DeepFense: A Unified, Modular, and Extensible Framework for Robust Deepfake Audio Detection
Authors: Yassine El Kheir, Arnab Das, Yixuan Xiao, Xin Wang, Feidi Kallel, Enes Erdem Erdogan, Ngoc Thang Vu, Tim Polzehl, +1 more
Organizations: German Research Center for Artificial Intelligence (DFKI), Germany · University of Stuttgart, Germany · National Institute of Informatics, Japan · Technical University of Berlin, Germany
Speech deepfake detection is a well-established research field with different models, datasets, and training strategies. However, the lack of standardized implementations and evaluation protocols limits reproducibility, benchmarking, and comparison across studies. In this work, we present DeepFense, a comprehensive, open-source PyTorch toolkit integrating the latest architectures, loss functions, and augmentation pipelines, alongside over 100 recipes. Using DeepFense, we conducted a large-scale evaluation of more than 400 models. Our findings reveal that while carefully curated training data improves cross-domain generalization, the choice of pre-trained front-end feature extractor dominates overall performance variance. Crucially, we show severe biases in high-performing models regarding audio quality, speaker gender, and language. DeepFense is expected to facilitate real-world deployment with the necessary tools to address equitable training data selection and front-end fine-tuning.
Figures & tables
Figure 1: System architecture of the DeepFense framework, illustrating the configuration-driven data pipeline, modular model engine (front-end, back-end, loss), unified training loop, and evaluation components, with built-in logging and experiment tracking.
ITW
ASV5
CodecFake
Average
Back-End
Orig.
Ours
Orig.
Ours
Orig.
Ours
Orig.
Ours
AASIST
10.46
7.18
19.24
18.94
38.79
34.36
22.83
20.16
MLP
8.13
7.57
18.76
15.43
35.76
34.44
20.88
19.15
Nes2Net
7.06
7.73
22.06
24.19
39.34
36.21
22.82
22.71
TCM
7.79
6.98
18.85
19.40
36.00
34.77
20.88
20.39
Table 1: EER (%) for Wav2Vec2 systems trained on ASV19 , evaluated on out-of-domain test sets before and after applying DeepFense. Best result per back-end in bold .
Back-End
Orig.
Ours
AASIST
7.12
6.64
Nes2Net
6.13
5.53
BiCrossMamba-ST
6.01
6.34
Table 2: EER (%) for Wav2Vec2 systems trained on ASV5 , evaluated on the ASV5 test set.
Back-End
Orig.
Ours
Nes2Net
0.49
0.55
AASIST
1.03
0.77
MLP
0.95
0.91
TCM
0.34
0.40
BiCrossMamba-ST
0.75
0.44
Table 3: EER (%) for EAT trained on EnvSDD and evaluated on CodecFake-A3 test set.
Figure 2: Boxplots of detection systems based on their EERs on the test set (x-axis). Each box groups 16 systems using the same front-end (from left to right: Wav2Vec2, WavLM, HuBERT, and EAT) but varied training data and back-ends. The three lines correspond to mean EERs of systems using either the ASV19 (blue), ADD23 (red), or CodecFake (green) training sets.
Figure 3: Mean EER (%) per front-end and evaluation benchmark, averaged over all back-ends and training sets.
Front-End
Mean EER (%)
Back-End
Mean EER (%)
Wav2Vec2
25.5
AASIST
30.3
EAT
31.4
Nes2Net
30.7
WavLM
32.4
TCM
30.8
Hubert
33.6
MLP
31.1
Table 4: Mean EER (%) for each front-end (averaged over all back-ends & training sets) and each back-end (averaged over all front-ends & training sets). ↓
Figure 4: Mean EER analysis across training datasets, front-ends, and back-ends. (a) Front-end perspective. (b) Back-end perspective.
Training Data
Mean EER
Std
CodecFake
22.3
0.047
HABLA
25.2
0.034
ASV5
26.9
0.029
ASV19
28.2
0.052
PartialSpoof
31.0
0.056
ADD23
50.8
0.063
Table 5: Macro-average EER across all 13 evaluation datasets, per training set (mean ± std over all front-end × back-end).
Figure 5: EER on non-speech deepfake detection.
Figure 6: EER Sensitivity to Audio Quality Across Systems trained on ASV19, and CodecFake
GARBE
Training
Front-End
Back-End
PESQ
NISQA-MOS
Gender
Language
ASV19
EAT
AASIST
0.0135
0.0196
0.1460
0.1886
Hubert
AASIST
0.1735
0.1626
0.1160
0.2397
Wav2Vec2
AASIST
0.1880
0.2396
0.1940
0.2147
Wav2Vec2
TCM
0.1769
0.2056
0.2110
0.2630
WavLM
AASIST
0.2579
0.2258
0.1450
0.2841
Table 6: GARBE fairness score of each system, computed over audio quality (PESQ and NISQA-MOS), gender, and languages.
