Le champ proche des jets est une zone où cohabitent des contributions de pression de nature hydrodynamique et acoustique. Leurs caractéristiques sont méconnues, conduisant à une définition controversée des frontières de cette région ainsi qu'à l'impossibilité d'y prédire les niveaux de pression effectivement rencontrés. Cette zone présente un fort intérêt pour l'étude de la dynamique du jet ainsi que pour celle du rôle que peuvent jouer les structures cohérentes dans les mécanismes de génération de bruit aéroacoustique. L'étude présentée ici porte sur un jet subsonique, de nombre de Mach Mj=0,3, et un jet supersonique, Mj=1,4. Elle s'appuie sur des mesures acoustiques en champs proche et lointain ainsi que des mesures de pression champ proche synchrones à des mesures de vitesse dans l'écoulement par vélocimétrie laser à effet Doppler (LDV). Une interaction forte entre les deux contributions de pression est mise en évidence par des pertes de cohérence importantes. Ce phénomène, dont un modèle est proposé, permet de définir précisément la frontière du champ proche par le produit du nombre d'onde et de la position radiale kr=1,3. L'analyse par décomposition orthogonale aux valeurs propres (POD), sur la base de laquelle est également proposée une méthode de normalisation des données, montre le caractère très cohérent du champ de pression proche à basse fréquence ainsi que l'importance de la prise en compte des contributions azimutales. Une séparation des contributions hydrodynamique et acoustique de pression est obtenue grâce à un filtrage POD. La champ de pression instantanée est déterminé sur une surface entourant le jet grâce à une extension au domaine spectral de l'estimation stochastique linéaire (LSE). Ces données sont utilisées pour estimer le rayonnement acoustique du jet par une formulation intégrale de Kirchhoff ainsi que pour extraire la structuration tridimensionnelle de l'écoulement.

An aluminum NACA 0018 airfoil testbed was constructed with 95 static pressure taps and 25 embedded microphones to enable novel time-resolved measurements of surface pressure. The main objective of this investigation is to utilize time-resolved surface pressure measurements to estimate salient flow characteristics in the separated flow region over the upper surface of an airfoil. The flow development over the airfoil was examined using hot wire anemometry and mean surface pressure for a range of Reynolds numbers from 80x103 to 200x103 and angles of attack from 0° to 18°. For these parameters, laminar boundary layer separation takes place on the upper surface and two flow regimes occur: (i) separation is followed by flow reattachment, so that a separation bubble forms and (ii) separation occurs without subsequent reattachment. Measurements of velocity and mean surface pressure were used to characterize the separated flow region and its effect on airfoil performance using the lift coefficient. In addition, the transition process and the evolution of disturbances were examined. The lift curve characteristics were found to be linked to the rate of change of the separation, transition, and reattachment locations with the angle of attack. For both flow regimes, transition was observed in the separated shear layer. Specifically, the amplification of disturbances within a band of frequencies in the separated shear layer resulted in laminar to turbulent transition. Validation of time-resolved surface pressure measurements was performed for Rec = 100x103 at α = 8° and α = 12°, corresponding to regimes of flow separation with and without reattachment, respectively. A comparative analysis of simultaneous velocity and time-resolved surface pressure measurements showed that the characteristics and development of velocity fluctuations associated with disturbances in the separated shear layer can be extracted from time-resolved surface pressure measurements. Specifically, within the separated flow region, the amplitude of periodic oscillations in the surface pressure signal associated with disturbances in the separated shear layer grew in the streamwise direction. In addition, the frequency at the spectral peak of the amplified disturbances in the separated shear layer was identified. Based on the results of the validation analysis, time-resolved surface pressure measurement analysis techniques were applied for a Reynolds number range from 60x103 to 130x103 and angles of attack from 6° to 16°. Within the separated flow region, the streamwise growth of surface pressure fluctuations is distinctly different depending on the flow regime. Specifically, within the separation bubble, the RMS surface pressure fluctuations increase in the streamwise direction and reach a peak just upstream of the reattachment location. The observed trend is in agreement with that observed for other separating-reattaching flows on geometries such as the forward and backward facing step and splitter plate with fence. In contrast to the separation bubble formation, when the separated shear layer fails to reattach to the airfoil surface, RMS surface pressure fluctuations increase in the streamwise direction with no maximum and the amplitude is significantly lower than those observed in the separation bubble. Surface pressure signals were further examined to identify the frequency, convective velocity, and spanwise uniformity of disturbances in the separated shear layer. Specifically, for both flow regimes, the fundamental frequency and corresponding Strouhal number exhibit a power-law dependency on the Reynolds number. Based on the available data for which velocity measurements were obtained in the separated flow region, the convective velocity matched the mean velocity at the wall-normal distance corresponding to the maximum turbulence intensity. A distinct increase in the convective velocity of disturbances in the separated shear layer was found when the airfoil was stalled in comparison to that found in the separation bubble. From statistical analysis of surface pressure signals in the spanwise direction, it was found that disturbances are strongly two-dimensional in the laminar portion of the separated shear layer and become three-dimensional through the transition process.

