Spatial semantic segmentation of sound scenes (S5) involves the detection and extraction of target sound events in audio files. However, in typical detection-extraction pipelines, errors in the detection module propagate to the extraction module, degrading overall performance. In this work, we develop a multichannel detection-extraction model and evaluate inference-time algorithms to improve the class detection and target sound extraction in a multi-stage setup. We evaluate two fitness functions: binary cross entropy and mixture consistency; two relabeling strategies: naive and conditional confusion matrix relabeling; and three source estimate evaluation models: a multi-label classifier, a fine-tuned single-source classifier, and an off-the-shelf audio judge. We thoroughly evaluate the impact of these design choices on the DCASE 2025 Task 4 dataset, and validate entropy-based gating of inference-time updates on the held-out evaluation set, laying the groundwork for inference-time updates of S5 systems in real-world scenarios.
Figures & tables
Fig. 1: Delta CA-SDRi vs. baseline CA-SDRi on the full test set for M2D with conditional CM relabeler and BCE+MC fitness function.
All (1500 ex.)
≤ 10 dB (563 ex.)
Oracle Gap (51 ex.)
Baseline: 12.24 dB
Baseline: 4.79 dB
Baseline: − 0.21 dB
Relabeler
Non-Target?
Fitness
Evaluator
Δ
CA-SDRi
Δoracle
Δ
CA-SDRi
Δoracle
Δ
CA-SDRi
Δoracle
CM
✗
BCE
M2D
-4.10
8.15
1.24
-0.43
4.37
3.08
2.61
2.40
10.04
CM
✓
BCE+MC
M2D
-4.02
8.22
1.42
-0.28
4.52
3.49
2.06
1.85
10.35
CM
✗
BCE
M2D-SC
-2.68
9.56
1.24
0.47
5.26
3.08
5.34
5.13
10.04
CM
✓
BCE+MC
M2D-SC
-2.48
9.77
1.42
0.84
5.64
3.49
5.22
5.01
10.35
Table 1: Performance of various prompt relabeling strategies, fitness functions, and evaluation models on the DCASE 2025 Task 4 test set. CM: Naïve Confusion matrix, CCM: Conditional Confusion matrix, BCE: Binary cross entropy, MC: Mixture consistency, M2D: Multilabel classifier, M2D-SC: Single-class classifier, SAJ: SAM Audio Judge. Δ refers to improvement over the baseline CA-SDRi, while Oracle is the theoretical maximum improvement from the hypotheses evaluated given an ideal fitness function, i.e., computing CA-SDRi against ground truth. Bold indicates best performance; underline indicates second best.
Feature
τ
N
CA-SDRi
Δ↑
Δg↑
BLg↓
BLr↑
SAJ R
0.97
5
12.25
+0.003
+0.90
12.21
12.24
SAJ P
1.77
57
12.28
+0.031
+0.82
9.68
12.35
SAJ O
0.99
17
12.25
+0.004
+0.35
11.62
12.25
SAJ F
1.01
55
12.27
+0.024
+0.67
12.04
12.25
H sum
3.20
24
12.28
+0.031
+1.96
2.85
12.40
H max
0.69
89
12.27
+0.023
+0.39
8.54
12.48
Table 2: Gating threshold sweep results on the test set. SAM Audio Judge scores (R=Recall, P=Precision, O=Overall quality, F=Faithfulness) gate clips with score ≤τ ; M2D entropy features gate clips with score >τ . Thresholds τ are oracle-selected by the sweep. N : number of gated clips. Δ : change in overall CA-SDRi vs. baseline (12.24 dB). Δg : mean CA-SDRi change for gated clips. BLg , BLr : mean baseline CA-SDRi for gated and remaining clips.
