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---
license: cc-by-4.0
language:
- qu
metrics:
- cer
pipeline_tag: automatic-speech-recognition
datasets:
- ivangtorre/second_americas_nlp_2022
tags:
- audio
- automatic-speech-recognition
- speech
- qu
- xlsr-fine-tuning
model-index:
- name: Wav2Vec2 XLSR 300M Quechua Model by M Romero and Ivan G Torre
results:
- task:
name: Speech Recognition
type: automatic-speech-recognition
dataset:
name: Americas NLP 2022 qu
type: second_americas_nlp_2022
args: qu
metrics:
- name: Test CER
type: cer
value: 16.02
---
This model was finetuned from a Wav2vec2.0 XLS-R model: 300M with the Quechua train parition of the Americas NLP 2022 dataset. This challenge took place during NeurIPSS 2022.
## Example of usage
The model can be used directly (without a language model) as follows:
```python
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
import torch
import torchaudio
# load model and processor
processor = Wav2Vec2Processor.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-quechua")
model = Wav2Vec2ForCTC.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-quechua")
# Pat to wav file
pathfile = "/path/to/wavfile"
# Load and normalize the file
wav, curr_sample_rate = sf.read(pathfile, dtype="float32")
feats = torch.from_numpy(wav).float()
with torch.no_grad():
feats = F.layer_norm(feats, feats.shape)
feats = torch.unsqueeze(feats, 0)
logits = model(feats).logits
# take argmax and decode
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(predicted_ids)
print("HF prediction: ", transcription)
```
This code snipnet shows how to Evaluate the wav2vec2-xlsr-300m-quechua in [Second Americas NLP 2022 Quechua dev set](https://huggingface.co/datasets/ivangtorre/second_americas_nlp_2022)
```python
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import torch
from jiwer import cer
import torch.nn.functional as F
librispeech_eval = load_dataset("ivangtorre/second_americas_nlp_2022", split="validation")
model = Wav2Vec2ForCTC.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-quechua")
processor = Wav2Vec2Processor.from_pretrained("ivangtorre/wav2vec2-xlsr-300m-quechua")
def map_to_pred(batch):
wav = batch["audio"][0]["array"]
feats = torch.from_numpy(wav).float()
feats = F.layer_norm(feats, feats.shape) # Normalization performed during finetuning
feats = torch.unsqueeze(feats, 0)
logits = model(feats).logits
predicted_ids = torch.argmax(logits, dim=-1)
batch["transcription"] = processor.batch_decode(predicted_ids)
return batch
result = librispeech_eval.map(map_to_pred, batched=True, batch_size=1)
print("CER:", cer(result["source_processed"], result["transcription"]))
```
## Citation
```bibtex
@misc{grosman2021xlsr-1b-russian,
title={Fine-tuned {XLS-R} 1{B} model for speech recognition in {R}ussian},
author={Grosman, Jonatas},
howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-xls-r-1b-russian}},
year={2022}
}
``` |