The noise requirements for naval and research vessels have seen an increasing demand for quieter ships in order to fulfil current regulations and to reduce the effects on marine life. Hence, new methods dedicated to the characterization of propeller noise, which is the main source of noise in the far-field, are needed. The study of cavitating propellers in closed-section is interesting for analyzing hydrodynamic performance but could involve significant difficulties for hydroacoustic study, especially due to reverberation and boundary layer noise in the tunnel. The aim of this paper is to present a numerical methodology for the identification of hydroacoustic sources on marine propellers using hydrophone arrays in a large hydrodynamic tunnel. The main difficulties are linked to the reverberation of the tunnel and the boundary layer noise that strongly reduce the signal-to-noise ratio. In this paper it is proposed to estimate the reflection coefficients using an inverse method and some reference transfer functions measured in the tunnel. This approach allows to reduce the uncertainties of the propagation model used in the inverse problem. In order to reduce the boundary layer noise, a cleaning algorithm taking advantage of the low rank and sparse structure of the cross-spectrum matrices of the acoustic and the boundary layer noise is presented. This approach allows to recover the acoustic signal even well under the boundary layer noise. The improvement brought by this method is visible on acoustic maps resulting from beamforming and DAMAS algorithms. Keywords—Acoustic imaging, boundary layer noise denoising, inverse problems, model adaptation.

Thanks to its easy implementation and robust performance, beamforming is applied for source localisation in several fields. Its effectiveness depends greatly on the array sensor configuration. This paper introduces a criterion to improve the array beampattern and increase the accuracy of sound source localisation. The beamwidth and the maximum sidelobe level are used to quantify the spatial variation of the beampattern through a new criterion. This criterion is shown to be useful, especially for the localisation of moving sources. A genetic algorithm is proposed for the optimisation of microphone placement. Statistical analysis of the optimised arrays provides original results on the algorithm performance and on the optimal microphone placement. An optimised array is tested to localise the sound sources of a high speed train. The results show an accurate separation.

Les exigences en matiere de bruit rayonne par les navires de la Marine ou de recherche engendrent le developpement de nouvelles methodes pour ameliorer leurs caracterisations. Le propulseur, qui est la source la plus importante en champ lointain, est generalement etudie en tunnel hydrodynamique. Cependant, compte tenu de la reverberation dans le tunnel et du niveau eleve du bruit de couche limite turbulente (CLT), la caracterisation peut s’ averer delicate. L'objectif de la these est d'ameliorer les capacites de mesures acoustiques du Grand Tunnel Hydrodynamique (GTH) de la DGA en matiere de bruits emis par les maquettes testees dans des configurations d'ecoulement.Un modele de propagation base sur la theorie des sources images est utilise afin de prendre en compte le confinement du tunnel. Les coefficients de reflexion associes aux parois du tunnel sont identifies par methode inverse et a partir de la connaissance de quelques fonctions de transfert. Un algorithme de debruitage qui repose sur l’ Analyse en Composantes Principales Robuste est egalement propose. Il s'agit de separer, de maniere aveugle ou semi-aveugle, l’ information acoustique du bruit de CLT en exploitant, respectivement, la propriete de rang faible et la structure parcimonieuse des matrices interspectrales du signal acoustique et du bruit. Ensuite, une technique d'imagerie basee sur la methode des sources equivalentes est appliquee afin de localiser et quantifier des sources acoustiques correlees ou decorrelees. Enfin, la potentialite des techniques proposees est evaluee experimentalement dans le GTH en presence d'une source acoustique et d'un ecoulement controle.

Reducing noise levels around airports remains a key issue due to traffic increase, even if aircraft are more and more silent. It however seems that a new gap cannot be expected keeping the present technology and several novel airplane concepts are considered for the next decades. It is expected that shielding turbofan noise with fuselage, wings, or empennage could be beneficial for sound pressure radiated towards the ground. Present work focuses on comparisons between engine under or over the wing, based on parametric studies avoiding long CFD-CAA calculations. Three fast complementary methods have been implemented to evaluate the effect of engine installation on sound pressure levels on the ground: an analytical model of reflection and diffraction, a boundary element model (BEM), and tests on a 1/10-scale wing model in a large anechoic chamber. Each method applies to a preferred frequency range (high, low, and medium, respectively), but some appraisals can be cross-checked. Results simulating a takeoff flight show that a judicious location of the engines can be fruitful in terms of noise management. An engine over the wing can provide a gain of 4 to 10 dB in the shadow zone on several kilometers around the aircraft route where a conventional design is the noisiest. Present study should be extended in the future to take into account the non-compactness of the acoustic sources and the refraction due to the flow. Other constraints in an aircraft design should also be examined (aerodynamic performance, structure resistance, fuel consumption...).

