Звукоизоляция материалов является ключевым вопросом в оценке конкурентоспособности на строительном рынке. Поэтому, для удовлетворения новым и перспективным нормам по шуму необходимы прорывные решения, основанные на глубоких научных исследованиях, создание работоспособных методов и методик оценки звукоизоляции строительных материалов и повышения коэффициента звукопоглощения. Для реализации этих целей требуются научно-исследовательские лаборатории, имеющие достаточную материально-техническую базу, удовлетворяющие требованиям стандартов. Главной целью диссертации является создание акустической реверберационной камеры для испытаний строительных изделий и конструкций на звукоизоляцию. Также диссертация посвящена проведению первичной аттестации камеры. / Soundproof materials is a key issue in assessing competitiveness in the construction market. Therefore, to meet new and promising noise norms, breakthrough solutions based on in-depth scientific research, the creation of efficient methods and techniques for assessing the sound insulation of building materials and increasing the sound absorption coefficient are necessary. To achieve these goals, research laboratories that have sufficient material and technical facilities that meet the requirements of standards are required. The main goal of the dissertation is to create an acoustic reverberation chamber for testing construction products and structures for soundproofing. Also the Thesis is devoted to the primary certification of the camera.

Detection of an underwater target with active sonar in shallow waters such as the Baltic sea is a big challenge. This since the sound beams from the sonar will be reflected on the surfaces, sea surface and sea bottom, and the water volume itself which generates reverberation. Reverberation which will be reflected back to the receiver, is strong in intensity which give rise to many false targets in terms of classifying a target in a surveillance area. These false targets are unwanted and a real target might benefit from these miss-classifications in terms of remaining undetected. It is especially hard if the signal-to-noise ratio (SNR) is approaching zero, i.e. the target strength and the reverberation strength are equal in magnitude. The classical approach to a target detection problem is to assign a threshold value to the measurement, and the data point exceeding the threshold is classified as a target. This approach does not hold for low levels of SNR, since a threshold would not have a statistical significance and could lead to neglecting important data. Track-before-detect (TrBD) is a proposed method for low-SNR situations which tracks and detects a target based on unthresholded data. TrBD enables tracking and detecting of weak and/or stealthy targets. Due to the issues with target detection in shallow waters, the hypothesis of this thesis is to investigate the possibility to implement TrBD, and evaluate the performance of it, when applied on a low-SNR target. The TrBD is implemented with a particle filter which is a recursive Bayesian solution to the problem of integrated tracking and detection. The reverberation data was generated by filtering white noise with an Autoregressive filter of order 1. The target is assigned to propagate according to a constant velocity state space model. Two types of TrBD algorithms are implemented, one which is trained on the background and one which is not. The untrained TrBD is able to track and detect the target but only for levels of SNR down to 4dB. Lower SNR leads to the algorithm not being able to distinguish the target signal from the reverberation. The trained TrBD on the other hand, is able to perform very well for levels of SNR down to 0dB, it is able to track and detect the target and neglect the reverberation. For trajectories passing through areas with high reverberation, the target was lost for a short period of time until it could be retracked again. Overall, the TrBD was successfully implemented on the self-generated data and has a good performance for various target trajectories.

