pat to wav file |
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pathfile = "/path/to/wavfile" |
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# Load and normalize the file |
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wav, curr_sample_rate = sf.read(pathfile, dtype="float32") |
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feats = torch.from_numpy(wav).float() |
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with torch.no_grad(): |
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feats = F.layer_norm(feats, feats.shape) |
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feats = torch.unsqueeze(feats, 0) |
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logits = model(feats).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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print("HF prediction: ", transcription) |
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``` |
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This code snipnet shows how to Evaluate the wav2vec2-xlsr-300m-kotiria in [Second Americas NLP 2022 Kotiria dev set](https://huggingface.co/datasets/ivangtorre/second_americas_nlp_2022) |
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```python |
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from datasets import load_dataset |
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor |
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import torch |
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from jiwer import cer |
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import torch.nn.functional as F |
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from datasets import load_dataset |
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import soundfile as sf |
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americasnlp = load_dataset("ivangtorre/second_americas_nlp_2022", "kotiria", split="dev") |
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kotiria = americasnlp.filter(lambda language: language['subset']=='kotiria') |
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model = Wav2Vec2ForCTC.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-kotiria") |
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processor = Wav2Vec2Processor.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-kotiria") |
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def map_to_pred(batch): |
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wav = batch["audio"][0]["array"] |
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feats = torch.from_numpy(wav).float() |
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feats = F.layer_norm(feats, feats.shape) # Normalization performed during finetuning |
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feats = torch.unsqueeze(feats, 0) |
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logits = model(feats).logits |
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predicted_ids = torch.argmax(logits, dim=-1) |
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batch["transcription"] = processor.batch_decode(predicted_ids) |
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return batch |
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result = kotiria.map(map_to_pred, batched=True, batch_size=1) |
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print("CER:", cer(result["source_processed"], result["transcription"])) |
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``` |
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## Citation |
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```bibtex |
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@article{romero2024asr, |
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title={ASR advancements for indigenous languages: Quechua, Guarani, Bribri, Kotiria, and Wa'ikhana}, |
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author={Romero, Monica and Gomez, Sandra and Torre, Iv{\'a}n G}, |
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journal={arXiv preprint arXiv:2404.08368}, |
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year={2024} |
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} |
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``` |