Fully few-shot class-incremental audio classification (FFCAC) requires recognizing new sound classes from only a handful of labeled examples per session, without forgetting previously learned classes and without any large base dataset. Existing methods typically freeze a pre-trained audio--language encoder and classify with point prototypes, but they suffer from significant performance degradation throughout the sessions due to generic feature representations. We propose SPECTRA, a framework built on a frozen encoder which adds three components. (i) a lightweight trainable adapter that calibrates the generic embeddings to the task; (ii) subspace feature replay, an exemplar-free anti-forgetting scheme that replays old classes by sampling from the low-rank subspace of their stored features; and (iii) a transductive optimal-transport refinement of prototypes at test time. Our central finding is that the subspace structure of the replay diminishes forgetting and outperforms naive Gaussian replay of equal variance. On three FFCAC benchmarks (NSynth-100, FSC-89, LS-100), SPECTRA improves average accuracy and reduces forgetting over current state-of-the-art methods, and our ablations statistically validate each component.
Figures & tables
Figure 1: (a) The FFCAC setting: sessions arrive over time, each introducing 5 new classes with K=5 shots and no base session; the model is evaluated on all classes seen so far (cumulative test set growing 5→25 ). (b) SPECTRA. An audio clip is embedded by a frozen audio–language encoder (PENGI f ), calibrated by a trainable residual-MLP adapter g , and mapped by TAPE’s closed-form orthogonalising transform Pt before a prototypical classifier. Green (training only): each class’s stored support features Mc give a low-rank SVD subspace from which synthetic old-class features x~c are sampled and passed through the same head, a rehearsal that counters forgetting without storing audio. Purple, dashed (inference only): a transductive Sinkhorn step refines the prototypes from the unlabelled query batch.
Figure 2: Replay geometry across sessions. As more classes are added, low-rank subspace replay remains class-aligned, whereas Gaussian replay produces overlapping clouds.
Method
S0
S1
S2
S3
S4
AA ↑
PD ↓
iCaRLCVPR’20 [ 15 ]
71.70
53.51
53.66
49.07
49.48
55.48
40.08
PODNETECCV’20 [ 5 ]
71.87
44.89
43.58
42.93
40.97
48.85
35.57
DERCVPR’21 [ 22 ]
74.40
61.42
60.06
53.73
44.42
58.81
38.95
CECCVPR’21 [ 23 ]
76.13
58.15
53.80
48.26
44.34
56.14
37.49
FACTCVPR’22 [ 24 ]
74.97
51.94
51.43
46.54
43.45
53.27
38.65
PANTMM’23 [ 13 ]
76.71
58.38
53.92
48.44
44.48
56.39
37.72
Table 1: Comparison of different methods on NSynth-100.
Method
S0
S1
S2
S3
S4
AA ↑
PD ↓
PANTMM’23 [ 13 ]
41.48
23.72
18.08
15.27
12.25
22.16
37.72
EDEInterspeech’24 [ 17 ]
53.25
37.65
35.59
32.69
27.60
37.36
30.05
AISPTASLP’25 [ 18 ]
62.66
48.92
43.81
37.73
33.43
45.31
29.23
CLAPICASSP’23 [ 6 ]
49.64
37.73
28.55
25.42
23.70
33.01
25.94
COOPIJCV’22 [ 26 ]
63.32
37.75
27.50
22.07
18.64
33.86
44.69
COCOOPCVPR’22 [ 25 ]
64.65
37.62
26.63
21.22
16.85
33.39
47.80
Table 2: Comparison of different methods on FSC-89.
Method
S0
S1
S2
S3
S4
S5
S6
S7
S8
S9
AA ↑
PD ↓
NSynth-100 ( 10 -way ×10 -session)
TAPE
95.6
92.2
90.8
89.6
88.4
86.5
85.1
84.0
82.9
81.5
87.66
14.11
SPECTRA
97.6
95.6
94.3
93.4
92.7
91.5
90.8
90.1
89.6
88.9
92.45
8.77
LS-100 ( 10 -way ×10 -session)
TAPE
90.1
85.4
80.7
77.9
75.2
73.4
70.7
69.0
67.2
65.7
75.53
24.43
SPECTRA
95.1
91.2
88.1
85.7
83.7
82.3
80.4
79.7
78.6
77.8
84.26
17.31
Table 3: Per-session accuracy (%) under harder protocols: 10 -way ×10 -session on NSynth-100 and LS-100, and 10 -way ×8 -session on FSC-89 (FSC-89 has only 89 classes, capping it at 8 sessions); 30 seeds.