Thesis (PhD)--Stellenbosch University, 2014. / ENGLISH ABSTRACT: Electromagnetic shielding plays a significant role in the protection of electronic equipment. Its application is essential for mitigating radio-frequency interference for the Karoo Array Telescope (MeerKAT) project in the Karoo region of the Northern Cape. In this context, time-domain (TD) methodology for small enclosure shielding effectiveness (SE) is developed using a reverberated environment technique. Interest revolves around measurement time speed-up and an extended SE response which covers the under-moded condition of small enclosures. Recommended IEC standard 61000-4-21 [1] e ciency of 0.75, for log-periodic dipole array (LPDA) antenna, is also validated from a reverberation chamber (RC) characterisation of a printed circuit-board (PCB) LPDA e ciency. A built-in pulse generator and the Square Kilometre Array (SKA) receiver RATTY form the main elements of the TD metrology. For validation purpose, a reference coaxial airline is built. The cable is characterised with computational codes (CST and FEKO) and is also modelled with Vance and Kley's analytical expressions. The results are compared with TD transfer impedance (Zt) measurement within the RC. The study shows that the cable fixture within an RC shapes the cable under-test (CUT) Zt. The airline itself is also introducing an oscillating component within Zt. The resonance is proportional to the CUT length and it is visible within the measurement data and the simulations. It is not, however, taken into account by the theoretical models. The consequence of an incorrect antenna efficiency on RC applications is also addressed using a PCB LPDA antenna efficiency investigation. The unknown LPDA is simulated with CST for the study. The result is compared to an RC measurement validating the IEC 61000-4-21 standard efficiency recommendation of 0.75 [1]. This methodology characterised the unknown antenna parameter from a reference dipole antenna efficiency we investigated with FEKO. Simulated Wheeler-cap techniques permitted the reference antenna validation. We found that an inaccurate LPDA efficiency has little effect at higher frequency if the IEC efficiency is adopted. However, a difference of more than 7 dB can arise at low frequency if the real efficiency differs by more than 0.3 with respect to the IEC value. The study highlights the importance of a correct antenna efficiency for accurate RC applications. The nested-enclosure technique is regarded as the conventional method of investigating small enclosures SE [2]. The technique is in general time-consuming and works for a frequency range higher than three times the enclosure under test (EUT) lowest cut-o value. Our TD metrology covers a frequency band up to 1.4 GHz which coincides with our enclosure under-moded region. The SE characterisation is not well-documented within this particular region. The dissertation contributes to this field using a non-stirred nested-enclosure configuration. In contrast to the conventional use of the nested-enclosure methodology [2], the source is placed here within the EUT and the enclosure is treated as a normal radiator. The SE definition according to the IEEE standard in [2] is followed and the enclosure total transferred-power is computed from the port's reflection coeficient. The approach does not require a stirrer for the EUT. Our measurement shows an SE agreement between the modfied and the appropriate nested-enclosure technique from 390 MHz up to 4 GHz. The investigation is faster, but in addition the TD spectrum gives a more detailed SE response than the FD approach. / AFRIKAANSE OPSOMMING: Elektromagnetiese skerming speel n belangrike rol in die beskerming van elektroniese komponente. Die toepassing daarvan is noodsaaklik vir die versagting van radiofrekwensie steurings in die Karoo Array Telescope (MeerKAT) projek wat tans ontwikkel word in die Karoo omgewing van die Noord-Kaap. In hierdie konteks is daar op n tydsgebied metode vir klein-omhulsel beskermingse ektiwiteit (BE) gefokus, deur gebruik te maak van n weerkatingsomgewing tegniek. Die belangstelling handel om metingstye te verkort en n uitgebreide BE reaksie wat die lae-modus toestand van klein omhulsels dek. Aanbeveelde International Electrotechnical Commission (IEC) standaard 61000-4-21 [2] e ektiwiteit van 0.75, vir log-periodiese dipool-skikking (LPDA) antennas, is ook bekragtig deur n weerkaatsingskamer karakterisering van n gedrukte-stroombaanbord (PCB) LPDS antenna e ektiwiteit. n Ingeboude impulsgenerator en die Square kilometre Array (SKA) ontvanger, RATTY, vorm deel van die hoof elemente van die tydsgebiedmetings. Vir bekragtigings doeleindes is n koaksiale lugtransmissielyn gebou. Hierdie kabel is gekarakteriseer deur numeriese sagteware (CST en FEKO) en is ook gemoduleer met behulp van Vance en Kley se analitiese uitdrukkings. Die resultate is vergelyk met tydsgebied-oordragsimpedansie (Zt) metings, wat in die weerkaatsingskamer gedoen is. Die studie wys dat kabel posisie binne in die weerkaatsingskamer die Zt van die kabel-onder-toets vervorm. Die lugstreep voeg ook n ossillerende component by tot die gemete Zt. Die resonansie is eweredig aan die lengte van die kabel en is duidelik sigbaar binne die meting- en simulasiedata. Dit is egter nie in-berekening gebring in die analitiese modelle nie. Die gevolge van n verkeerde antenna e ektiwiteit in die weerkaatsingskamer toepassing is ook aangespreek deur ondersoek na die e ektiwiteit van n PCB LPDA. Die onbekende LPDA is gesimuleer met CST vir hierdie studie. Die resultate is vergelyk met n weerkaatsingskamer meting, wat die IEC 61000-4-21 standaard e ektiwiteit van 0.75 [1] bekragtig. Die metode karakteriseer die onbekende antenna veranderlike deur n verwysings dipool antenna e ektiwiteit wat ge-ondersoek is in FEKO. Gesimuleerde Wheeler-cap tegnieke het die veri kasie van die verwysings antenna resultate toegelaat. Daar is gevind dat n onakkurate LPDA e ektiwiteit n klein e ek op die ho er frekwensies het, as die IEC e ektiwiteit aangeneem is. Daar kan egter n verskil van 7dB voorkom by laer frekwensies, as die werklike e ektiwiteit met meer as 0.3 van die IEC waarde verskil. Hierdie studie lig die belangrikheid van n korrekte antenna e ektiwiteit uit vir akkurate weerkaatsingskamer toepassings. Die geneste-omhulsel tegniek word beskou as die konvensionele metode vir die bestudering van beskermingse ektiwiteit vir klein omhulsels. Die tegniek is gewoontlik tyd-rowend en werk net vir frekwensies wat drie maal ho er is as die af-sny waarde van die omhulselonder- toets. Ons tydgebiedmeting dek net n frekwensieband tot 1.4GHz, wat ooreenstem met ons omhulsel lae-modus gebied. Die beskermingse ektiwiteit karakterisering, in hierdie spesi eke veld, is nie goed gedokumenteer nie. Hierdie verhandeling dra by tot hierdie veld deur gebruik te maak van n onversteurde geneste-omhulsel kon gurasie. In teenstelling met die konvensionele gebruik van die geneste-omhulsel metode, is die bron geplaas binne die omhulsel-onder-toets en word dit gebruik as n gewone uitstraler. Die de nisie van beskermingse ektiwiteit volgens die IEEE standaard in [2], is gevolg en die totale oordragskrag van die omhulsel is bereken deur gebruik te maak van n poort se re eksie ko e si ent. Die benadering benodig nie n steurder vir die omhulsel-onder-toets nie. Ons metings se beskermingse ektiwiteit het ooreengestem met die veranderde - en die geskikte geneste-omhulsel tegniek, van 390MHz tot 4GHz in die spektrum. Hierdie ondersoek is vinniger en lewer n meer gedetailleerde beskermingse ektiwiteit reaksie as die frekwensiegebied benadering.

