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1from transformers import AutoFeatureExtractor, Wav2Vec2BertModel
2import torch
3from datasets import load_dataset
4
5dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation")
6dataset = dataset.sort("id")
7sampling_rate = dataset.features["audio"].sampling_rate
8
9processor = AutoProcessor.from_pretrained("UmarRamzan/w2v2-bert-ngram-urdu")
10model = Wav2Vec2BertModel.from_pretrained("UmarRamzan/w2v2-bert-ngram-urdu")
11
12# audio file is decoded on the fly
13inputs = processor(dataset[0]["audio"]["array"], sampling_rate=sampling_rate, return_tensors="pt")
14with torch.no_grad():
15 outputs = model(**inputs)