In aeroacoustic experiments of aircraft models and/or components, arrays of microphones are utilized to spatially isolate distinct sources and mitigate interfering noise which contaminates single-microphone measurements. Array measurements are still biased by interfering noise which is coherent over the spatial array aperture. When interfering noise is accounted for, existing algorithms which aim to both spatially isolate distinct sources and determine their individual levels as measured by the array are complex and require assumptions about the nature of the sound field. This work develops a processing scheme which uses spatially-defined phase constraints to remove correlated, interfering noise at the single-channel level. This is achieved through a merger of Conditioned Spectral Analysis (CSA) and the Generalized Sidelobe Canceller (GSC). A cross-spectral, frequency-domain filter is created using the GSC methodology to edit the CSA formulation. The only constraint needed is the user-defined, relative phase difference between the channel being filtered and the reference channel used for filtering. This process, titled Constrained Spectral Conditioning (CSC), produces single-channel Fourier Transform estimates of signals which satisfy the user-defined phase differences. In a spatial sound field mapping context, CSC produces sub-datasets derived from the original which estimate the signal characteristics from distinct locations in space. Because single-channel Fourier Transforms are produced, CSC's outputs could theoretically be used as inputs to many existing algorithms. As an example, data-independent, frequency-domain beamforming (FDBF) using CSC's outputs is shown to exhibit finer spatial resolution and lower sidelobe levels than FDBF using the original, unmodified dataset. However, these improvements decrease with Signal-to-Noise Ratio (SNR), and CSC's quantitative accuracy is dependent upon accurate modeling of the sound propagation and inter-source coherence if multiple and/or distributed sources are measured. In order to demonstrate systematic spatial sound mapping using CSC, it is embedded into the CLEAN algorithm which is then titled CLEAN-CSC. Simulated data analysis indicates that CLEAN-CSC is biased towards the mapping and energy allocation of relatively stronger sources in the field, which limits its ability to identify and estimate the level of relatively weaker sources. It is also shown that CLEAN-CSC underestimates the true integrated levels of sources in the field and exhibits higher-than-true peak source levels, and these effects increase and decrease respectively with increasing frequency. Five independent scaling methods are proposed for correcting the CLEAN-CSC total integrated output levels, each with their own assumptions about the sound field being measured. As the entire output map is scaled, these do not account for relative source level errors that may exist. Results from two airfoil tests conducted in NASA Langley's Quiet Flow Facility show that CLEAN-CSC exhibits less map noise than CLEAN yet more segmented spatial sound distributions and lower integrated source levels. However, using the same source propagation model that CLEAN assumes, the scaled CLEAN-CSC integrated source levels are brought into closer agreement with those obtained with CLEAN. / Ph. D.

This thesis investigates the impedance of acoustic liners, to attenuate noise originating from jet engines and enable compliance with international standards and regulations regarding noise from airplane jet engines. Experimental tests of two supplied liners were conducted in an impedance tube; one liner with known and predictable properties, and one liner with unknown properties. The tests included tonal excitations in the formats of stepped sine and random noise with frequencies within set boundaries. After post-processing of the captured data, the desired impedance could be analysed in terms of excitated frequencies and sound pressure levels. The conclusions from this project are that both of the liners deviated from their expected behavior, which was that liner 1 should have been unaffected by the alternated sound pressure levels, and liner 3 should have shown bigger affection due to the changed sound pressure level. Since the results were different than expected, there might have been minor sources of error during the measurements. It could be investigated if there is leakage from the mounting of the liners, or if the 3D printing resolution is sufficient. Because of limitations in time, there is more left in this project to investigate. Therefore, conducting similar studies where more frequencies, sound pressure levels, and multi-tonal measurements can be included, is suggested as future work.

