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import gradio as gr |
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import torch |
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import soundfile as sf |
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import os |
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import matplotlib.pyplot as plt |
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import numpy as np |
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import os |
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import soundfile as sf |
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import torch |
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import torch.nn as nn |
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import torch.nn.functional as F |
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from torch.utils.data import Dataset, DataLoader |
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor, Wav2Vec2ForSequenceClassification |
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from sklearn.model_selection import train_test_split |
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import re |
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from collections import Counter |
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from sklearn.metrics import classification_report |
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model = Wav2Vec2ForSequenceClassification.from_pretrained("facebook/wav2vec2-base-960h", num_labels=2).to(device) |
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model_path = "dysarthria_classifier12.pth" |
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if os.path.exists(model_path): |
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print(f"Loading saved model {model_path}") |
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model.load_state_dict(torch.load(model_path)) |
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu") |
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processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h") |
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def predict(file_path): |
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max_length = 100000 |
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model.eval() |
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with torch.no_grad(): |
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wav_data, _ = sf.read(file_path.name) |
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inputs = processor(wav_data, sampling_rate=16000, return_tensors="pt", padding=True) |
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input_values = inputs.input_values.squeeze(0) |
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if max_length - input_values.shape[-1] > 0: |
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input_values = torch.cat([input_values, torch.zeros((max_length - input_values.shape[-1],))], dim=-1) |
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else: |
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input_values = input_values[:max_length] |
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input_values = input_values.unsqueeze(0).to(device) |
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inputs = {"input_values": input_values} |
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logits = model(**inputs).logits |
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logits = logits.squeeze() |
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predicted_class_id = torch.argmax(logits, dim=-1).item() |
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return predicted_class_id |
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iface = gr.Interface(fn=predict, inputs="file", outputs="text") |
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iface.launch() |
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