This thesis was conducted on behalf of Butong AB, who wanted to test and develop an environmental friendly, so called green sound barrier, which combines both art and science.Different configurations of the product were proposed by the company with various filling materials, as it was predicted that the filling materials would be the main sound absorbent among all parts of the structure.The thesis work started by selecting the best of the proposed fillings which could be of interest - that is those which were expected to have high sound absorption coefficients. The selection process was based on experience, reading and advice. The main idea behind the selection process was saving cost for the company as well as effort.Impedance tube method was used for performing the measurements on samples of the green sound barriers, in order to calculate the acoustical properties of each material and every construction, as it was considerably reliable, cheap and fast to use.The measurements were done according to a combination between standards described in ISO 10534-2:1998 and ASTM E2611-09, for performing test measurements using the impedance tube.This master thesis gives an explanation of the predicted absorption characteristics of the green sound barriers including the usage of different fillings, as well as the advantages and disadvantages of using it in real life applications.

People with hearing loss (HL) often express a desire for the particular hearing device that will yield the best speech recognition. The problem with fulfilling that desire is that a vast number of hearing devices are available from which to choose. In recent years, medical device regulatory agencies have generally viewed hearing devices (i.e., hearing aids), particularly the hardware and even the software, as substantially equivalent. The purpose of this manuscript is to: 1) Synthesize a number of variables about the person, environment in which hearing occurs, as well as characteristics of the hearing aids that can impact speech recognition. 2) Describe a created tool entitled Realistic Expectations 2 (RE2), which has application to the clinical needs of estimating expected speech recognition with and without hearing aids. RE2 is available as a supplemental file to this manuscript and may be useful for comparing estimates against measures of speech recognition ability in addition to assisting with the explanation of the operation and limitations of hearing aids. When expected speech recognition is achieved, subsequent development of speech and language can continue based on circumstance, cognitive status, and cultural-specific learning, as well as personal and societal betterment efforts like education, rehabilitation, and therapy.

L'étude porte sur les pertes acoustiques non-linéaires localisées à la sortie d'un tube cylindrique. Trois aspects sont envisagés : tout d'abord la mise en évidence du phénomène par la mesure de l'impédance terminale du tube, puis la modélisation physique du phénomène et enfin l'application aux instruments de musique à vent. Le travail comprend trois parties qui traitent des trois aspects envisagés.<br /><br />Dans une première partie, des mesures de l'impédance terminale réalisées à l'aide d'une méthode à deux microphones montrent que les pertes à la sortie du tube - partie réelle de l'impédance terminale dans le cadre de l'approximation du premier harmonique - augmentent avec l'amplitude de la vitesse acoustique. Les résultats montrent que l'importance de ces pertes dépend fortement du rayon de courbure des bords intérieurs à la sortie de tube. En outre, pour les faibles rayons de courbure, deux régimes sont mis en évidence. L'existence de ces deux régimes est confirmée par des observations utilisant la vélocimétrie par imagerie de particules (PIV) réalisées en collaboration avec l'Université d'Edimbourg : dans les deux cas un anneau tourbillonnaire est formé à la sortie du tube mais dans le cas du premier régime (faibles vitesses acoustiques) l'anneau reste accroché aux bords du tube alors que dans le cas du second régime (fortes vitesses acoustiques) il est expulsé.<br /><br />La seconde partie concerne la modélisation du phénomène dans le but de mieux comprendre les mécanismes physiques mis en jeu. La théorie du bruit tourbillonnaire (``vortex sound theory'') est appliquée afin d'estimer directement les pertes à la sortie du tube. Trois calculs utilisant cette théorie sont menés : le premier, analytique, sur la base d'un unique anneau tourbillonnaire fixe ou mobile, le second à partir des mesures par PIV et le troisième par la méthode numérique dite des réseaux de Boltzmann. Les trois calculs conduisent à des résultats similaires qui démontrent que les pertes non linéaires trouvent leur origine dans la formation d'anneaux tourbillonnaires en sortie de tube. Ces résultats sont confrontés avec succès aux résultats issus des mesures d'impédance. <br /><br />La troisième partie analyse les conséquences que peuvent avoir les pertes non linéaires dans le fonctionnement d'un instrument de musique à trous latéraux. Il est montré expérimentalement et à l'aide de simulations numériques que la dynamique de jeu d'un instrument à vent dépend directement des pertes dans l'instrument et que cette dynamique peut être étendue lorsque les pertes à la sortie sont minimisées par exemple en chanfreinant les bords des trous latéraux.