Rolling noise is the major contribution to train pass-by noise. Array processing provides noise maps of the source location on the train. The rolling noise contribution is located around the wheels at 2 kHz on those maps and the rail is not well identified as a source, whereas models, like TWINS, estimate that its acoustical contribution can be in the same order of magnitude as the wheel contribution in the medium frequency range (1000 Hz). Flexion wave causes the rail to radiate in privileged direction due to fluid structure interaction. Beamforming processing identifies this acoustical radiation only under particular conditions, which is an explanation to the underestimation of the rail radiation on the global noise map of the train. This article presents a simple radiation model of the rail and the simulation result provided from beamforming processing to localize the theoretical radiation angle. An experimentation is performed using the classical 2 D adapted for train pass-by source localisation. The rail is excited in the vertical direction using a shaker. The radiation angle is not retrieved: the use of the classical array is not relevant to measure the radiation angle.

A number of sound field separation techniques have been proposed for different purposes. However, these techniques just consider the separation of sound fields in the space domain and are restricted to stationary sound fields. When the sound fields are nonstationary, it is also necessary to perform the separation in the time domain. Therefore, on the basis of the propagation principle of sound pressure in the time-wavenumber domain, a nonstationary sound field separation technique with two closely spaced parallel measurement surfaces is proposed. It can separate the nonstationary signals generated by the primary sources in both time and space domains when the disturbing sources exist on the other side of the measurement plane. The signals in time and space domains are separated by using the spatial Fourier transform method and the time domain deconvolution method. A simulation involving two monopoles driven by nonstationary signals demonstrates that the method proposed can remove the influence of disturbing sources in both time and space domains. The feasibility of this method is also demonstrated by an experiment with two loudspeakers located on two sides of measurement planes. Additionally, to comment more objectively on the separation results, some indicators are computed in both the simulation and experiment. (C) 2012 Acoustical Society of America. [DOI: 10.1121/1.3683249]

Time domain holography and real-time nearfild acoustic holography are two available Methods to reconstruct nonstaitionary sound fields. However, both two methods require all Sources to be located on one side of the measurement surface, while the other side is in the Free field. In many real situations, some disturbing sources exist on the other side. To Elimminate the influence of disturbing sources, a nonstationay sound field separation Technique in the time-wavenumber domain is proposed. It can separate out the Nonstationary signals generated by the objective sources from the mixed ones in both time And space domain holographhy and real-time nearfield acoustic holography respectively. A Simulation demonstrates the validity of the proposed separation technique to remove the Influence of disturbing sources in both time an space domains, and the feasibility to Reconstruct nonstationary sound fields using the separated data based on time domain Holography and real-time nearfild acoustic holography.

This study aims to automate the simulation of activity diagram (AD) in accordance with the OMG SysML specifications. We use the concept of model-driven engineering to transform AD into VHDL-AMS. This transformation is depicted in two transformations: Activity Diagram (AD) to Petri net (PN) and PN to VHDL-AMS. The first step, although has little importance for simulation, is used for formal verification, domain regularly used in addition with simulation. All simulations were implemented with SystemVision. The second step, allows to execute and simulate a system behaviour modelled by an AD, which is conformed to AD meta-model.

The concept of system level, introduced in Electronic System Level(ESL), offers a system-model representation before thinking about the partition between software and hardware. One ESL initiative is model-based systems engineering (MBSE), whose objective is to reduce ambiguity of specification interpretation, to verify specification in early design steps and to generate code automatically. In this work, we propose a new design methodology that follows the principles of MBSE. Our methodology obeys the EIA-632 standard that defines the systems engineering best practices. It uses SysML as the system description language and HiLeS Designer as a verification tool. We illustrate our methodology in the design of an intelligent remote keyless entry system and apply it to specify and develop new products for a Colombian company that specializes in the commercialization and distribution of electric energy.