EER
Training
Front-End
Back-End
F
M
Δ
ASV19
EAT
AASIST
0.584
0.578
+0.005
Hubert
AASIST
0.447
0.321
+0.126
Wav2Vec2
AASIST
0.257
0.183
+0.075
Wav2Vec2
TCM
0.275
0.200
+0.074
WavLM
AASIST
0.455
0.347
+0.108
Table 7: EER by gender for each system. F = Female, M = Male, Δ=EERF−EERM .
Figure 7: EER (%) per language for all system configurations trained on ASVspoof 2019 (left) and CodecFake (right). Languages are sorted by mean EER (lowest at top). Each dot is one system; colour encodes front-end; marker shape encodes back-end. Grey bars show the min–max EER range across all configurations.
Audio deepfake detection models determine whether speech is genuine or artificially generated, but high overall accuracy can mask substantial performance disparities across demographic groups. In this work, we investigate gender bias in audio deepfake detection using the ASVspoof5 dataset. We use ASVspoof5 under a controlled custom split designed to isolate gender-composition effects. We train attack-specific models on nine training sets with different gender compositions, ranging from female-only to male-only. We use a ResNet18 classifier with LogSpectrogram and WavLM-Base+ features, and we evaluated six post-hoc threshold calibration methods. Experimental results show that training data composition strongly predicts bias direction, with the underrepresented gender performing worse at test time. WavLM-Base+ features are shown to produce gender performance gaps 3.0 to 4.3 times larger than LogSpectrogram under identical training conditions, and balanced training is found to reduce LogSpectrogram bias but leave WavLM bias largely intact. Moreover, all six calibration strategies, including Oracle calibration with full test-set label access, leave the Equal Error Rate gap unchanged at 1.317 pp, confirming that threshold adjustment cannot correct underlying score distribution disparities. Overall, these findings suggest that gender fairness in audio deepfake detection must be addressed at training time, as post-hoc methods can only partially mitigate the resulting disparities
Aishwarya R. Fursule, Vamshi Nallaguntla, Shruti Kshirsagar +1
School of Computing, Wichita State University, Wichita, KS, USA · Institut national de la recherche scientifique (INRS–EMT), Montreal, QC, Canada · INRS-UQO Mixed Research Unit on Cybersecurity, Gatineau, QC, Canada
Speech deepfake detection has expanded in scope with increasingly heterogeneous spoofing mechanisms, including speech synthesis, voice conversion, vocoder reconstruction, and neural-codec resynthesis. The resulting spoofing artifacts can be further shaped by variability in source speech, recording environments, and transmission channels. This variability makes robust generalization across heterogeneous conditions a central requirement for practical detection systems. This report presents Teffic-Audio, a general speech deepfake detection system designed for comprehensive evaluation environment. Teffic-Audio adopts a straightforward detector architecture consisting of a Conformer-based speech encoder, multi-head attentive statistics pooling, and a binary classifier. Rather than relying on additional architectural complexity, the system improves generalization through its training recipe, which integrates multi-source data, attack- and source-balanced sampling, and diverse audio augmentation. Trained only with open-source data, Teffic-Audio achieves a pooled EER of 1.454% on the 14 test sets of Speech-DF-Arena, outperforming all currently public systems on the leaderboard. It also obtains the lowest EER on five individual test sets and shows a favorable performance-complexity trade-off compared with larger leading systems. Overall, Teffic-Audio provides a strong and practical reference system for general speech deepfake detection.
Audio deepfake detection systems often fail to generalize across domains because they rely on features tied to specific attacks or recording conditions. Self-supervised speech models offer rich multi-layer representations, yet existing approaches either use a single layer or fuse all layers indiscriminately, and only reveal layer importance after training. We propose a model-agnostic, two-stage methodology that identifies informative depth zones before any task-specific model is trained. In the first stage, lightweight XGBoost probes evaluate each transformer layer's cross-domain discriminative power, producing a layer ranking. In the second stage, a compact neural classifier fuses only the selected layers through per-layer attention pooling and a shared bottleneck projection, while the backbone remains frozen. Applied across three backbones, the probing reveals two key findings. First, informative layers cluster in depth zones rather than at uniquely optimal positions: within-zone substitutions fall within multi-seed noise, while zone violations degrade performance by up to 5x. Second, the probing produces backbone-specific selections rather than a fixed layer recipe. On XLS-R-300M, four probing-selected layers with 1.34M trainable parameters achieve 4.94 +/- 0.32% equal error rate on In-The-Wild and 5.07% cross-domain average over four shared datasets, a 28% relative improvement over the best prior frozen-backbone result (Xiao and Vu, 2025) using all 25 layers with identical training data.
Marjan Beheshti, Majid Rostami, Bo Chen
Department of Computer Science, Michigan Technological University, Houghton, MI, USA