Audio Signal Enhancement with Low-Rank Time-Frequency Synthesis Models

The low-rank time-frequency synthesis (LRTFS) model assumes that the synthesis coefficients of the unobserved signal follow a latent low-rank variance structure. This work explores the potential of the LRTFS model in simultaneous noise reduction and source separation tasks. To this end, we devise a robust expectation-maximisation algorithm to approximate the type-II estimator of the latent low-rank components. To circumvent analytic marginalisation of the synthesis coefficients (E-step), integration is performed numerically through a lightweight MCMC sampler, thus bypassing explicit matrix inversion. The proposed sampler exploits the structure of the chosen dictionary—the complex modified discrete cosine transform (ℂMDCT)—to navigate the parameter space in a series of inexpensive joint draws. Moreover, the non-negative factorisation sub-problem (M-step) is supplemented with domain-specific regularisation schemes aimed at fostering high-quality audio reconstructions. The proposed technique is tested on a collection of denoising examples, and its potential for signal decomposition is explored in a series of case studies.

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Publication Details

Published
2025-06-08
DOI
https://doi.org/10.1109/ssp64130.2025.11073473
Primary Topic
Speech and Audio Processing
Type
article
Field-Weighted Citation Impact
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article

Audio Signal Enhancement with Low-Rank Time-Frequency Synthesis Models

Rubén M. Clavería, Simon Godsill
Speech and Audio Processing
article

Audio Signal Enhancement with Low-Rank Time-Frequency Synthesis Models

Rubén M. Clavería, Simon Godsill
article en

Abstract

The low-rank time-frequency synthesis (LRTFS) model assumes that the synthesis coefficients of the unobserved signal follow a latent low-rank variance structure. This work explores the potential of the LRTFS model in simultaneous noise reduction and source separation tasks. To this end, we devise a robust expectation-maximisation algorithm to approximate the type-II estimator of the latent low-rank components. To circumvent analytic marginalisation of the synthesis coefficients (E-step), integration is performed numerically through a lightweight MCMC sampler, thus bypassing explicit matrix inversion. The proposed sampler exploits the structure of the chosen dictionary—the complex modified discrete cosine transform (ℂMDCT)—to navigate the parameter space in a series of inexpensive joint draws. Moreover, the non-negative factorisation sub-problem (M-step) is supplemented with domain-specific regularisation schemes aimed at fostering high-quality audio reconstructions. The proposed technique is tested on a collection of denoising examples, and its potential for signal decomposition is explored in a series of case studies.

University of Cambridge (GB)
Affordable and clean energy
Openalex Percentile: Top 76%
Speech and Audio Processing
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