Codage audio perceptuel à bas débit par Décomposition Modale Empirique (EMD)

National audienceCet article présente une nouvelle technique de compression audio à bas débit, basée sur la décomposition EMD (Empirical Mode Decomposition), en association avec un modèle d'audition (seuil de masquage). Par un processus de tamisage, le signal audio est décomposé en une somme finie de composantes de type AM-FM, appelées IMF (Intrinsic Mode Function) parfaitement décrites par leurs extrema. En adoptant un seuillage approprié, contrôlé par le modèle psycho-acoustique, seuls les extrema pertinents d'une IMF sont codés. Le nombre de bits alloué au codage des extrema seuillés varie d'une IMF à une autre et respecte la contrainte d'inaudibilité de l'erreur de quantification. Les techniques de seuillage des extrema et d'allocation des bits sur lesquelles repose le procédé de compression proposé, garantissent un bas débit et une bonne qualité d'écoute du signal codé-décodé. Les résultats obtenus sur différents signaux audio, mettent en évidence l'intérêt de l'approche proposée. Comparé à une compression par ondelette et au codeur MP3, le codeur proposé présente un gain de performances significatif en termes de taux de compression et de qualité d'écoute

Tatouage pour le renforcement de la qualité audio des systèmes de communication bas débit

L'objectif de cette thèse est d'étudier l'idée du tatouage dans le traitement du son.Les recherches en tatouage audio se sont principalement tournées vers des applications sécuritaires ou de transmission de données auxiliaires. Une des applications visées par ce concept consiste à améliorer la qualité du signal hôte ayant subi des transformations et ceci en exploitant l'information qu'il véhicule. Le tatouage audio est donc considéré comme mémoire porteuse d'informations sur le signal originel. La compression à bas débit des signaux audio est une des applications visée par ce concept. Dans ce cadre, deux objectifs sont proposés : la réduction du pré-écho et de l'amollissement d'attaque, deux phénomènes introduits par les codeurs audio perceptifs, en particulier les codeurs AAC et MP3; la préservation de l'harmonicité des signaux audio dégradée par les codeurs perceptifs à extension de bande, en particulier le codeur HE-AAC.La première partie de ce manuscrit présente les principes de base des systèmes de codage bas débit et étudie les différentes distorsions introduites par ces derniers. Fondées sur cette étude, deux solutions sont proposées. La première, visant principalement la réduction du pré-écho, consiste à corriger l'enveloppe temporelle du signal après réception en exploitant la connaissance a priori de l'enveloppe temporelle du signal original, supposée transmise par un canal auxiliaire à faible débit (< 500 bits/s). La seconde solution vise à corriger les ruptures d'harmonicité générées par les codeurs à extension de bande. Ce phénomène touche essentiellement les signaux fortement harmoniques (exemple : violon) et est perçu comme une dissonance. Une préservation de l'harmonicité des signaux audio par des opérations de translation spectrale est alors proposée, les paramètres étant là encore transmis par un canal auxiliaire à faible débit.La seconde partie de ce document est consacrée à l'intégration du tatouage audio dans les techniques de renforcement de la qualité des signaux audio précitées. Dans ce contexte, le tatouage audio remplace le canal auxiliaire précédent et œuvre comme une mémoire du signal originel, porteuse d'informations nécessaires pour la correction d'harmonicité et la réduction de pré-écho. Cette seconde partie a été précédée par une étape approfondie de l'évaluation des performances de la technique de tatouage adoptée en terme de robustesse à la compression MPEG (MP3, AAC et aacPlus).The goal of this thesis is to explore the idea of watermark for sound enhancement. Classically, watermark schemes are oriented towards security applications or maximization of the transmitted bit rates. Our approach is completely different. Our goal is to study how an audio watermarking can improve the quality of the host audio signal by exploiting the information it conveys. The audio watermarking is considered as a memory that carries information about the original signal.The low bitrate compression of audio signals is one of the applications covered by this concept. In this context, two objectives are proposed: reducing the pre-echo and the attack softening, two phenomena introduced by the perceptual audio coders, particularly AAC and MP3 encoders ; preserving the harmonicity of audio signals, distorted by coders with bandwidth extension, especially HE-AAC encoder. These coders are limited in the reconstruction of the high-frequency spectrum mainly because of the potential unpredictability of the fine structure of the latter, as well as imperfect indicators of tonal to noise.The first part of this manuscript presents the basic principles of low rate coding systems and studies the various distortions introduced by the latter. Based on this study, two solutions are proposed. The first one, principally aimed at reducing the pre-echo, consist in correcting the time envelope of the signal after reception by exploiting the prior knowledge of the temporal envelope of the original signal, which is assumed transmitted by an auxiliary channel at low bitrates (<500 bps). The second solution is to correct the harmonicity generated by coders with bandwidth extension. This primarily affects strongly harmonic signals (e.g. violin) and is perceived as a dissonance. We propose then to preserve the harmonicity of audio signals by spectral translations. The parameters being passed again by an auxiliary channel at low bitrates.The second part of this document is dedicated to the integration of audio watermarking techniques in the solution presented in the first part. In this context, the audio watermarking replaces the previous auxiliary channel and is regarded as a memory of the original signal, carrying information necessary for the correction of harmonicity and the pre-echo reduction.PARIS5-Bibliotheque electronique (751069902) / SudocSudocFranceF

Audio encoding using Huang and Hilbert transforms

International audienceIn this paper an audio coding scheme based on the Empirical Mode Decomposition (EMD) in association with the Hilbert transform is presented. The audio signal is decomposed adaptively into intrinsic oscillatory components called Intrinsic Mode Functions (IMFs) by EMD, and the associated instantaneous amplitudes and the instantaneous phases are calculated. The basic principle of the proposed approach consists in encoding the instantaneous amplitudes by linear prediction and the instantaneous phases by scalar quantization. The decoder recovers the original signal from IMFs reconstruction by demodulation and summation. The compression method is applied to different audio signals, and results are compared to MP3 a variable bit rate coder and to wavelet approaches