Method
S0
S1
S2
S3
S4
AA ↑
PD ↓
PANTMM’23 [ 13 ]
85.70
52.20
39.17
32.95
29.88
47.98
37.72
EDEInterspeech’24 [ 17 ]
91.90
70.23
54.21
46.97
45.94
61.85
45.96
AISPTASLP’25 [ 18 ]
91.20
70.26
53.71
47.08
45.15
61.48
46.05
CLAPICASSP’23 [ 6 ]
18.96
9.06
5.91
4.75
3.37
8.41
15.59
COOPIJCV’22 [ 26 ]
49.14
27.64
18.48
14.48
11.01
24.15
38.14
COCOOPCVPR’22 [ 25 ]
46.90
24.32
15.75
10.86
7.84
21.14
39.06
Table 4: Comparison of different methods on LS-100.
In the task of few-shot class-incremental audio classification, the number of classes is assumed to always increase without considering the possibility of decrease. However, the number of classes generally increases or decreases in practice. In this paper, we investigate a problem of Few-shot Class-variable Incremental Audio Classification (FCIAC), in which the number of classes increases or decreases. We propose a FCIAC method using prototype adaptation and pseudo class-variable training. The model in our method consists of an encoder and a classifier. The classifier is initialized by a class-variable prototype adaptation network, whose structure dynamically changes with the change of classes. In addition, we design a pseudo class-variable training strategy to enhance the model's adaptability to changing classes. Experiments on three public datasets show that our method exceeds previous methods in average accuracy. The code is at: https://github.com/cgq2971-afk/FCIAC.
Yanxiong Li, Guoqing Chen, Qianqian Li +1
School of Electronic and Information Engineering, South China University of Technology, Guangzhou, China
Most existing audio classification methods suppose that each query (testing) sample belongs to a class of support (training) samples, and misrecognize samples of unseen classes as seen classes (cannot reject samples of unseen classes). In this study, we propose a method for Few-shot Open-set Audio Classification (FOAC), which can recognize query samples of seen classes after updating the model using a few support samples, and meanwhile reject query samples from unseen classes. We design a model consisting of an encoder and a classifier. The encoder is the backbone of a ResNet used for extracting embeddings. The classifier consists of prototype generators of few-shot classes and open-set classes. Prototypes of few-shot classes are obtained by fusing the class-discriminative information of support and query embeddings and by assigning larger weighting coefficient to representative part of the support embeddings. One prototype is generated for open-set classes using the proposed prototype generator. The encoder is trained with abundant samples of base classes in supervised manner, and then the prototypes of base classes are generated under the supervision of a joint loss. The classifier is trained using a few samples of few-shot classes in a meta-training way. Three public datasets (LS-100, NSynth-100, and FSC-89) are used to assess the performance of our method. Experiments show that our method has advantage over prior methods in AUROC and accuracy. This advantage has statistical significance for most prior methods. Our method has lower computational complexity than most prior methods. The code is at https://github.com/Jessytan/FOAC-AIFP.
Yanxiong Li, Jiaxin Tan, Qianqian Li +3
School of Electronic and Information Engineering, South China University of Technology, Guangzhou 510640, China · Faculty of Information Technology and Communication Sciences, Tampere University, FI-33720 Tampere, Finland
Few-shot audio classification requires audio--language models to distinguish variable acoustic events from limited labeled data. PALM's main formulation adapts encoded class representations without changing pretrained encoders, but its class representations remain input-independent and classification uses audio features without class-conditioned refinement. We introduce DF-PALM, a dynamic feature-space prompting framework that combines audio-conditioned class representations with asymmetric cross-attention for audio refinement. Class representations are conditioned after text encoding, held fixed across fusion layers for each input, and reused for final scoring. Both pretrained encoders remain frozen, avoiding audio-conditioned text-encoder passes. With PENGI as the frozen backbone, the main 16-shot evaluation covered 11 benchmarks and six baselines. DF-PALM achieved 80.69% mean accuracy and 80.32% macro-averaged F1, exceeding PALM by 4.11 and 4.30 percentage points, respectively. Paired benchmark-level tests supported both gains after Holm correction. Across four shot settings, DF-PALM achieved the highest mean accuracy and macro-averaged F1 among the four evaluated prompt-learning methods. The gains extended to LAION-CLAP and OpenFLAM, with significant improvements over PALM in both metrics. At 50 candidate classes, operator-counted inference floating-point operations increased by 0.39% relative to PALM. These results support instance-conditioned feature-space adaptation as an effective approach to few-shot audio classification. Our code is publicly available at GitHub https://github.com/sunyulin0421/DF-PALM.
Yulin Sun, Shenghua Li, Yong Dou +3
College of Computer Science and Technology, National University of Defense Technology, Changsha, China. · State Key Laboratory of Complex & Critical Software Environment