Abstract Noise has become an increasing public health problem and has become serious environment pollution in our daily life. This indicates that it is in time to control and reduce noise from traffic and installations in homes and houses. Today a plethora of products are available for business, but none for the private market. The project describes a start up of development of a sound absorbing wall panel for the private market. It will examine whether it is possible to make a wall panel that can lower the sound pressure level with 3 dB, or reach 0.3 s in reverberation time, in a normally furnished bedroom and still follow the demands of price and environmental awareness. To start the project a limitation was made to use the textiles available per meter within the range of IKEA. The test were made according to applicable standards and calculation of reverberation time and sound pressure level using Sabine’s formula and a formula for sound pressure equals sound effect. During the project, tests were made whether it was possible to achieve a sound classification C on a A-E grade scale according to ISO 11654, where A is the best, with only textiles or if a classic sound absorbing mineral wool had to be used. To reach a sound classification C, a weighted sound absorption coefficient (αw) of 0.6 as a minimum must be reached. The project resulted in that it is technical possible to achieve a sound classification C with only textiles even though for this project another combination was chosen for proceeding with the calculations, because of account to price and environmental awareness. The calculations showed that it is possible to lower the reverberation time to 0.3 s in a normally furnished bedroom with 7 wall panels, and to achieve a lowering of the sound pressure level with 3dB with 7 wall panels. This project showed promising results and leave openings for further research with only textiles and further calculations where more factors are taken under consideration to get more precise and reliable results. / Program: Textilingenjörsutbildningen

Thesis (Elec. E.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and, (M.S.)--Massachusetts Institute of Technology, Dept. of Ocean Engineering, 1976. / Microfiche copy available in Archives and Engineering. / Includes bibliographical references. / by James B. Gallemore. / M.S.

