import argparse import logging import time import torch from transformers import AutoFeatureExtractor, ASTForAudioClassification from datasets import load_dataset import torch_neuronx # ensure Neuron backend is available logging.basicConfig(level=logging.INFO) logger = logging.getLogger(__name__) def main(): parser = argparse.ArgumentParser(description="Run AST (Audio Spectrogram Transformer) on Neuron") parser.add_argument( "--model", type=str, default="MIT/ast-finetuned-audioset-10-10-0.4593", help="AST model name on Hugging Face Hub", ) parser.add_argument("--batch-size", type=int, default=1, help="Batch size") args = parser.parse_args() torch.set_default_dtype(torch.float32) torch.manual_seed(42) # Load dataset and extract features dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation") dataset = dataset.sort("id") sampling_rate = dataset.features["audio"].sampling_rate feature_extractor = AutoFeatureExtractor.from_pretrained(args.model) inputs = feature_extractor( dataset[0]["audio"]["array"], sampling_rate=sampling_rate, return_tensors="pt" ) # Load AST model model = ASTForAudioClassification.from_pretrained( args.model, torch_dtype=torch.float32, attn_implementation="eager" ) model.eval() # Pre-run once to fix shapes before compilation with torch.no_grad(): logits = model(**inputs).logits # Compile forward pass model.forward = torch.compile(model.forward, backend="neuron", fullgraph=True) # Warmup warmup_start = time.time() with torch.no_grad(): _ = model(**inputs) warmup_time = time.time() - warmup_start # Actual run run_start = time.time() with torch.no_grad(): logits = model(**inputs).logits run_time = time.time() - run_start # Decode result predicted_class_ids = torch.argmax(logits, dim=-1).item() predicted_label = model.config.id2label[predicted_class_ids] logger.info("Warmup: %.2f s, Run: %.4f s", warmup_time, run_time) logger.info("Predicted label: %s", predicted_label) if __name__ == "__main__": main() """ Works """