Η παρούσα διπλωματική εργασία εκπονήθηκε στο τμήμα Ηλεκτρολόγων Μηχανικών και Τεχνολογίας Υπολογιστών του Πανεπιστημίου Πατρών και το θέμα της αφορά τη σχεδίαση της ηλεκτρολογικής εγκατάστασης στα αρχιτεχνονικά σχέδια της κύριας αίθουσας του Συνεδριακού και Πολιτιστικού Κέντρου του Πανεπιστημίου Πατρών (Σ.Π.Κ) και στην ακουστική μελέτη του αμφιθεάτρου. Για τη σχεδίαση της ηλεκτρολογικής εσωτερικής εγκατάστασης χρησιμοποιήθηκαν τα σύμβολα και οι κανονισμοί από τον ΕΛΟΤ HD384. Για την κατασκευή των σχεδίων χρησιμοποιήθηκε το λογισμικό AutoCAD 2008 και για την ακουστική μελέτη το λογισμικό CATT-Acoustic v7.2I,το οποίο είναι ένας προσομοιωτής μοντέλου ακουστικής χώρου. Η διπλωματική εργασία χωρίζεται σε 6 κεφάλαια όπου στο 1ο κεφάλαιο γίνεται μια γενική περιγραφή για το Σ.Π.Κ. Στο 2ο κεφάλαιο γίνεται μια γενική περιγραφή για την Εσωτερική Ηλεκτρική Εγκατάσταση (Ε.Η.Ε) , γενικές οδηγίες-κανονισμοί και αναφορά στα διάφορα καλώδια και αγωγούς που χρησιμοποιούνται. Στο 3ο κεφάλαιο, εξετάζονται στοιχεία φωτοτεχνίας και εξηγείται αναλυτικά η σημαντικότητα της χρήσης του σωστού είδους τεχνητού φωτισμού στο Σ.Π.Κ . Στη συνέχεια, στο 4ο κεφάλαιο γίνεται η παρουσίαση του κάθε ηλεκτρικού κυκλώματος που σχεδιάστηκε και τι αγωγοί, ασφάλειες και συσκευές προτείνονται για την υλοποίησή του. Στο 5ο κεφάλαιο, παρουσιάζεται η θεωρία ακουστικής και στο 6ο κεφάλαιο παρουσιάζεται η μεθοδολογία που ακολουθήθηκε για τη διεξαγωγή της ακουστικής μελέτης και παρουσιάζονται τα αποτελέσματα και τα συμπεράσματα της ακουστικής προσομοίωσης. / The present dissertation has been done in the department of Electrical and Computer Engineering in the University of Patras and the subject concerned the design of electrical installation for the architectural plans of Conference and Cultural Center of the University of Patras (C.C.C.), as well as the acoustic study of an amphitheatre inside C.C.C. For the design of this electrical installation symbols and regulations from ELOT HD 384 were used. The construction of the electrical plans was done on the software AUTOCAD 2008 and the acoustic study was done using the CATT-acoustic v.32, which is software for simulating rooms acoustic. The dissertation is divided in to 6 chapters. In the first chapter, a general description of C.C.C. is given. In the 2nd chapter a general description of the electrical installation is given, general instructions and regulations and reference to cables and wires that are used. In the 3rd chapter, elements of light are examined and the importance of using the correct kind of technical light in C.C.C. Next, in the 4th chapter, every electrical circuit that is designed is presented with the appropriate cables, fuses and appliances for its implementation. In the 5th chapter, the acoustic theory is explained and in the 6th chapter the methodology for the acoustic simulation and the results are presented.

Describing planned acoustic design by single number ratings yields a weak link to the subjective event, especially when the single number ratings are interpreted by others than experienced acousticians. When developing infrastructure, tools for decision making needs to address visual and aural perception. Visual perception can be addressed using game engines and this has enabled the establishment of tools for visualizations of planned constructions in virtual reality. Audio engines accounting for sound propagation in the game engine environment are steadily developing and have recently been made available. The aim of this project is to simulate airborne sound insulation by extending the support of recently developed audio engines directed towards virtual reality applications. The case studied was airborne sound insulation between two adjacent rooms in a building, the sound transmitted to the receiving room through the building structure resulting from sound pressure exciting the structural elements in the adjacent source room into vibration. The receiving room composed modelled space in the game engine Unreal Engine and Steam Audio was the considered audio engine. Sound transmission was modelled by filtering based on calculations of transmission loss via direct and flanking paths using the model included in the standard EN 12354-1. It was verified that the filtering technique for modelling sound transmission reproduced attenuations in correspondence with the predicted transmission loss. Methodology was established to quantify the quality of the audio engine room acoustics simulations. A room acoustics simulation was evaluated by comparing the reverberation time derived from simulation with theoretical predictions and the simulated reverberation time showed fair agreement with Eyring’s formula above its frequency threshold. The quality of the simulation of airborne sound insulation was evaluated relating the sound field in simulation to insulation classification by the standardized level difference. The spectrum of the simulated standardized level difference was compared with the corresponding sound transmission calculation for a modelled scenario. The simulated data displayed noticeable deviations from the transmission calculation, caused by the audio engine room acoustics simulation. However, the simulated data exhibited cancellation of favourable and unfavourable deviations from the transmission calculation resulting in a mean difference across the spectrum below the just noticeable difference of about 1 dB. Single number ratings was compared and the simulated single number rating was within the standard deviation of how the transmission model calculates predictions for a corresponding practical scenario measured in situ. Thus, the simulated data shows potential and comparisons between simulated data, established room acoustics simulation software and in situ measurements should further be made to deduce whether the deviations entails defects in the airborne sound insulation prediction or is an error imposed by the audio engine room acoustics simulation.