Audio encoding based on the empirical mode decomposition

National audienceThis paper deals with a new approach for perceptual audio encoding, based on the Empirical Mode Decomposition (EMD). The audio signal is decomposed adaptively into intrinsic oscillatory components by EMD called Intrinsic Mode Functions (IMFs), which can be fully described by their extrema. These extrema are encoded after an appropriate thresholding scheme controlled by a psycho-acoustic model. The decoder recovers the original signal after IMFs reconstruction by means of spline interpolation and their summation. The proposed approach is applied to different audio signals and results are compared to wavelets and to MPEG1-layer3 (MP3)approaches. Relying on exhaustive simulations, the obtained results show that the proposed compression scheme performs better than the MP3 and the wavelet approach in terms of bit rate and audio quality

Effect of angiotensin-converting enzyme inhibitor and angiotensin receptor blocker initiation on organ support-free days in patients hospitalized with COVID-19

IMPORTANCE Overactivation of the renin-angiotensin system (RAS) may contribute to poor clinical outcomes in patients with COVID-19. Objective To determine whether angiotensin-converting enzyme (ACE) inhibitor or angiotensin receptor blocker (ARB) initiation improves outcomes in patients hospitalized for COVID-19. DESIGN, SETTING, AND PARTICIPANTS In an ongoing, adaptive platform randomized clinical trial, 721 critically ill and 58 non–critically ill hospitalized adults were randomized to receive an RAS inhibitor or control between March 16, 2021, and February 25, 2022, at 69 sites in 7 countries (final follow-up on June 1, 2022). INTERVENTIONS Patients were randomized to receive open-label initiation of an ACE inhibitor (n = 257), ARB (n = 248), ARB in combination with DMX-200 (a chemokine receptor-2 inhibitor; n = 10), or no RAS inhibitor (control; n = 264) for up to 10 days. MAIN OUTCOMES AND MEASURES The primary outcome was organ support–free days, a composite of hospital survival and days alive without cardiovascular or respiratory organ support through 21 days. The primary analysis was a bayesian cumulative logistic model. Odds ratios (ORs) greater than 1 represent improved outcomes. RESULTS On February 25, 2022, enrollment was discontinued due to safety concerns. Among 679 critically ill patients with available primary outcome data, the median age was 56 years and 239 participants (35.2%) were women. Median (IQR) organ support–free days among critically ill patients was 10 (–1 to 16) in the ACE inhibitor group (n = 231), 8 (–1 to 17) in the ARB group (n = 217), and 12 (0 to 17) in the control group (n = 231) (median adjusted odds ratios of 0.77 [95% bayesian credible interval, 0.58-1.06] for improvement for ACE inhibitor and 0.76 [95% credible interval, 0.56-1.05] for ARB compared with control). The posterior probabilities that ACE inhibitors and ARBs worsened organ support–free days compared with control were 94.9% and 95.4%, respectively. Hospital survival occurred in 166 of 231 critically ill participants (71.9%) in the ACE inhibitor group, 152 of 217 (70.0%) in the ARB group, and 182 of 231 (78.8%) in the control group (posterior probabilities that ACE inhibitor and ARB worsened hospital survival compared with control were 95.3% and 98.1%, respectively). CONCLUSIONS AND RELEVANCE In this trial, among critically ill adults with COVID-19, initiation of an ACE inhibitor or ARB did not improve, and likely worsened, clinical outcomes. TRIAL REGISTRATION ClinicalTrials.gov Identifier: NCT0273570

A finite memory non stationary LMS algorithm for tracking radio-mobile channels with abrupt jumps

Dans ce papier nous montrons que l'algorithme Finite Memory Non Stationary LMS (FM-NSLMS) est efficace pour la poursuite des canaux de transmission radio-mobile dont les variations temporelles sont à la fois caractérisées par un modèle markovien d'ordre P et affectées de ruptures brusques. La non stationnarité du canal de transmission est représentée par un modèle markovien d'ordre P. L'algorithme FM-NSLMS est conçu pour la poursuite des canaux à variations temporelles markoviennes. Par ailleurs, la capacité de mémorisation introduite dans l'algorithme lui permet de tenir compte de la nature récursive de la non stationnarité. Ainsi, il est capable de poursuivre avec une grande vitesse de convergence les sauts des paramètres markovien de certaines valeurs à d'autres

A NEW APPROACH FOR SPARSE PHASE RETRIEVAL BASED ON SUPPORT LIFTING AND LEAST SQUARE ESTIMATION

To recover a signal x from the magnitude of a possible linear transform of it, problem known as Phase Retrieval (PR), signal sparsity property has been used to guide the uniqueness of the solution. This paper presents herein a new method for sparse phase retrieval (SPR). Based on a lifting operation, we reduce the problem of SPR to solving a linear system with regards to a vectorized version of xx T. We then use the struc-tured sparsity property of this vectorized form to interpret this operation rather as a lifting operation of the signal support. The signal support is identified iteratively using the gradient pursuit principle in conjunction with subsequent refinements aiming to control the stability of the updated solution. A simple least square estimation on the lifted support is then brought out, iteratively and if required, to determine the lifted solution; from which a rank−1 decomposition is achieved to recover the signal of interest. Simulation results confirm the efficiency of the so-called Greedy Support-Lifting Based algorithm (GSuLA) with acceptable complexity. Robustness of the algorithm is also assured for noisy measurements

High-frequency tonal components restoration in low-bitrate audio coding using multiple spectral translations

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