File size: 5,119 Bytes
055c1d3 8420222 e591813 055c1d3 1910bce 6f2fece 1910bce 6f2fece 1910bce 6f2fece 1910bce 6f2fece 1910bce e401779 1910bce 6f2fece 1910bce 6f2fece ef242c5 e401779 ef242c5 6f2fece e401779 ef242c5 1910bce 6f2fece 236e9ef ef242c5 1910bce 6f2fece 1910bce |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 |
---
license: apache-2.0
datasets:
- 012shin/fake-audio-detection-augmented
language:
- en
metrics:
- accuracy
- f1
- recall
- precision
base_model:
- MIT/ast-finetuned-audioset-10-10-0.4593
pipeline_tag: audio-classification
library_name: transformers
tags:
- audio
- audio-classification
- fake-audio-detection
- ast
widget:
- text: "Upload an audio file to check if it's real or synthetic"
inference:
parameters:
sampling_rate: 16000
audio_channel: "mono"
model-index:
- name: ast-fakeaudio-detector
results:
- task:
type: audio-classification
name: Audio Classification
dataset:
name: fake-audio-detection-augmented
type: 012shin/fake-audio-detection-augmented
metrics:
- type: accuracy
value: 0.9662
- type: f1
value: 0.9710
- type: precision
value: 0.9692
- type: recall
value: 0.9728
---
# AST Fine-tuned for Fake Audio Detection
This model is a binary classification head fine-tuned version of [MIT/ast-finetuned-audioset-10-10-0.4593](https://huggingface.co/MIT/ast-finetuned-audioset-10-10-0.4593) for detecting fake/synthetic audio. The original AST (Audio Spectrogram Transformer) classification head was replaced with a binary classification layer optimized for fake audio detection.
## Model Description
- **Base Model**: MIT/ast-finetuned-audioset-10-10-0.4593 (AST pretrained on AudioSet)
- **Task**: Binary classification (fake/real audio detection)
- **Input**: Audio converted to Mel spectrogram (128 mel bins, 1024 time frames)
- **Output**: Binary prediction (0: real audio, 1: fake audio)
- **Training Hardware**: 2x NVIDIA T4 GPUs
## Training Configuration
```python
{
'learning_rate': 1e-5,
'weight_decay': 0.01,
'n_iterations': 1500,
'batch_size': 16,
'gradient_accumulation_steps': 8,
'validate_every': 500,
'val_samples': 5000
}
```
## Dataset Distribution
The model was trained on a filtered dataset with the following class distribution:
```
Training Set:
- Fake Audio (0): 29,089 samples (53.97%)
- Real Audio (1): 24,813 samples (46.03%)
Test Set:
- Fake Audio (0): 7,229 samples (53.64%)
- Real Audio (1): 6,247 samples (46.36%)
```
## Model Performance
Final metrics on validation set:
- Accuracy: 0.9662 (96.62%)
- F1 Score: 0.9710 (97.10%)
- Precision: 0.9692 (96.92%)
- Recall: 0.9728 (97.28%)
# Usage Guide
## Model Usage
```python
import torch
import torchaudio
import soundfile as sf
import numpy as np
from transformers import AutoFeatureExtractor, AutoModelForAudioClassification
# Load model and move to available device
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
model_name = "WpythonW/ast-fakeaudio-detector"
extractor = AutoFeatureExtractor.from_pretrained(model_name)
model = AutoModelForAudioClassification.from_pretrained(model_name).to(device)
model.eval()
# Process multiple audio files
audio_files = ["audio1.wav", "audio2.mp3", "audio3.ogg"]
processed_batch = []
for audio_path in audio_files:
# Load audio file
audio_data, sr = sf.read(audio_path)
# Convert stereo to mono if needed
if len(audio_data.shape) > 1 and audio_data.shape[1] > 1:
audio_data = np.mean(audio_data, axis=1)
# Resample to 16kHz if needed
if sr != 16000:
waveform = torch.from_numpy(audio_data).float()
if len(waveform.shape) == 1:
waveform = waveform.unsqueeze(0)
resample = torchaudio.transforms.Resample(
orig_freq=sr,
new_freq=16000
)
waveform = resample(waveform)
audio_data = waveform.squeeze().numpy()
processed_batch.append(audio_data)
# Prepare batch input
inputs = extractor(
processed_batch,
sampling_rate=16000,
padding=True,
return_tensors="pt"
)
inputs = {k: v.to(device) for k, v in inputs.items()}
# Get predictions
with torch.no_grad():
logits = model(**inputs).logits
probabilities = torch.nn.functional.softmax(logits, dim=-1)
# Process results
for filename, probs in zip(audio_files, probabilities):
fake_prob = float(probs[0].cpu())
real_prob = float(probs[1].cpu())
prediction = "FAKE" if fake_prob > real_prob else "REAL"
print(f"\nFile: {filename}")
print(f"Fake probability: {fake_prob:.2%}")
print(f"Real probability: {real_prob:.2%}")
print(f"Verdict: {prediction}")
```
## Limitations
Important considerations when using this model:
1. The model works best with 16kHz audio input
2. Performance may vary with different types of audio manipulation not present in training data
3. Very short audio clips (<1 second) might not provide reliable results
4. The model should not be used as the sole determiner for real/fake audio detection
## Training Details
The training process involved:
1. Loading the base AST model pretrained on AudioSet
2. Replacing the classification head with a binary classifier
3. Fine-tuning on the fake audio detection dataset for 1500 iterations
4. Using gradient accumulation (8 steps) with batch size 16
5. Implementing validation checks every 500 steps |