A time-domain plane wave superposition method is proposed to reconstruct nonstationary sound fields. In this method, the sound field is expressed as a superposition of time convolutions between the estimated time-wavenumber spectrum of the sound pressure on a virtual source plane and the time-domain propagation kernel at each wavenumber. By discretizing the time convolutions directly, the reconstruction can be carried out iteratively in the time domain, thus providing the advantage of continuously reconstructing time-dependent pressure signals. In the reconstruction process, the Tikhonov regularization is introduced at each time step to obtain a relevant estimate of the time-wavenumber spectrum on the virtual source plane. Because the double infinite integral of the two-dimensional spatial Fourier transform is discretized directly in the wavenumber domain in the proposed method, it does not need to perform the two-dimensional spatial fast Fourier transform that is generally used in time domain holography and real-time near-field acoustic holography, and therefore it avoids some errors associated with the two-dimensional spatial fast Fourier transform in theory and makes possible to use an irregular microphone array. The feasibility of the proposed method is demonstrated by numerical simulations and an experiment with two speakers.

A new method to measure the total energy density of waves traveling in opposite directions in ducts is suggested in order to completely eliminate phase errors that lead to bias errors and are difficult to control in industrial tests. Only the auto-power spectral densities are measured by the three microphones. The inversion of a linear system based on a propagation model, where the two opposite waves are partially coherent, makes it possible to obtain the energy density. The sensitivity of this method to errors in the speed of sound, errors of microphone calibration and errors of microphone positions in the duct is analyzed. To complete the study on the robustness of the method, an evaluation of the statistical errors is carried out. The total uncertainty is used to make recommendations on the choice of the experimental parameters. The selection of the frequency limits permits to maintain the measurement uncertainty within a given confidence interval.

This study aims to automate the transformation of activity diagrams (AD) to Petri nets (PN). Based on specifications given by the Object Management Group (OMG), we have established transformation rules in ATLAS Transformation Language (ATL) to obtain a model consistent with our Petri Net meta-model (model2model). The semantic of Activity Diagram was verified with PetriNet2Tina transformation (model2text) and has allowed us to verify that was the same in the corresponding PN. This verification is done with the "model-checker" TIme petri Net Analyzer (TINA) and Linear Temporal Logic (LTL) language. The user needs only to set up the Activity Diagram from the stakeholder requirements; the transformation and verification is automatic. Petri Net formalism could enable us to provide valuable information on a Activity Diagram, to execute and simulate it.

Les encoffrements et capotages sont sans doute les moyens les plus couramment utilisés pour réduire les nuisances dues au bruit dans l'industrie. Même si la réduction du bruit à la source reste la priorité, on les trouve autour des machines, des groupes turboalternateurs, encapsulant des moteurs ou

Near-field acoustic holography is a measuring process for locating and characterizing stationary sound sources from measurements made by a microphone array in the near-field of the acoustic source plane. A technique called real-time near-field acoustic holography (RT-NAH) has been introduced to extend this method in the case of nonstationary sources. This technique is based on a formulation which describes the propagation of time-dependent sound pressure signals on a forward plane using a convolution product with an impulse response in the time-wavenumber domain. Thus the backward propagation of the pressure field is obtained by deconvolution. Taking the evanescent waves into account in RT-NAH improves the spatial resolution of the solution but makes the deconvolution problem "ill-posed" and often yields inappropriate solutions. The purpose of this paper is to focus on solving this deconvolution problem. Two deconvolution methods are compared: one uses a singular value decomposition and a standard Tikhonov regularization and the other one is based on optimum Wiener filtering. A simulation involving monopoles driven by nonstationary signals demonstrates, by means of objective indicators, the accuracy of the time-dependent reconstructed sound field. The results highlight the advantage of using regularization and particularly in the presence of measurement noise.

Verification and Validation (V&V) on embedded systems design is a crucial topic today. It is essential in systems design to create methods to measure the modeled behavior correctness in order to give more reliability to the design itself. Using formalisms such as Finite State Machine or Petri Nets, it is possible to verify formally or by simulation the design behavior. In this work, we present HiLeS Designer CAD tool to model and to verify systems formally and by simulation. This tool uses its own formalism that allows modeling heterogeneous systems in hierarchical levels, representing the logic behavior by Petri nets and the continuous behavior using VHDL-AMS. The Petri net part can be formally analyzed. The composed model can be transformed to a unique executable virtual prototype in VHDL-AMS. A remote keyless system is presented as an embedded system example.

