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app.py
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import joblib
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from transformers import AutoFeatureExtractor, Wav2Vec2Model
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import torch
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import librosa
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import numpy as np
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from sklearn.linear_model import LogisticRegression
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import gradio as gr
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import os
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device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
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class CustomWav2Vec2Model(Wav2Vec2Model):
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def __init__(self, config):
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super().__init__(config)
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self.encoder.layers = self.encoder.layers[:9]
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truncated_model = CustomWav2Vec2Model.from_pretrained("facebook/wav2vec2-xls-r-2b")
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class HuggingFaceFeatureExtractor:
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def __init__(self, model, feature_extractor_name):
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self.device = device
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self.feature_extractor = AutoFeatureExtractor.from_pretrained(feature_extractor_name)
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self.model = model
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self.model.eval()
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self.model.to(self.device)
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def __call__(self, audio, sr):
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inputs = self.feature_extractor(
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audio,
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sampling_rate=sr,
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return_tensors="pt",
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padding=True,
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)
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inputs = {k: v.to(self.device) for k, v in inputs.items()}
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with torch.no_grad():
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outputs = self.model(**inputs, output_hidden_states=True)
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return outputs.hidden_states[9]
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FEATURE_EXTRACTOR = HuggingFaceFeatureExtractor(truncated_model, "facebook/wav2vec2-xls-r-2b")
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classifier,scaler, thresh = joblib.load('logreg_margin_pruning_ALL_with_scaler+threshold.joblib')
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def segment_audio(audio, sr, segment_duration):
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segment_samples = int(segment_duration * sr)
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total_samples = len(audio)
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segments = [audio[i:i + segment_samples] for i in range(0, total_samples, segment_samples)]
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return segments
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def process_audio(input_data, segment_duration=10):
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audio, sr = librosa.load(input_data, sr=16000)
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if len(audio.shape) > 1:
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audio = audio[0]
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segments = segment_audio(audio, sr, segment_duration)
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segment_predictions = []
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output_lines = []
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eer_threshold = thresh - 5e5 # small margin error due to feature extractor space differences
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for idx, segment in enumerate(segments):
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features = FEATURE_EXTRACTOR(segment, sr)
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features_avg = torch.mean(features, dim=1).cpu().numpy()
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features_avg = features_avg.reshape(1, -1)
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decision_score = classifier.decision_function(features_avg)
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decision_score_scaled = scaler.transform(decision_score.reshape(-1, 1)).flatten()
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if decision_score_scaled >= eer_threshold:
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pred = 1
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confidence_percentage = decision_score_scaled[0] * 100
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else:
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pred = 0
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confidence_percentage = (1 - decision_score_scaled[0]) * 100
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segment_predictions.append(pred)
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line = f"Segment {idx + 1}: {'Real' if pred == 1 else 'Fake'} (Confidence: {round(confidence_percentage, 2)}%)"
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output_lines.append(line)
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overall_prediction = 1 if sum(segment_predictions) > (len(segment_predictions) / 2) else 0
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overall_line = f"Overall Prediction: {'Real' if overall_prediction == 1 else 'Fake'}"
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output_str = overall_line + "\n" + "\n".join(output_lines)
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return output_str
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def gradio_interface(audio):
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if audio:
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return process_audio(audio)
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else:
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return "please upload an audio file"
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interface = gr.Interface(
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fn=gradio_interface,
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inputs=[gr.Audio(type="filepath", label="Upload Audio")],
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outputs="text",
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title="SOL2 Audio Deepfake Detection Demo",
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description="Upload an audio file to check if it's AI-generated",
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)
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interface.launch(share=True)
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logreg_margin_pruning_ALL_with_scaler+threshold.joblib
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@@ -0,0 +1,3 @@
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version https://git-lfs.github.com/spec/v1
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oid sha256:dbdc1bd864d2b997fc873499f6485891998d5f829af4746064426e942ba580dc
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size 16779
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