KanAdapter: A Kolmogorov-Arnold Network-based Plug-and-Play Module for Efficient Fine-tuning of Foundation Speech Models
Authors: Phuong Tuan Dat, Phuong Khai Minh, Tran Huy Dat
Organizations: Dept. of Electrical and Computer Engineering, National University of Singapore · School of Computing, National University of Singapore · Institute of Advanced Intelligence and Computing (IAIC), A∗STAR
Fully fine-tuning self-supervised learning (SSL) speech models for downstream tasks is computationally prohibitive, and existing parameter-efficient fine-tuning approaches predominantly rely on MLP-based adapters whose fixed activation functions limit their representational expressiveness under tight parameter budgets. We propose KanAdapter, a lightweight adapter framework that replaces conventional MLP bottlenecks with Group-Rational Kolmogorov-Arnold Network (GR-KAN) modules for more expressive and parameter-efficient adaptation. Following a parallel bottleneck design, KanAdapter inserts trainable GR-KAN branches alongside frozen Transformer encoder blocks and leverages weight transfer from pre-trained MLP layers for stable initialization. Across speaker verification, speech emotion recognition, and deepfake detection, KanAdapter achieves up to 97.5% reduction in trainable parameters relative to full fine-tuning while remaining highly competitive, and consistently outperforms AdaptFormer under comparable parameter budgets. In continual learning, it yields up to 83.6% error reduction over full fine-tuning and MLP-based adapters, which we attribute to the localized nature of GR-KAN's rational activations that mitigates catastrophic forgetting. To our knowledge, this is the first work to explore KAN-based modules for parameter-efficient fine-tuning of speech foundation models.
Figures & tables
Figure 1: The proposed architecture: a) Standard Transformer layer b) AdaptKANFormer architecture c) KanAdapter.
Method
#Trainable
EER (%)
Params
Vox1-O
Vox1-E
Vox1-H
Full Fine-tuning
364M
0.49
0.44
1.58
LoRA
4M
2.25
2.19
3.27
AdaptFormer
8M
1.44
1.35
2.70
KanAdapter
9M
0.52
0.50
1.93
Table 1: Speaker verification results on VoxCeleb1. Bold denotes the best result, underline denotes the second-best result.
Method
#Trainable Params
Development set
Test set
F1-Macro
F1-Micro
F1-Macro
F1-Micro
Full Fine-tuning
331M
0.5279
0.3433
0.3321
0.3201
LoRA
8M
0.3126
0.2011
0.1822
0.1794
AdaptFormer
14M
0.4123
0.2742
0.2310
0.2040
KanAdapter
16M
0.5127
0.3436
0.3290
0.3180
Table 2: Speech emotion recognition results on MSP-Podcast. Bold denotes the best result, underline denotes the second-best result.
Model
#Trainable
EER (%)
params
LA19
LA21
DF21
LA5
ITW
Full Fine-tuning
319M
0.27
1.21
4.26
15.38
8.34
LoRA
9M
2.77
5.72
6.09
24.65
10.94
AdaptFormer
15M
0.53
4.93
6.51
19.55
10.21
KanAdapter
17M
0.34
1.32
5.93
18.40
7.96
Table 3: Performance of proposed method compared with baseline model on deepfake detection task. Bold denotes the best result, underline denotes the second-best result.
Model
EER (%)
LA19
LA21
DF21
LA5
Full Fine-tuning
5.25
4.74
6.92
8.60
LoRA
7.19
9.64
9.28
20.11
AdaptFormer
4.94
4.24
3.38
11.11
KanAdapter
0.86
3.59
2.76
6.75
Table 4: Continual Learning Performance: Fine-tuning Results on ASVspoof5 Following ASVspoof19 Pre-training. Bold denotes the best result, underline denotes the second-best result.
Figure 2: Performance with bottleneck dimension d^ .
SSL Model
Mode
#Trainable Params
LA21
DF21
WavLM Large 6 6 6 https://huggingface.co/microsoft/wavlm-large
Fine-tuning Transformer-based foundation models has become the dominant strategy for domain adaptation in audio and speech processing. To reduce the computational and memory costs of this process, parameter-efficient transfer learning (PETL) methods have been widely explored. Meanwhile, Mamba, a recent state-space model, has emerged as a promising alternative to Transformers for sequence modeling. In this work, we present MambAdapter, a parameter-efficient transfer learning approach that integrates Mamba into low-rank bottleneck adapters. Our design combines parameter sharing across adapters with the injection of a lightweight Mamba module, enabling more effective modeling of audio features. We demonstrate that MambAdapter matches or outperforms strong PETL baselines on four audio classification tasks and five speech recognition languages, even when operating under reduced parameter budgets.
Frozen speech foundation models (SFMs) make downstream adaptation efficient: the backbone can stay fixed while a task learns layer fusion and a lightweight classifier. Full adversarial fine-tuning is a standard route to robustness, but generating adversarial examples and updating the backbone for every task sacrifices that efficiency. We ask whether robustness can instead be learned before future tasks are known. For a frozen backbone and linear classifier, robustness can be understood through the interaction between representation stability and decision-boundary margin. This leads directly to our design: we stabilize representations across the hidden layers, rather than only the final layer, while preserving clean representations; after clean adaptation selects the layer mixture, we keep it fixed and enlarge only the classifier margin, without downstream adversarial examples. We evaluate Wav2Vec2, HuBERT, and WavLM Large on four tasks under adaptive 30 dB attacks. Across 12 backbone-task pairs, hierarchical robustification improves robust accuracy by 46.4 pp, while margin refinement adds 4.0 pp for 1.1 pp of clean accuracy. Code and configurations are available at https://github.com/arefmousavi/hierarchical-robust-sfm.
Adapting multilingual speech foundation models to low-resource languages remains difficult, especially for languages that are poorly represented during pre-training. While parameter-efficient fine-tuning (PEFT) reduces the cost of adapting large models, conventional approaches such as LoRA rely on generic low-rank parameterizations and do not explicitly use downstream task information to define the adaptation subspace. To investigate whether task-informed PEFT can better support low-resource ASR, we apply Fisher-Whitened Cross-Covariance Analysis (FCCA) to Whisper and Qwen3-ASR, and introduce two complementary extensions: Asymmetric-Coupled FCCA (AC-FCCA), which exploits structured cross-layer sharing, and Adaptive-Rank FCCA (AR-FCCA), which reallocates adaptation capacity across projection matrices under a fixed parameter budget. Under controlled multilingual experiments, we evaluate these approaches on languages that are poorly represented or unsupported during pre-training alongside well-represented languages. Standard FCCA is competitive with, and usually outperforms, trainable-parameter-budget-matched LoRA. AR-FCCA provides the most consistent improvement over standard FCCA across both model architectures, with statistically significant gains in several evaluation settings, while retaining the same number of trainable parameters. These results show that task-informed subspace construction can be effective for low-resource speech adaptation, and that adaptive rank allocation provides a robust way to improve parameter efficiency without increasing model capacity.
Asmee Mishra, Mengjie Qian, Brechtje Post +1
Department of Engineering, University of Cambridge, United Kingdom · Phonetics Laboratory, University of Cambridge, United Kingdom