Vertical-axis wind turbines (VAWTs) have with time been outrivaled by the today more common and economically feasible horizontal-axis wind turbines (HAWTs). However, VAWTs have several advantages which still make them interesting, for example, the VAWTs can have the drive train at ground level and it has been argued that they have lower noise emission. Other proposed advantages are suitability for both up-scaling and floating offshore platforms. The work within this thesis is made in collaboration between Halmstad University and Uppsala University. A 200-kW semi-guy-wired VAWT H-rotor, owned by Uppsala University but situated in Falkenberg close to Halmstad, has been the main subject of the research although most results can be generalized to suit a typical H-rotor. This thesis has three main topics regarding VAWTs: (1) how the wind energy extraction is influenced by turbulence, (2) aerodynamical noise generation and (3) eigenfrequencies of the semi-guy-wired tower. The influence from turbulence on the wind energy extraction is studied by evaluating logged operational data and examining how the power curve and the tip-speed ratio for maximum Cp is impacted by turbulence. The work has showed that the T1-turbine has a good ability to extract wind energy at turbulent conditions, indicating an advantage in energy extraction at turbulent sites for VAWTs compared to HAWTs.The noise characteristics are studied experimentally, and models of the two most likely aerodynamic noise mechanisms are applied. Here, inflow-turbulence noise is deemed as the prevailing noise source rather than turbulent-boundary-layer trailing-edge noise (TBL-TE) which is the most important noise mechanism for HAWTs. The overall noise emission has also been measured and proven low compared to similar sized HAWTs. The eigenfrequencies of a semi-guy-wired tower are also studied. Analytical expressions describing the first-mode eigenfrequency of both tower and guy wire has been derived and verified by experiments and simulations.

This study aims to develop an integrated methodology for finding water leaks in a water distribution network. The integrated methodology is formed from SCADA System, Customer Information System (CIS), and Geographic Information System. The methodology is based on forming district-metered areas (DMA) and sub-DMAs in pressure zones by isolation of the network. Leaking spots in the network are localised by step testing within the DMA. With leak noise loggers leaking spots are localized with an increased accuracy and finally pinpointed by ground microphones. Minimum night flows are observed from the SCADA system before and after the repairs of the leaks to calculate physical water loss percentage in the DMA. Monthly non-revenue water percentage is calculated using the data obtained from SCADA and CIS. With a buffer analysis on the water distribution network data, the benefit of the leak noise loggers is maximized and the working time with the ground microphones are minimized. The methodology is applied in two different DMAs in Antalya water distribution network with different characteristics. In the first DMA, only the developed methodology is applied and a decrease of 19.2% is achieved in physical water losses. In the second DMA, pressure reduction is added to the methodology and a decrease of 4.9% is achieved.

The angular location of an acoustic source can be estimated by measuring an acoustic direction of incidence based solely on the noise produced by the source. Methods for determining the direction of incidence based on sound intensity, the phase of cross-spectral functions, and cross-correlation functions are available. In this current work, we implement Dominant Frequency SElection (DFSE) algorithm. Direction of arrival (DOA) estimation usingmicrophone arrays is to use the phase information present in signals from microphones that are spatially separated. DFSE uses the phase difference between the Fourier transformedsignals to estimate the direction ofarrival (DOA)and is implemented using a three-element ’L’ shaped microphone array, linear microphone array, and planar 16-microphone array. This method is based on simply locating the maximum amplitude from each of the Fourier transformed signals and thereby deriving the source location by solving the set of non-linear least squares equations. For any pair of microphones, the surface on whichthe time difference ofarrival (TDOA) is constant is a hyperboloidoftwo sheets. Acoustic source localization algorithms typically exploit this fact by grouping all microphones into pairs, estimating the TDOA of each pair, then finding the point where all associated hyperboloids most nearly intersect. We make use of both closed-form solutions and iterative techniques to solve for the source location.Acoustic source positioned in 2-dimensional plane and 3-dimensional space have been successfully located.