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.

La spatialisation des champs sonores basses fréquences doit satisfaire des contraintes spécifiques qui interdisent l'utilisation des techniques de restitution sonore classiques. En revanche, certaines stratégies héritées du contrôle actif du bruit offrent une solution intéressante : il s'agit d'utiliser un réseau de haut-parleurs pour contrôler le champ sonore au niveau de microphones situés en surface de la zone dans laquelle on cherche à reproduire les sons. L'objectif de cette thèse est l'étude numérique et expérimentale de ces méthodes surfaciques de reproduction sonore. Dans un premier temps, on a simulé le comportement de dispositifs utilisant ce type de stratégie en conditions de champ libre. Ces simulations ont ensuite été confrontées aux résultats d'une expérience de contrôle actif réalisée en chambre anéchoïque. Finalement, on a réalisé une étude de faisabilité concernant l'utilisation des stratégies surfaciques de reproduction sonore dans un local dédié. Les résultats montrent que ce local pourrait permettre de reproduire avec précision des ondes planes basses fréquences se propageant autour d'un auditeur.

Majorly the intension behind the engineering besides expecting cool and futuristic is to get appropriate eye contact and emotions back into the teleconferencing domain that two dimensional setups simply cannot provide. Under this system, the distant participant can make clear visual communication with particular people in her or his own frame of perspective. Teleconferencing is a communication technology that allows users at two or more different localizations to interact by making a face-to-face assembling environment. TC systems carry both audio, video and data streams throughout the session it has been gaining popularity in all government sectors. From the most recent demonstration of such a fantast manner of teleconferencing from university of southern California, “Attaining visual communication in a One-to-Many 3D Video Teleconferencing System”, receive a 3D teleconferencing developing a 3D teleconferencing is not only concerning the video but also experiencing a 3D audio by users. A 3D audio system can be described as a reliable audio captured by positioning of speakers. In this thesis we effort to develop a 3D audio system where two microphones and two speakers are used. This structure designed based on the behavior of the human ear while capturing sounds. I studied different usable methods for such structure and then I designed a new system which will be robust and friendly user. The idea of this new system from the scientist Zuccarelli’s theory(1983) which he said that human ear not only capture the sounds it emits sounds as well, and he designed holophonic for the recording sounds from human ear in scientific manner but he did not reveal. I took the concept from him then I captured all the positions of sounds in spherical form. I found that the sound is coming from which direction depending on the pattern of the sound signal; to capture the sounds and to find the directions I used interference and diffraction of the head. An ideal microphone arrangement therefore should be able to guide maximum directivity towards speech no matter which direction it initiates. Directional microphone technology has been used in hearing instruments since the late 1960s, and has been shown to effectively improve speech understanding in background noise. In a futurist implementation of directional microphones system can be interested for industrial and medical applications as well. / In this thesis I have taken the reference of 3D video teleconference by Southern California to design 3D audio teleconference. For teleconference only video is not clear, both 3D audio and video give very good communication, expressions, emotions virtually like the remote people are residing just beside us. I have implemented one structure using two microphones and two speakers, I have implemented this structure in real time using Matlab and done experiments practically. I fixed two directional microphones at distance of 17 cm apart. With one speaker I sent signal at a frequency of 2.5 KHz and the positions are varies in spherical form and observed all positions frequency spectrums and signal patterns by using phase delays. Then I have taken two speakers one is just nearer to the microphone to capture the sound coming from microphone that is fixed at 1.25 KHZ. The other speaker is at 2.5 KHz and varies in spherical form, observed all the positions magnitudes, spectrums and patterns. Instead of placing one microphone just nearer to the microphone I just kept one obstacle between microphone and speaker with fixed frequency and the other speaker is again varies spherically and observed all the positions spectrums and patterns. This is called head diffraction. Finally I found all the variations in all directions in signal strength, pattern and in spectrums. I got very great differences in two positions in front and back. I implemented 3d space for Audio Teleconference. From the above results I have concluded as follows. Here I have compared my results before interference and after interference. Before interference I have used one speaker (2.5 KHz) and two microphones and tested signal level in front and back positions. The signal strength in the front position is stronger than the than the back position. In this stage I could not achieve same signal strengths in front and back positions. To achieve this I have chosen interference with two speakers .One speaker is placed at a fixed position near to the microphones with constant emitting frequency (1.25 KHz) and other speaker is moving in all directions. In this method again I compared my signal in front and back positions. Here the signal strength is almost same in both positions. Finally I have tried to implement same method with head diffraction. In this method again the signal strength is fluctuated in front and back positions. Finally the implemented method is best method for audio teleconferencing room. Using this method we can communicate from any direction of the teleconference room. No need to sit exactly nearer to the microphone. The audio signal strength is almost similar from all the directions of the teleconference room. If there are any obstacles in the teleconference room this method will not be successful.