Les systèmes d’assistance à la réverbération sont des dispositifs électroacoustiques installés dans les salles de spectacle pour moduler leur acoustique en fonction du type de représentation qui s’y déroule. Afin de pouvoir dimensionner ces systèmes en amont de leur installation, ce travail s’intéresse au développement, à la mise en œuvre et à la mise à l’épreuve de modèles prévisionnels de l’effet de ses systèmes sur les caractéristiques acoustiques d’une salle, en se concentrant sur les systèmes dits régénératifs diagonaux. Le premier modèle présenté est basé sur une approche systémique exacte et sur l’utilisation de simulations numériques. Il s’agit d’un modèle dont le principe est déjà décrit dans la littérature spécialisée, mais auquel est intégré ici un algorithme de détermination automatique des paramètres de réglage d’un système d’assistance à la réverbération reproduisant la méthode manuelle. Parce que l’utilisation de simulations numériques impose une modélisation détaillée de la salle et un important temps de calcul, ce modèle n’est pas compatible avec la réactivité demandée lors d’une phase d’avant-projet. Dans cette optique, deux autres modèles bases sur les hypothèses de champ diffus et, par là même, plus rapides d’exécution, sont développés. L’un repose aussi sur une approche systémique exacte, mais utilise la théorie stochastique de l’acoustique géométrique des salles plutôt que des simulations numériques. L’autre repose sur une approche énergétique simple. Les confrontations de ces deux modèles avec celui reposant sur des simulations numériques sont exposées pour cinq salles, en considérant la prévision d’évolution de six indices acoustiques courants due à l’introduction d’un système d’assistance à la réverbération. Il en ressort que ces deux modèles aboutissent à des erreurs prévisionnelles comparables et que celles-ci sont équivalentes à celles des formules de Sabine ou d’Eyring ou de la théorie révisée des champs diffus de Barron et Lee appliquées dans une salle sans système. Parallèlement, l’étude et la prévision de l’effet d’un système d’assistance à la réverbération régénératif en augmentation du couplage de deux espaces mal couples au sein d’une même salle sont présentées. Il est montré que ce type d’utilisation permet effectivement une augmentation du couplage et que celle-ci peut être correctement abordée à partir d’un modèle énergétique développé ici. Il est aussi montré à partir de simulations numériques, que cette utilisation permet d’homogénéiser les caractéristiques acoustiques entre les deux espaces couplés. / Reverberation enhancement systems (RES) are electro-acoustic devices installed in auditoria to adapt the listening conditions according to the performances. In order to design these systems before their installation, this work deals with the development, the implementation and the testing of prediction tools of the effect of these systems on the acoustic characteristics of a room, by focusing on diagonal regenerative RES. The first proposed tool is based on a systemic approach and numerical simulations. This tool, whose principle is already described in the literature, is here enhanced by the use of an algorithm that automatically fine-tunes the system parameters. As numerical simulations require detailed 3D models of rooms and high computation times, this tool is not compatible with the responsiveness needed for a preliminary design stage. With this in mind, two other models based on the diffuse field assumptions are also developed to predict the action of a RES. One of them is also based on a systemic approach but it uses the stochastic geometric theory of room acoustics instead of numerical simulations. The other model is based on a simple energetic approach. The confrontation of the results given by these two models with the results given by the model based on numerical simulations has been carried out in five rooms considering the effect of a RES on six common room acoustic criteria. It appeared that these two models lead to prediction errors similar to those obtained with the Sabine or Eyring’s formula, or the Barron and Lee’s revised theory applied in a room without RES. In the same time, the study of the effect of a RES used to enhance the coupling between two poor coupled room is presented. It is shown that this particular use of a RES actually increases the coupling effect, and that this effect can be correctly described and predicted by an energetic model developed here. It is also shown by numerical simulations that this system can balance the listening conditions between two coupled rooms.

