736 research outputs found
A CONSTRAINED MATCHING PURSUIT APPROACH TO AUDIO DECLIPPING
© 2011 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works
Audio Inpainting
(c) 2012 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other users, including reprinting/ republishing this material for advertising or promotional purposes, creating new collective works for resale or redistribution to servers or lists, or reuse of any copyrighted components of this work in other works. Published version: IEEE Transactions on Audio, Speech and Language Processing 20(3): 922-932, Mar 2012. DOI: 10.1090/TASL.2011.2168211
A Proper version of Synthesis-based Sparse Audio Declipper
Methods based on sparse representation have found great use in the recovery
of audio signals degraded by clipping. The state of the art in declipping has
been achieved by the SPADE algorithm by Kiti\'c et. al. (LVA/ICA2015). Our
recent study (LVA/ICA2018) has shown that although the original S-SPADE can be
improved such that it converges significantly faster than the A-SPADE, the
restoration quality is significantly worse. In the present paper, we propose a
new version of S-SPADE. Experiments show that the novel version of S-SPADE
outperforms its old version in terms of restoration quality, and that it is
comparable with the A-SPADE while being even slightly faster than A-SPADE
Sparsity and cosparsity for audio declipping: a flexible non-convex approach
This work investigates the empirical performance of the sparse synthesis
versus sparse analysis regularization for the ill-posed inverse problem of
audio declipping. We develop a versatile non-convex heuristics which can be
readily used with both data models. Based on this algorithm, we report that, in
most cases, the two models perform almost similarly in terms of signal
enhancement. However, the analysis version is shown to be amenable for real
time audio processing, when certain analysis operators are considered. Both
versions outperform state-of-the-art methods in the field, especially for the
severely saturated signals
Revisiting Synthesis Model of Sparse Audio Declipper
The state of the art in audio declipping has currently been achieved by SPADE
(SParse Audio DEclipper) algorithm by Kiti\'c et al. Until now, the
synthesis/sparse variant, S-SPADE, has been considered significantly slower
than its analysis/cosparse counterpart, A-SPADE. It turns out that the opposite
is true: by exploiting a recent projection lemma, individual iterations of both
algorithms can be made equally computationally expensive, while S-SPADE tends
to require considerably fewer iterations to converge. In this paper, the two
algorithms are compared across a range of parameters such as the window length,
window overlap and redundancy of the transform. The experiments show that
although S-SPADE typically converges faster, the average performance in terms
of restoration quality is not superior to A-SPADE
Audio Declipping with Social Sparsity
International audienceWe consider the audio declipping problem by using iterative thresholding algorithms and the principle of social sparsity. This recently introduced approach features thresholding/shrinkage operators which allow to model dependencies between neighboring coefficients in expansions with time-frequency dictionaries. A new unconstrained convex formulation of the audio declipping problem is introduced. The chosen structured thresholding operators are the so called \emph{windowed group-Lasso} and the \emph{persistent empirical Wiener}. The usage of these operators significantly improves the quality of the reconstruction, compared to simple soft-thresholding. The resulting algorithm is fast, simple to implement, and it outperforms the state of the art in terms of signal to noise ratio
Lorentzian Iterative Hard Thresholding: Robust Compressed Sensing with Prior Information
Commonly employed reconstruction algorithms in compressed sensing (CS) use
the norm as the metric for the residual error. However, it is well-known
that least squares (LS) based estimators are highly sensitive to outliers
present in the measurement vector leading to a poor performance when the noise
no longer follows the Gaussian assumption but, instead, is better characterized
by heavier-than-Gaussian tailed distributions. In this paper, we propose a
robust iterative hard Thresholding (IHT) algorithm for reconstructing sparse
signals in the presence of impulsive noise. To address this problem, we use a
Lorentzian cost function instead of the cost function employed by the
traditional IHT algorithm. We also modify the algorithm to incorporate prior
signal information in the recovery process. Specifically, we study the case of
CS with partially known support. The proposed algorithm is a fast method with
computational load comparable to the LS based IHT, whilst having the advantage
of robustness against heavy-tailed impulsive noise. Sufficient conditions for
stability are studied and a reconstruction error bound is derived. We also
derive sufficient conditions for stable sparse signal recovery with partially
known support. Theoretical analysis shows that including prior support
information relaxes the conditions for successful reconstruction. Simulation
results demonstrate that the Lorentzian-based IHT algorithm significantly
outperform commonly employed sparse reconstruction techniques in impulsive
environments, while providing comparable performance in less demanding,
light-tailed environments. Numerical results also demonstrate that the
partially known support inclusion improves the performance of the proposed
algorithm, thereby requiring fewer samples to yield an approximate
reconstruction.Comment: 28 pages, 9 figures, accepted in IEEE Transactions on Signal
Processin
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