The accurate perception of spoken English is influenced by many variables, including the listener's native language, reverberation, and background noise. Few studies of speech perception by bilingual listeners have carefully controlled for second language proficiency and even fewer have presented speech in everyday listening environments that contain noise and reverberation. In the present study, detailed language background, language proficiency information, and individual language samples were collected and the speech stimuli were presented in a variety of quiet, noisy, and reverberant listening environments. The effects of noise and reverberation on the perception of American English monosyllabic words was examined for two groups of young listeners with normal hearing: 1) monolingual American English speakers and 2) Spanish-English bilinguals who acquired both languages prior to age 6 years, exhibited similar spoken proficiency in both languages, and spoke English without a noticeable accent. An innovative test of virtual speech perception was used to assess word recognition in two listening environments typical of everyday communication: a simulated noisy anechoic environment and a simulated noisy reverberant environment. Word recognition was also measured in quiet and in an unprocessed noisy environment. For each noisy listening environment (unprocessed, anechoic, reverberant), three signal-to-noise ratios were employed. Results indicate that early bilingualism negatively affects perception of words presented in noisy listening environments. Significantly poorer word recognition was observed for the bilingual listeners than for the monolingual listeners in all three noisy environments and at all noise levels. Both groups exhibited similar word recognition in quiet. The results were surprising considering the high level of spoken language proficiency exhibited by all bilingual listeners. It is often assumed that highly proficient Spanish-English speakers are equally proficient at understanding English; however, these data indicate that the speech understanding of this group may be overestimated in natural listening situations.

On s'intéresse à la localisation de l'axe du cône de Mach lié au déplacement supersonique d'un source acoustique en milieu réverbérant à l'aide d'un nombre limité de microphones. L'application faisant l'objet du financement doctoral est la localisation de tireurs embusqués, dans le cadre de la protection du combattant. Le principe du retournement temporel stipule que, lors de la réémission des mesures renversées dans le temps depuis leurs positions respectives, tout se passe comme si le champ de pression acoustique se propageait en remontant le temps et focalisait à l'endroit de la distribution de sources. Cette focalisation est d'autant plus efficace que les positions de capteurs ont tendance à former une surface fermée autour des sources. On considère une distribution discrète de microphones à hauteur constante dans une rue rectiligne. Dans une première approche, le déplacement supersonique est modélisé selon le principe de Huygens-Fresnel comme une sommation de sources monopolaires. Cela permet une formulation analytique du problème direct pouvant être résolue par des simulations numériques simples en accord avec un montage expérimental de synthèse de cône de Mach au moyen d'une ligne de haut-parleurs. La résolution du problème inverse se fait en calculant numériquement le champ rétropropagé depuis les positions des microphones jusque dans des tranches verticales disposées le long de la rue. L'utilisation d'un critère statistique d'ordre supérieur permet de réduire la contribution des termes de sources dans le champ rétropropagé, à l'origine d'une divergence du champ de pression autour des positions de microphones. L'axe du cône de Mach peut alors être localisé avec une bonne précision angulaire. Dans une deuxième approche, on considère un front d'onde conique réel en déplacement supersonique. Les temps d'arrivée obéissent à des lois de retard bien précises. Notre modèle de réverbération permet de prédire la plupart des pics correspondant à l'arrivée au point de mesure des signaux impulsionnels réfléchis. La réverbération n'a pas un effet aussi bénéfique dans le cadre d'une source supersonique réelle que dans le cas d'une source statique, ni même que dans celui de la ligne de haut-parleurs, car la source supersonique présente des contraintes géométriques particulières dont le retournement temporel ne tient pas compte, étant donnée la différence de géométrie des ondes directe (un cône) et retournée dans le temps (une superposition d'ondes sphériques). Cependant, la focalisation autour du passage de l'axe est observée dans les tranches horizontales, avec une qualité dépendant fortement de l'agencement de l'antenne de microphones. / We are interested into the localisation of the Mach cone --- created by the supersonic motion of an acoustic source into a reverberant medium --- by means of a limited amount of microphones. The application concerns sniper localisation, hence protection of the soldier. Theory of time-reversal states that when reemitting measures reversed in time from their respective positions, everything happens as if the pressure field were propagating travelling backwards in time and focused on the source distribution. This result is enhanced when the microphone distribution over space forms a closed surface surrounding the sources, according to Kirchhoff-Helmholtz formula. Aiming at a physically and technologically implementable method, we consider a discrete distribution of microphones, disposed at a constant height in a reverberating area reproducing the geometry of a straight narrow street.The problem shows specific constraints due to its specific geometry that we aim to use sparsely. In a first approach, we consider the supersonic displacement from the viewpoint of Huygens-Fresnel principle as a discrete sum of monopolar sources, which allows an analytical formulation of the direct problem solvable by simple numerical simulations. This model is in agreement with an experimental framework, were the Mach cone is synthesized by means of a loudspeaker array. The resolution of the inverse problem is made by computing the time-reversed pressure field into vertical slices disposed along the street. The use of a fourth-order spatiotemporal statistical criterion (kurtosis) allows to reduce unwanted contribution of source term causing a divergence around microphone positions. A maximum of kurtosis is observed around the intersection between the time-reversal slice and the Mach cone axis, allowing localisation of the latter with good angular precision. In a second approach, we investigate the previous method on a real Mach cone. It is necessary to adopt the point of view of dynamical space-time geometry to understand the behaviour of the times of arrival of the different image sources. It is also necessary to take into account the finiteness of the reberberating walls. Our geometrical theory of reverberation allows a modelisation of the direct problem in good agreement with experimental measurements. We show that the effect of reverberation does not enhance time-reversed focalisation as well as it does with a static source. This is mainly due to geometric constraints that are not reproduced in the time-reversed wavefront superposition. The resolution of the inverse problem strongly depends on the disposition of the microphone set: on the 9 configurations, only one allows detection of the axis.