Dans un premier temps, nous avons exploré la reconnaissance des signatures électromagnétiques des familles d'appareils électriques à l'aide d'un apprentissage supervisé et d'une validation croisée. Les résultats obtenus sont peu fiables. L'exactitude de la classification est de 54.5±26.1% pour les familles d'appareils électriques domestiques testés: la bouilloire, la cuisinière (le four traditionnel et les plaques de cuisson), la hotte de la cuisine, la ventilation de la salle de bain, le fer à repasser, le grille-pain, les luminaires de l'ensemble réfrigérateur-congélateur et les luminaires des lampes. Néanmoins, à partir d'un apprentissage non supervisé, les signatures électromagnétiques permettent tout de même de différencier les appareils selon leur type de charge, résistive ou inductive. Dans un second temps, nous avons cherché du côté des profils de puissance. Les fronts détectés sur le profil de puissance peuvent permettre d'identifier l'allumage et l'arrêt des appareils électriques. Par la suite, à l'aide d'un apprentissage supervisé, les différentiels des fronts de puissance sont associés à un appareil électrique. Le système à été validé lors de la réalisation d'un repas au cours duquel une personne à utiliser l'une des plaques de cuisson, le four traditionnel, la ventilation de la salle de bain, le four à micro-ondes et une machine à expresso. Initialement, le taux de vraie reconnaissance est de seulement 58.2%, le taux de non reconnaissance est de 26.5% et le taux de fausse reconnaissance est de 15.3%. En effet, certains appareils ont des consommations électriques similaires. De plus, plusieurs appareils peuvent fonctionner simultanément. Dans un troisième temps, des règles d'inférences statiques et dynamiques ont été mises en place pour améliorer la reconnaissance des appareils électriques. Par exemple: sur quelle(s) ligne(s) électriques(s) un appareil est branché. Le taux de vraie reconnaissance obtenu passe alors à 95.5%, le taux de non reconnaissance passe à 0.3%. Le système de reconnaissance est limité aux appareils électriques ayant une consommation de puissance supérieure à 70 Watts en raison du bruit électrique présent sur l'une des lignes électriques. Malgré les corrections apportées par les règles d'inférence, certains appareils sont difficilement différenciables. Par exemple: le four traditionnel dans la cuisine et le fer à repasser dans la chambre restent très difficiles à reconnaître. Dans un quatrième temps, nous avons cherché à encore améliorer le taux de vraie reconnaissance en incorporant des informations contextuelles non liées au réseau électrique. L'information apportée par les événements sonores permet de localiser spatialement le résident afin de lever certaines ambiguïtés sur la reconnaissance des appareils. Nous avons développé un système de détection et de localisation des événements sonores basé sur la mise en évidence des transitoires du signal sonore. La localisation de la personne à partir de l'environnement sonore à été validé lors de la réalisation des routines du matin par une personne: se lever, s'habiller, faire sa toilette, préparer et manger un petit déjeuner et enfin s'apprêter pour sortir. La fusion des données sonores et électriques n'a pas été implémentée dans le cadre de cette thèse, mais nos travaux montrent qu'il est possible d'obtenir les données sonores et électriques nécessaires pour ce faire. Finalement, des expérimentations intégrant des assistants cognitifs ont été réalisées avec dix personnes non affectées cognitivement. Le scénario consiste à préparer un pâté chinois, prendre un médicament et faire le repassage. Une application d'aide à la cuisine «SemAssist» et une autre de rappel d'activités «Ap@lz» ont aussi été utilisées au cours de l'expérimentation. Ces expérimentations ont permis d'évaluer en conditions réelles le système de supervision de l'environnement électrique basé sur le profil de puissance. Le taux de vraie reconnaissance est de 87.2±5.0%, le taux de non reconnaissance est de 10.5±4.1% et le taux de fausse reconnaissance est de 2.3±2.7%.