L'apparition récente de microphones numériques MEMS a ouvert de nouvelles perspectives pour le développement de systèmes d'acquisition acoustiques massivement multi-canaux de grande envergure. De tels systèmes permettent de localiser des sources acoustiques avec de bonnes performances. En revanche, de nouvelles contraintes se posent. La première est le flux élevé de données issues de l'antenne, devant être traitées en un temps raisonnable. La deuxième contrainte est de connaître la position des nombreux microphones déployés in situ. Ce manuscrit propose des méthodes répondant à ces deux contraintes. Premièrement, une étude du système d'acquisition est présentée. On montre que les microphones MEMS sont adaptés pour des applications d'antennerie. Ensuite, un traitement en temps réel des signaux acquis via une implémentation parallèle sur GPU est proposé. Cette stratégie répond au problème de flux de données. On dispose ainsi d'un outil d'imagerie temps réel de sources large bande, permettant d'établir un diagnostic dynamique de la scène sonore.Deuxièmement, différentes méthodes de calibration géométrique pour la détermination de la position des microphones sont exposées. Dans des conditions réelles d'utilisation, les méthodes actuelles sont inefficaces pour des antennes étendues et à grand nombre de microphones. Ce manuscrit propose des techniques privilégiant la robustesse du processus de calibration. Les méthodes proposées couvrent différents environnements acoustiques réels, du champ libre au champ réverbérant. Leur efficacité est prouvée par différentes campagnes expérimentales. / Recently, digital MEMS microphones came out and have opened new perspectives. One of them is the design of large-aperture and massively multichannel acoustical acquisition systems. Such systems meet good requirements for efficient source localization. However, new problems arise. First, an important data flow comes from the array, and must be processed fast enough. Second, if the large array is set up in situ, retrieving the position of numerous microphones becomes a challenging task. This thesis proposes methods addressing these two problems. The first part exhibits the description of the acquisition system, which has been developed during the thesis. First, we show that MEMS microphone characteristics are suitable for array processing applications. Then, real-time processing of channel signals is achieved by a parallel GPU implementation. This strategy is one solution to the heavy data flow processing issue. In this way, a real-time acoustic imaging tool was developed, and enables a dynamic wide-band diagnosis, for an arbitrary duration.The second part presents several robust geometric calibration methods: they retrieve microphone positions, based only on the array acoustic signals. Indeed, in real-life conditions, the state of the art methods are inefficient with large arrays. This thesis proposes techniques that guarantee the robustness of the calibration process. The proposed methods allow calibration in the different existing soundscapes, from free field to reverberant field. Various experimental scenarios prove the efficiency of the methods.

This study investigates whether different stereo techniques used as a main array influences the preferred level from spot microphones when combined in a mix. Time delay compensation and its influence on spot microphone level was also examined. A clarinet soloist and a violin &amp; piano duo were recorded as stimuli. A listening test was conducted where subjects were asked to set the level on spot microphone channels of a clarinet, and violin in combination with several stereo techniques. A/B, X/Y, ORTF, and Blumlein were examined. In general, results suggested that there wasn’t a significant difference in preferred spot microphone level between stereo techniques. Time delay compensation could not be proven to significantly influence the preferred spot microphone level.