Feature
τ
N
CA-SDRi
Δ↑
Δg↑
BLg↓
BLr↑
SAJ R
0.97
4
8.61
0.000
− 0.10
11.59
8.60
SAJ P
1.77
81
8.50
− 0.106
− 2.13
10.52
8.51
SAJ O
0.99
35
8.58
− 0.030
− 1.41
9.56
8.59
SAJ F
1.01
73
8.55
− 0.057
− 1.27
10.69
8.51
H sum
3.20
37
8.60
− 0.005
− 0.24
4.62
8.70
H max
0.69
95
8.54
− 0.070
− 1.20
7.58
8.67
Table 3: Gating on the held-out DCASE 2025 Task 4 evaluation set (1620 clips) with thresholds τ frozen from Table 2 . Δ : change in overall CA-SDRi vs. baseline (8.61 dB); other columns as in Table 2 .
Spatial semantic segmentation of sound scenes (S5) jointly performs audio source separation and tagging on a multichannel audio mixture. Therefore, an S5 system can be decomposed into an audio tagging model and a separation model. The baseline S5 system in task~4 of the DCASE 2025 challenge uses M2D and ResUNet models for tagging and separation, respectively. These are large in overall size and therefore not appropriate for use in embedded devices. The main aim of this paper is to assess the performance loss in an S5 system when using lightweight tagging and separation models instead. To this end, we first compare the tagging performance of lightweight audio tagging models, namely PANNs and DyMNs, in terms of label prediction accuracy against the baseline M2D model. Secondly, we assess separation performance in terms of the scale-invariant signal-to-distortion ratio of the baseline ResUNet separator relative to that of a convolutional-based separator using SuDoRMRF. Finally, we combine the tagging and separation models across various configurations and assess the overall performance of the S5 system using the class-aware signal-to-distortion ratio.
Mayank Mishra, Paul Magron, Romain Serizel
Universit´e de Lorraine, CNRS, Inria, LORIA, F-54000 Nancy, France
This paper presents an overview of the Detection and Classification of Acoustic Scenes and Events (DCASE) 2026 Challenge Task 6, Audio Moment Retrieval (AMR) from Long Audio. Given a several-minute-long audio recording and a free-form text query, AMR aims to retrieve temporal moments in the recording that match the query, where each moment is represented by a pair of start and end timestamps. This task requires effective cross-modal alignment and long-range temporal modeling. We describe the task definition, the evaluation metrics, the development and evaluation datasets, and a baseline system that combines a pre-trained MS-CLAP feature extractor with a Detection Transformer (DETR)-based moment-detection network. On the development data, the baseline trained on a manually annotated dataset and a synthetic dataset achieved [email protected] of 13.56%, indicating that AMR in long audio remains a challenging problem. The challenge attracted 21 teams, which submitted 59 systems in total. The three best systems achieved [email protected] of 48.59%, roughly 3.5 times the baseline score. The results show that strengthening the audio-text feature extractor and the moment-detection network led to substantial performance improvements. Furthermore, the top three teams boosted performance by applying confidence score calibration or ensembling across different temporal resolutions of features.
Target speaker extraction (TSE) aims to recover a target speaker's speech from a mixture using a reference utterance as a cue. Most TSE systems adopt conditional auto-encoder architectures with one-step inference. Inspired by test-time scaling, we propose a training-free multi-step inference method that enables iterative refinement with a frozen pretrained model. At each step, new candidates are generated by interpolating the original mixture and the previous estimate, and the best candidate is selected for further refinement until convergence. Experiments show that, when ground-truth target speech is available, optimizing an intrusive metric (SI-SDRi) yields consistent gains across multiple evaluation metrics. Without ground truth, optimizing non-intrusive metrics (UTMOS or SpkSim) improves the corresponding metric but may hurt others. We therefore introduce joint metric optimization to balance these objectives, enabling controllable extraction preferences for practical deployment.
Zhenghai You, Ying Shi, Lantian Li +1
Beijing University of Posts and Telecommunications, China · Center for Speech and Language Technologies, BNRist, Tsinghua University, China