This thesis describes an experimental investigation of the response of lean turbulent swirling flames with different degrees of premixedness (i.e. different mixture patterns) to acoustic forcing using the same burner configuration and varying only the fuel injection strategy. Special emphasis was placed on the amplitude dependence of their response. Also, the behaviour of self-excited fully premixed flames was examined. kHz OH* chemiluminescence was used to study qualitatively the heat release response of the flames, while kHz OH Planar Laser Induced Fluorescence (PLIF) was employed to understand the response of the flame structure and the behaviour of the various parts of the flame. The Proper Orthogonal Decomposition (POD) method was used to extract the dominant structures of the flame and their periodicity. In the first part of the thesis, self-excited oscillations were induced by extending the length of the duct downstream of the bluff body. It was found that the longer the duct length and the higher the equivalence ratio, the stronger the self-excited oscillations were, with the effect of duct length being much stronger. The dominant frequencies of the system were found to increase with equivalence ratio and bulk velocity and decrease with duct length. For some conditions, three simultaneous periodic motions were observed, where the third motion oscillated at a frequency equal to the difference of the other two frequencies. A novel application of the POD method was proposed to estimate the convection velocity from the most dominant reaction zone structures detected by OH* chemiluminescence imaging. For a range of conditions, the convection velocity was found to be in the range of 1.4-1.7 bulk flow velocities at the inlet of the combustor. In the second part, the response of fully premixed, non-premixed with radial fuel injection (NPR) and axial fuel injection (NPA) flames was investigated and compared. All systems exhibited a nonlinear response to acoustic forcing. The highest response was observed by the NPR flame, followed by the fully premixed and the non-premixed with axial fuel injection flame. The proximity of forced flames to blow-off was found to be critical in their heat release response, as close to blow-off the flame response was significantly lower than that farther from blow-off. In the NPR and NPA systems, it was shown that the acoustic forcing reduced the stability of the flame and the stability decreased with the increase in forcing amplitude. In the fully premixed system, the flame area modulations constituted an important mechanism of the system, while in the NPR system both flame area and equivalence ratio modulations were important mechanisms of the heat release modulations. The quantification of the local response of the various parts of the flame at the forcing frequency showed that the ratio RL (OH fluctuation at 160 Hz to the total variance of OH) was greater in the inner shear layer region than in the other parts in the case of NPR and NPA flames. In fully premixed flames, greater RL values were observed in large regions on the downstream side of the flame than those in the ISL region close to the bluff body. The ratio of the convection velocity to the bulk velocity was estimated to be 0.54 for the NPR flame, while it was found to be unity for the respective fully premixed flame. In the last part of the thesis, the response of ethanol spray flames to acoustic oscillations was investigated. The nonlinear response was very low, which was reduced closer to blow-off. The ratio RL was the highest in the spray outer cone region, downstream of the annular air passage, while RL values were very low in the inner cone region, downstream of the bluff body. Unlike NPR and fully premixed flames, in case of spray and NPA systems, it was found that forcing did not affect greatly the flame structure. The understanding of the nonlinear response of flames with different degrees of premixedness in a configuration relevant to industrial systems contributes to the development of reliable flame response models and lean-burn devices, because the degree of premixedness affects greatly the flame response. Also, the understanding of the behaviour of forced spray flames is of great interest for industrial applications, contributing to the development of thermoacoustic models for liquid fuelled combustors. Finally, the estimation of the convection velocity is of importance in the modelling of self-excited flames and flame response models, since the convection velocity affects the flame response significantly.

This project presents the description, design and the implementation of a 4-channel microphone array that is an adaptive sub-band generalized side lobe canceller (GSC) beam former uses for video conferencing, hands-free telephony etc, in a noisy environment for speech enhancement as well as noise suppression. The side lobe canceller evaluated with both Least Mean Square (LMS) and Normalized Least Mean Square (NLMS) adaptation. A testing structure is presented; which involves a linear 4-microphone array connected to collect the data. Tests were done using one target signal source and one noise source. In each microphone’s, data were collected via fractional time delay filtering then it is divided into sub-bands and applied GSC to each of the subsequent sub-bands. The overall Signal to Noise Ratio (SNR) improvement is determined from the main signal and noise input and output powers, with signal-only and noise-only as the input to the GSC. The NLMS algorithm significantly improves the speech quality with noise suppression levels up to 13 dB while LMS algorithm is giving up to 10 dB. All of the processing for this thesis is implemented on a computer using MATLAB and validated by considering different SNR measure under various types of blocking matrix, different step sizes, different noise locations and variable SNR with noise. / Mamun Ahmed E-mail: mamuncse99cuet@yahoo.com

ABSTRACT The detection of an emergency signal can be estimated by the cancellation of surrounding noise and achieving the desired signal in order to alert the automobilist. The aim of the thesis is to detect the emergency signal arriving nearer to the automobilist carrying hearing aids. Recent studies show that this can be achieved by designing various kinds of fixed and adaptive beam formers. A beam former does spatial filtering in the sense that it separates two signals with overlapping frequency content originating from distinctive directions. In this contribution, robust beam former namely Wiener beam former is designed and analyzed collaboratively in a group under the consideration of hearing aid constraints such as the microphone distance. A fractionally delay (FD) are designed to get a maximally flat group delay. The studies had been carried out by comparing noise cancellation algorithms like LMS, NLMS, LLMS and RLS algorithms. By comparing Omni-directional and multi-directional microphones the SNR can be studied. In this thesis work, first proposing appropriate microphone array setup with improved beam forming techniques by using required adaptive algorithm (NLMS) in order to get better quality using the Microphone arrays. Microphone arrays have been widely used to improve the performance of speech recognition systems as well as to benefit for people who need hearing aids. With the help of microphone arrays, it can choose to focus on signals from a specific direction. To getting better signal quality in microphone array using adaptive algorithms, these are help in the noise suppression in accordance with the different beam forming techniques. The proposed system is implemented successfully and validated using MATLAB simulation tool. The emergency signal is different in different countries, so we identify any type of emergency signal by training through neural networks. / Vamshi Krishna Lakum: +46760190899

