Please use this identifier to cite or link to this item: http://bura.brunel.ac.uk/handle/2438/32360
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dc.contributor.authorZhou, L-
dc.contributor.authorLiu, H-
dc.contributor.authorGan, L-
dc.contributor.authorZhou, Y-
dc.contributor.authorNiedźwiecki, M-
dc.contributor.authorTruong, T-K-
dc.date.accessioned2025-11-18T08:27:37Z-
dc.date.available2025-11-18T08:27:37Z-
dc.date.issued2025-11-07-
dc.identifierORCiD: Hongqing Liu https://orcid.org/0000-0002-2069-0390-
dc.identifierORCiD: Lu Gan https://orcid.org/0000-0003-1056-7660-
dc.identifierORCiD: Yi Zhou https://orcid.org/0000-0001-7445-226X-
dc.identifierORCiD: Maciej Niedźwiecki https://orcid.org/0000-0002-8769-1259-
dc.identifierArticle number: 110390-
dc.identifier.citationZhou, L. et al. (2026) 'A novel sparse adaptive filter for suppressing impulsive disturbance in audio signals', Signal Processing, 241, 110390, pp. 1 - 12. doi: 10.1016/j.sigpro.2025.110390.en_US
dc.identifier.issn0165-1684-
dc.identifier.urihttps://bura.brunel.ac.uk/handle/2438/32360-
dc.descriptionData availability: The codes, datasets and detailed parameters setting record are shared on https://github.com/minikatty/Lq_JSLMP.git.en_US
dc.descriptionSupplementary data are available online at: https://www.sciencedirect.com/science/article/pii/S0165168425005067?via=ihub#appSB .-
dc.description.abstractThis work studies the sparse adaptive filter designs for audio signal recovery in the presence of impulsive disturbance. By exploiting the sparse representation of desired signal and compressibility of impulsive disturbance, a joint sparse least mean p-norm (JSLMP) optimization, in which ℓp-norm (1 ≤ p ≤ 2) measures the data fidelity and ℓq-norm (0 ≤ q ≤ 1) enforces sparse solutions, is developed, termed as ℓq-JSLMP. The filter weights update is derived using gradient descent, and the Adam and variable step size (VSS) are integrated to accelerate convergence and avoid potential local minima. For the special case of q = 1, namely ℓ1-JSLMP, its convergence condition and mean square deviation (MSD) analysis are derived. Finally, an application framework for processing corrupted audio signals is developed. Extensive experiments are conducted on both synthetic and real-measured impulsive noise data, comparing the proposed method with traditional algorithms as well as the deep learning-based GTCRN model. Results demonstrate that the proposed method yields superior perceptual quality and significantly lower memory consumption compared to GTCRN under impulsive disturbance.en_US
dc.description.sponsorshipThis work was jointly supported by the National Natural Science Foundation of China under Grant 61801066 and by Program for Changjiang Scholars and Innovative Research Team in University IRT16R72.en_US
dc.format.extent1 - 12-
dc.format.mediumPrint-Electronic-
dc.languageEnglish-
dc.language.isoen_USen_US
dc.publisherElsevieren_US
dc.rightsCreative Commons Attribution-NonCommercial-NoDerivatives 4.0 International-
dc.rights.urihttps://creativecommons.org/licenses/by-nc-nd/4.0/-
dc.subjectimpulsive disturbanceen_US
dc.subjectleast mean p-normen_US
dc.subjectadaptive filteren_US
dc.subjectsparse reconstructionen_US
dc.subjectadaptive step-sizeen_US
dc.subjectAdam optimizeren_US
dc.subjectspeech enhancementen_US
dc.titleA novel sparse adaptive filter for suppressing impulsive disturbance in audio signalsen_US
dc.typeArticleen_US
dc.date.dateAccepted2025-11-06-
dc.identifier.doihttps://doi.org/10.1016/j.sigpro.2025.110390-
dc.relation.isPartOfSignal Processing-
pubs.issueApril 2026-
pubs.publication-statusPublished-
pubs.volume241-
dc.identifier.eissn1872-7557-
dc.rights.licensehttps://creativecommons.org/licenses/by-nc-nd/4.0/legalcode.en-
dcterms.dateAccepted2025-11-06-
dc.rights.holderElsevier B.V.-
Appears in Collections:Dept of Electronic and Electrical Engineering Research Papers

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