Existing audio deepfake detection (ADD) methods frequently exhibit limited generalization, as fixed bottleneck architectures fail to adapt to the heterogeneous information densities inherent in diverse spoofing artifacts. Moreover, standard augmentation often leads to a pathological robustness-accuracy trade-off. In this paper, we propose a unified ADD framework to address these challenges. Our approach integrates: (1) an Adaptive Audio Feature Integration (AAFI) module that dynamically selects optimal latent dimensions via a compression pool and a stochastic exploration mechanism, ensuring flexible representations across multiple types of dynamic attacks; and (2) a Mel-Band Adversarial Perturbation (Mel-BAP) strategy that applies targeted regularization in the mel-spectrogram domain to encourage the model to learn invariant decision boundaries. Extensive evaluations on benchmarks including ADD 2022, FakeOrReal, and In-the-Wild demonstrate that our model based on Whisper-small achieves a state-of-the-art average Equal Error Rate (EER) of 1.51%. Compared to existing SOTA models such as the 3-billion-parameter Resemble-Detect-3B-Omni, our framework reduces the parameter count by up to 91.6% while achieving a 60.7% improvement in average EER. In a separate cross-lingual evaluation, the Whisper-base configuration achieves EERs of 3.91-5.65% on five unseen languages. The framework also maintains strong resilience to real-world acoustic distortions. This work provides a scalable and efficient "plug-in" solution for enhancing trust and security in the audio Web ecosystem.
Figures & tables
Figure 1: Schematic of the robust audio forgery detection pipeline. Left: Mel-BAP generates band-selective perturbations via PGD as targeted adversarial regularization. Center: AAFI implements an adaptive information bottleneck, using a router to dynamically select compressed features from heterogeneous pools. Right: Fusion of multi-resolution features for sequence-to-sequence decoding.
Data Group
top- k number
Whisper-base
top-2
top-3
top-4
top-5
TTS
3.91
0.85
0.83
0.99
1.18
VC
4.85
1.88
1.67
2.11
3.32
NV
8.44
3.44
3.50
4.34
6.07
Average
5.73
2.06
2.00
2.48
3.52
Table 1: Performance Comparison of Adaptive Routing (top- k ) in the AAFI Module versus Whisper base. EER (%) results (lower is better).
Model
BD-EN
BD-CN
BD-UT
Average
w/o Mel-BAP
1.94
1.68
1.19
1.60
Mel-BAP Only
1.35
1.20
0.91
1.14
AAFI + Mel-BAP
1.01
0.82
0.53
0.79
Table 2: Ablation study on the effectiveness of Mel-BAP across SpeechFake BD subsets. Lower EER (%) indicates better performance.
Strategy
Frequency Band
Mel-bin Indices
Clean EER (%)
(a) Mel-BAP (Ours)
4–8 kHz
[60, 80]
0.91
(b) Full-band PGD
0–8 kHz
[0, 80]
1.15
(c) Low-band PGD
0–4 kHz
[0, 60]
1.28
Table 3: Ablation of adversarial frequency band constraints on the BD-UT dataset (Clean EER).
Model
w/o Mel-BAP
with Mel-BAP
Global Aug
Clean
Attack
Clean
Attack
Clean
Attack
Base
1.19
3.57
0.91
2.44
2.33
4.68
Small
0.95
3.18
0.87
2.25
1.65
4.01
Table 4: Robustness comparison against standard augmentation under real-world distortions. EER (%) is reported on BD-UT.