Il y a plus d'un siècle, les conditions de travail ont fortement évolué sous l'influence de l'industrialisation et notamment à partir de nouvelles méthodes de travail du type Tayloriennes. Des bureaux ouverts à l'Action Office des années 50, c'est plus de 60% de la population active qui est concernée aujourd'hui en Europe. L'évolution des réglementations, liée à la prise de conscience collective des effets psychosomatiques des nuisances sonores, entraîne l'implication par les maîtrises d'ouvrages de bureaux d'études acoustiques pour la préconisation et la mise en œuvre dans la construction et la réhabilitation des espaces tertiaires. La rencontre d'acteurs scientifiques et industriels a donné lieu à ce travail de thèse, étudiant la propagation de l'énergie acoustique pour des espaces dont une des dimensions est différente des autres.La méthode se réduit à la conservation du tenseur énergie-impulsion puis à un système d'équations couplées sur l'intensité acoustique et sur la densité d'énergie. C'est un système hyperbolique d'équations linéaires aux dérivés partielles du premier ordre. Une méthode d'intégration sur une à deux dimensions de l'espace permet d'introduire les coefficients d'absorption et de diffusion moyens. Nous introduisons le potentiel d'intensité et nous écrivons le système sous la forme d'une équation hyperbolique linéaire aux dérivées partielles du second ordre impliquant la densité d'énergie, l'intensité acoustique ou le potentiel d'intensité sur une ou deux dimensions. Nous proposons une méthode analytique approchée permettant de vérifier les résultats à une dimension.Pour la conception acoustique des plateaux de bureaux, la modélisation informatique est un outil remarquable souffrant pourtant de limitations restreignant ses applications. Nous résolvons le formalisme introduit dans ce travail par la méthode des différences finies dans le domaine temporel sur une et deux dimensions. Les schémas utilisés sont stables et explicites et peu couteux en mémoires informatiques. Le fait que nous nous intéressions à une variable énergétique permet de considérer un pas de modélisation spatial important - de l'ordre du mètre - et d'accélérer d'autant les calculs.Le partenariat industriel nous a notamment permis d'accéder à des espaces de type plateaux de bureaux. Nous comparons les résultats des modélisations avec des mesures in situ conduites avec un microphone SoundField ST250 permettant l'estimation de la densité d'énergie et de l'intensité acoustique. / More than a century ago, working conditions have evolved under the influence of industrialization and especially of new management methods such as the Taylorism. From Open-Spaces to Action Offices in the 1950s, more than 60 % of the European working population is concerned today. The evolution of regulations, linked to the collective awareness of the psychosomatic effects of noise, has led clients to request the involvement of acoustical consultants for giving recommendations and supervising their implementation in constructions and rehabilitations of office spaces. This is why scientific and industrial stakeholders joined forces for this thesis dedicated to the propagation of sound energy within rooms characterized by one dimension different from the others.The method developed in this thesis reduces the conservation of the energy-stress tensor to a system of coupled equations for the sound intensity and the sound energy density. It is a hyperbolic system of linear, partial differential equations of first order. Integrating this system on one or two space dimensions leads to the introduction of the mean absorption and diffusion coefficients. We then introduce an intensity potential and write the system in the form of a linear hyperbolic equation involving partial derivatives of second order for the energy density, the sound intensity, or the intensity potential in one or two dimensions. We also propose an analytical approximated method to verify the results in one dimension.For the acoustic design of open-space offices, computer modelling is an outstanding tool. Yet limitations restrict its applications. We solve the equations introduced in this work by the finite-difference time-domain method in the one- and two-dimensional cases. We use stable and explicit schemes that require little computer memory. Considering energy variables allows the use of large spatial steps - of the order of the metre - and accelerates the calculations.The industrial partnership notably gave us access to open-space offices. We compare the results of the modelling with in situ measurements carried out with a SoundField ST250 microphone that makes it possible to estimate the sound energy density and the sound intensity.