La caracterisation des sources sonores par traitement d'antennes, en niveau, en spectre et en localisation sur le materiel ferroviaire en mouvement demande de plus en plus de precision et de justesse. En antennerie, la position des capteurs definit les proprietes de l'antenne et ainsi les performances de la mesure. L'objectif du travail est de disposer d'un outil permettant de concevoir une antenne en fonction des besoins et du contexte de la mesure. Dans un premier temps, des criteres tels que la largeur du lobe principal, l'attenuation des lobes secondaires et la constance de la largeur du lobe principal en fonction de l'angle de focalisation sont identifies comme agissant sur les performances de l'antenne. L'objectif est alors de determiner les geometries d'antenne permettant de satisfaire au mieux les criteres proposes. L'optimisation est realisee par un algorithme genetique. Une population d'antennes est generee en placant aleatoirement des microphones a differents emplacements predefinis suivant un maillage fin de forme carree, garantissant une mise en œuvre assez simple. L'optimalite de chaque antenne est evaluee selon un ou plusieurs criteres afin d'obtenir un classement des antennes testees. L'algorithme cree une nouvelle population d'antennes a partir des antennes les plus mauvaises mais en disposant des ensembles de capteurs dans la configuration des meilleures antennes, les capteurs etant choisis aleatoirement. Cette nouvelle population d'antennes est classee a son tour. Le processus est repete sur plusieurs iterations : la population converge vers une antenne optimale, censee etre la solution du probleme. L'article presente le maillage des positions possibles pour les capteurs. Le fonctionnement de l'algorithme est egalement detaille et sa robustesse est testee sur cent executions ; les proprietes des antennes issues de ces optimisations sont ensuite comparees a celles d'antennes classiques. Enfin, une simulation de localisation de source par ces antennes met en avant la pertinence de l'optimisation realisee.

Verification is one of the most important tasks into the process of systems design. This time-consuming task guarantees the correct functionality of the system. We propose to use HiLeS Designer tool to model and to verify heterogeneous systems in order to reduce functionality risks and time. This tool allows verifying in two ways: formal verification on the model logical sequence and verification by simulation where a virtual prototype on VHDL-AMS is generated from the model. A cane sugar production process is presented as illustrative example.

L'holographie acoustique temps-reel est une methode qui permet de reconstruire le champ acoustique rayonne dans le plan des sources continuellement au cours du temps a partir de mesures effectuees dans le champ proche par une antenne de microphones. Cette methode, qui agit comme si les capteurs acoustiques etaient integres au sein meme des sources sonores, est basee sur un traitement dans l'espace temps-nombre d'onde. Il met en œuvre un filtrage temporel du spectre de nombre d'onde instantane au niveau de l'antenne de microphones par un filtre inverse dont la reponse impulsionnelle est obtenue par inversion de la reponse impulsionnelle directe d'expression analytique connue. Cette inversion est neanmoins rendue delicate par le caractere mal pose du probleme et peut notamment etre realisee par filtrage optimal au sens des moindres carres a l'aide d'un filtre de Wiener. Une autre solution est l'utilisation d'une decomposition en valeurs singulieres couplee a une regularisation de Tikhonov lorsque le produit de convolution dans l'espace temps-nombre d'onde est exprime par un produit matriciel. L'etude presentee concerne la comparaison des champs acoustiques reconstruits au niveau des sources dans le cas de ces deux techniques d'inversion et en particulier lorsqu'un bruit de mesure vient perturber les donnees. Des criteres objectifs permettent d'evaluer la pertinence de la reconstruction des signaux temporels sur le plan des sources mais aussi du champ acoustique spatial retro-propage. Ces criteres mesurent la ressemblance des signaux reconstruits avec des signaux de reference. Dans le cas d'une application numerique avec trois sources monopolaires rayonnant des signaux non stationnaires, les resultats obtenus permettent de juger de l'interet de la technique employant la regularisation qui est moins sensible au bruit de mesure que la methode d'inversion par filtrage optimal. L'holographie acoustique temps-reel apparait ainsi comme un moyen de diagnostiquer au cours du temps l'evolution d'un systeme generant du bruit.