Model ID
# Params
ADD 2022 (T1)
FakeOrReal
In-the-Wild
ASV2019
ASV2021 LA
Average
Literature and Commercial SOTA (Speech-DF-Arena 2025)
Resemble-Detect-3B-Omni [ 40 ]
3,000 M
9.96
2.34
1.35
2.53
3.02
3.84
DLMSL-SpeakSure-v0.1 [ 40 ]
658.6 M
15.14
0.27
1.28
0.04
0.08
3.36
Whispeak [ 40 ]
98.9 M
11.94
1.02
1.27
0.39
3.58
3.64
DF_Arena_1B_V_1 [ 41 ]
1,000 M
22.21
2.92
0.91
1.14
4.66
6.37
Momenta [ 40 ]
350 M
25.47
2.87
4.63
1.21
4.88
7.81
Table 5: Comprehensive Zero-shot EER (%) results across 29 systems. All models are evaluated without fine-tuning. The Average is recalculated based on five key benchmarks: In-the-Wild, ASV2019, ASV2021 LA, FakeOrReal, and ADD 2022 Track 1. Values in bold indicate the best results per column. Lower EER indicates superior performance.
Appendix figures & tables8 assets
Supplementary material from the paper’s appendix.
Appendix
Language
Whisper-base EER (%)
Improv. ↓
w/o AAFI+BAP
w/ AAFI+BAP
Spanish (es)
7.11
5.65
1.46
French (fr)
5.17
3.92
1.25
Japanese (ja)
5.52
4.19
1.33
Italian (it)
4.66
3.91
0.75
Korean (ko)
4.50
4.02
0.48
Appendix
Table 6: Cross-lingual evaluation results on unseen languages (SpeechFake MD). EER (%) is compared between the baseline and our proposed method.
ID
Sub-ID
Model
Name
Training Data
Nc
SR
BPS
A
A1
SpeechTokenizer
16k
Librispeech
8
16
4
B
B1
HiFi-Codec
hifi_16k_320d
LibriTTS
4
16
2
B2
hifi_16k_320d_large_uni
VCTK
4
16
2
B3
hifi_24k_320d
AISHELL
4
24
3
C
C1
AudioDec
24k_320d
Valentini
8
24
6.4
D
D1
EnCodec
16k
Common Voice, DAPS
12
16
6
Appendix
Table 7: Details of adopted neural audio codec models. A−F represents different codec frameworks. Nc represents the number of codebooks, SR represents audio sampling rate (in kHz), and BPS represents bits per second.
Subset
Content
Number of Samples
Real
Fake
BD-EN
English-only
64,000
1,128,882
BD-CN
Chinese-only
62,180
824,134
BD-UT
Unseen Techniques
37,854
50,000
BD-TTS
TTS-generated
126,180
1,014,107
BD-VC
VC-generated
126,180
535,205
Appendix
Table 8: Statistics of SpeechFake Bilingual Dataset (BD) Subsets.
Figure 2: Cross-testing results of anti-spoofing models on CodecFake test subsets. We report equal error rate (EER), where lower is better. Rows A-F6 are ADD whisper models trained on one training subset of CodecFake. Columns A-F6 indicate the specific CodecFake test subset evaluated.
Configuration
EER (%)
Utilization Entropy ↑
Baseline (w/o AAFI)
1.19
-
AAFI w/o Noise
1.05
0.74
AAFI w/ Noise
0.91
0.92
Appendix
Table 9: Ablation study of Gaussian noise in AAFI on the BD-UT dataset.
Number of PGD iterations
EER (%)
Estimated Training Time Increase
K=0 (Original in Table 4 )
0.95
-
K=1
1.02
∼ 25%
K=3
0.89
∼ 70%
K=5 (Our Setting)
0.87
∼ 115%
K=10
0.86
∼ 220%
Appendix
Table 10: Ablation study on the effect of PGD iteration number ( K ) in Mel-BAP on the BD-UT dataset.
Attack Method
Budget ( ϵ )
Baseline EER (%)
Ours (Full) EER (%)
No Attack (Clean)
-
1.19
0.53
PGD Attack ( L∞ )
0.002
11.42
3.81
CW Attack ( L2 )
2.0
15.85
5.24
Adaptive PGD ( L∞ )
0.005
22.10
7.93
Appendix
Table 11: Test-time adversarial evaluation on the BD-UT dataset. EER (%) is reported.