This thesis investigates the design and the analysis of acoustic signal processing algorithms in reverberant rooms. Reverberation poses a major challenge to acoustic signal processing problems. It degrades speech intelligibility and causes many acoustic algorithms that process sound to perform poorly. Current solutions to the reverberation problem frequently only work in lightly reverberant environments. There is need to improve the reverberant performance of acoustic algorithms.¶ The approach of this thesis is to explore how the intrinsic properties of reverberation can be exploited to improve acoustic signal processing algorithms. A general approach to soundfield modelling using statistical room acoustics is applied to analyze the reverberant performance of several acoustic algorithms. A model of the underlying structure of reverberation is incorporated to create a new method of soundfield reproduction.¶ Several outcomes resulting from this approach are: (i) a study of how more sound capture with directional microphones and beamformers can improve the robustness of acoustic equalization, (ii) an assessment of the extent to which source tracking can improve accuracy of source localization, (iii) a new method of soundfield reproduction for reverberant rooms, based upon a parametrization of the acoustic transfer function and (iv) a study of beamforming to directional sources, specifically exploiting the directionality of human speech.¶ The approach to soundfield modelling has permitted a study of algorithm performance on important parameters of the room acoustics and the algorithm design. The performance of acoustic equalization and source tracking have been found to depend not only on the levels of reverberation but also on the correlation of pressure between points in reverberant soundfields. This correlation can be increased by sound capture with directional capture devices. Work on soundfield reproduction has shown that, though reverberation significantly degrades the performance of conventional techniques, by accounting for the reverberation it is possible to design reproduction methods that function well in reverberant environments.

<p>To effectively demonstrate the strength of using digital signal processing when producing sound effects, a sound effects demo is used at the lectures of the course TSRT78, Digital signal processing, which is given at the university in Linköping.</p><p>The amount of effects, that in an instructive way can be used for an educational purpose, are many and the existing version of the sound effects demo is somewhat limited in its range of effects.</p><p>This reports main focus lies in the presentation of what kind of effects which can be interesting in this kind of demo. All of the effects are presented with their background theory and examples on how they can be implemented in software, mainly with the focus on MATLABTM. Investigations on how well the effects can be run in realtime, in the toolbox SimulinkTM, has been made.</p><p>In the report there is also a presentation of a new version of the sound effect demo that has been produced with user friendlieness and further updates in mind. In the new demo all of the effects are implemented, according to their presentations. The report finishes with suggestions for further work on the sound effects demo.</p>

<p>This thesis covers the development of the Reverberation Chamber as a measurement tool for cell phone tests in electronic production. It also covers the development of the Scattered Field Chamber as a measurement tool for simulations of real propagation environments.</p><p>The first part is a more ”general knowledge about Reverberation Chambers”-part that covers some important phenomena like unstirred power and position dependence that might occour in a small Reverberation Chamber used for cell phone tests. Knowing how to deal with these phenomenas, give the possibility to use the chamber as a measurement tool for production tests even though it is too complex for a simple test of the antenna function.</p><p>The second part shows how to alter some important propagation parameters inside the chamber to fit some real world propagation environments. The 3D plane wave distribution, the polarization and the amplitude statistics of the plane waves are all altered with simple techniques that are implementable all together. A small, shielded anechoic box with apertures is used to control 3D plane wave distribution and polarization. Antennas that introduce unstirred power in the chamber are used to control the statistics.</p><p>Keywords: Propagation environment, Mean Effective Gain, Reverberation Chamber, Scattered Field Chamber, Channel model</p>