--- language: Bengali datasets: - OpenSLR metrics: - wer tags: - bn - audio - automatic-speech-recognition - speech license: cc-by-sa-4.0 model-index: - name: XLSR Wav2Vec2 Bengali by Arijit results: - task: name: Speech Recognition type: automatic-speech-recognition dataset: name: OpenSLR type: OpenSLR args: ben metrics: - name: Test WER type: wer value: 32.45 --- # Wav2Vec2-Large-XLSR-Bengali Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) Bengali using a subset of 40,000 utterances from [Bengali ASR training data set containing ~196K utterances](https://www.openslr.org/53/). Tested WER using ~4200 held out from training. When using this model, make sure that your speech input is sampled at 16kHz. Train Script can be Found at : train.py Data Prep Notebook : https://colab.research.google.com/drive/1JMlZPU-DrezXjZ2t7sOVqn7CJjZhdK2q?usp=sharing Inference Notebook : https://colab.research.google.com/drive/1uKC2cK9JfUPDTUHbrNdOYqKtNozhxqgZ?usp=sharing ## Usage The model can be used directly (without a language model) as follows: ```python import torch import torchaudio from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor processor = Wav2Vec2Processor.from_pretrained("arijitx/wav2vec2-large-xlsr-bengali") model = Wav2Vec2ForCTC.from_pretrained("arijitx/wav2vec2-large-xlsr-bengali") # model = model.to("cuda") resampler = torchaudio.transforms.Resample(TEST_AUDIO_SR, 16_000) def speech_file_to_array_fn(batch): speech_array, sampling_rate = torchaudio.load(batch) speech = resampler(speech_array).squeeze().numpy() return speech speech_array = speech_file_to_array_fn("test_file.wav") inputs = processor(speech_array, sampling_rate=16_000, return_tensors="pt", padding=True) with torch.no_grad(): logits = model(inputs.input_values).logits predicted_ids = torch.argmax(logits, dim=-1) preds = processor.batch_decode(predicted_ids)[0] print(preds.replace("[PAD]","")) ``` **Test Result**: WER on ~4200 utterance : 32.45 %