Category
16/32-d
64-d
128/256-d
NV (High-fidelity)
8%
33%
59%
TTS (Standard)
64%
22%
14%
Noisy (SNR 10dB)
48%
35%
17%
Appendix
Table 12: Stratified analysis of AAFI routing distributions across varying attack types and severities.
Recent advances in generative audio models have enabled highly realistic synthetic speech, increasing the importance of reliable audio deepfake detection (ADD) systems. While prior studies have primarily focused on adversarially optimized perturbations, the robustness of ADD systems under realistic signal transformations remains insufficiently understood. In this work, we investigate the impact of audio watermarking on ADD systems by treating watermarking as a structured, non-adversarial perturbation rather than a conventional attack mechanism. Using a watermark-based evaluation framework built upon WavMark, we evaluate multiple self-supervised learning (SSL), Convolutional Neural Network (CNN) and Graph Neural Netrowk (GNN)-based ADD models across several benchmark datasets. Beyond conventional detection metrics, we further analyze watermark-induced representation shifts using Fréchet Distance, cosine similarity, and L2 distance in the embedding space. Experimental results reveal a strong dataset-dependent behavior: watermarking causes substantial performance degradation on ASVspoof 2021 LA and DF, while exhibiting limited impact on ASVspoof 2024, FoR, and ITW. Moreover, large embedding-space shifts are strongly associated with severe detection degradation, suggesting that watermark-induced perturbations can substantially alter the feature representations relied upon by current ADD systems. These findings demonstrate that benign signal transformations designed for content protection can expose previously overlooked robustness vulnerabilities in audio deepfake detection systems. Our code is available at https://github.com/ziqian0925/wm-ADD-robustness.git
Zi Qian Yong, Ajinkya Kulkarni, Julia Lau +4
School of Information Technology, Monash University, Malaysia campus · Idiap Research Institute, Switzerland
Achieving robust generalization against unseen attacks remains a challenge in Audio Deepfake Detection (ADD), driven by the rapid evolution of generative models. To address this, we propose a framework centered on hard sample classification. The core idea is that a model capable of distinguishing challenging hard samples is inherently equipped to handle simpler cases effectively. We investigate multiple reconstruction paradigms, identifying the diffusion-based method as optimal for generating hard samples. Furthermore, we leverage multi-layer feature aggregation and introduce a Regularization-Assisted Contrastive Learning (RACL) objective to enhance generalizability. Experiments demonstrate the superior generalization of our approach, with our best model achieving a significant reduction in the average Equal Error Rate (EER) compared to the baseline.
Bo Cheng, Songjun Cao, Xiaoming Zhang +3
Department of Electronic and Electrical Engineering, Southern University of Science and Technology, Shenzhen, China · Tencent Youtu Lab, China
Audio deepfake detection (ADD) must remain effective when new spoofing attacks emerge after deployment. Emerging audio language model (ALM)-based ADD methods are built on predefined supervision from ground-truth labels or verified forensic rationales. However, this paradigm overlooks an ALM's own mistakes, which indicate where targeted supervision is most needed. To this end, we first introduce evolving spoofing environments for ALM-based ADD, where a new attack becomes dominant while previously observed attacks persist. Motivated by the above learning-from-mistakes perspective, we further propose SE-ADD, a self-evolving framework that iteratively adapts an ALM via low-rank adaptation (LoRA) using mistake-driven supervision built from its verdicts and self-generated forensic cues. All training samples receive direct authenticity supervision, while misclassified ones receive additional cue-augmented supervision. As verdicts and cues are regenerated by the updated ALM, the resulting supervision evolves accordingly. Experiments on two ALMs demonstrate the effectiveness of SE-ADD in generalizing to unseen attacks, reducing the equal error rate (EER) from 36.72% to 7.52% for Qwen2-Audio and from 19.93% to 3.97% for MOSS-Audio.
Rong Wan, Wei Xie, Jiaxi Li +4
University of Surrey · Guangxi University · Shenzhen University of Advanced Technology +1