The aim of the diploma thesis was to design the reconstruction of the music hall in the existing building of the private music school D-Music. The building is located in Kromeriz, cadastral area Kromeriz. I emphasize the design from the perspective of acoustics. The thesis also evaluated the noise load from the adjacent road II. class, for design hole fillings. It is a five-storey building with one underground and four above-ground floors. The music hall is located on the first floor and is designed multifunctionaly - for music with mobile acoustic absorbers for speech.

Bei der Überführung eines wissenschaftlichen Laborsystems zur automatischen Spracherkennung in eine reale Anwendung ergeben sich verschiedene praktische Problemstellungen, von denen eine der Verlust an Erkennungsleistung durch umgebende akustische Störungen ist. Im Gegensatz zu additiven Störungen wie Lüfterrauschen o. ä. hat die Wissenschaft bislang die Störung des Raumhalls bei der Spracherkennung nahezu ignoriert. Dabei besitzen, wie in der vorliegenden Dissertation deutlich gezeigt wird, bereits geringfügig hallende Räume einen stark störenden Einfluss auf die Leistungsfähigkeit von Spracherkennern. Mit dem Ziel, die Erkennungsleistung wieder in einen praktisch benutzbaren Bereich zu bringen, nimmt sich die Arbeit dieser Problemstellung an und schlägt Lösungen vor. Der Hintergrund der wissenschaftlichen Aktivitäten ist die Erstellung von funktionsfähigen Sprachbenutzerinterfaces für Gerätesteuerungen im Wohn- und Büroumfeld, wie z.~B. bei der Hausautomation. Aus diesem Grund werden praktische Randbedingungen wie die Restriktionen von embedded Computerplattformen in die Lösungsfindung einbezogen. Die Argumentation beginnt bei der Beschreibung der raumakustischen Umgebung und der Ausbreitung von Schallfeldern in Räumen. Es wird theoretisch gezeigt, dass die Störung eines Sprachsignals durch Hall von zwei Parametern abhängig ist: der Sprecher-Mikrofon-Distanz (SMD) und der Nachhallzeit T60. Um die Abhängigkeit der Erkennungsleistung vom Grad der Hallstörung zu ermitteln, wird eine Anzahl von Erkennungsexperimenten durchgeführt, die den Einfluss von T60 und SMD nachweisen. Weitere Experimente zeigen, dass die Spracherkennung kaum durch hochfrequente Hallanteile beeinträchtigt wird, wohl aber durch tieffrequente. In einer Literaturrecherche wird ein Überblick über den Stand der Technik zu Maßnahmen gegeben, die den störenden Einfluss des Halls unterdrücken bzw. kompensieren können. Jedoch wird auch gezeigt, dass, obwohl bei einigen Maßnahmen von Verbesserungen berichtet wird, keiner der gefundenen Ansätze den o. a. praktischen Einsatzbedingungen genügt. In dieser Arbeit wird die Methode Harmonicity-based Feature Analysis (HFA) vorgeschlagen. Sie basiert auf drei Ideen, die aus den Betrachtungen der vorangehenden Kapitel abgeleitet werden. Experimentelle Ergebnisse weisen die Verbesserung der Erkennungsleistung in halligen Umgebungen nach. Es werden sogar praktisch relevante Erkennungsraten erzielt, wenn die Methode mit verhalltem Training kombiniert wird. Die HFA wird gegen Ansätze aus der Literatur evaluiert, die ebenfalls praktischen Implementierungskriterien genügen. Auch Kombinationen der HFA und einigen dieser Ansätze werden getestet. Im letzten Kapitel werden die beiden Basistechnologien Stimm\-haft-Stimmlos-Entscheidung und Grundfrequenzdetektion umfangreich unter Hallbedingungen getestet, da sie Voraussetzung für die Funktionsfähigkeit der HFA sind. Als Ergebnis wird dargestellt, dass derzeit für beide Technologien kein Verfahren existiert, das unter Hallbedingungen robust arbeitet. Es kann allerdings gezeigt werden, dass die HFA trotz der Unsicherheiten der Verfahren arbeitet und signifikante Steigerungen der Erkennungsleistung erreicht. / Automatic speech recognition (ASR) systems used in real-world indoor scenarios suffer from performance degradation if noise and reverberation conditions differ from the training conditions of the recognizer. This thesis deals with the problem of room reverberation as a cause of distortion in ASR systems. The background of this research is the design of practical command and control applications, such as a voice controlled light switch in rooms or similar applications. Therefore, the design aims to incorporate several restricting working conditions for the recognizer and still achieve a high level of robustness. One of those design restrictions is the minimisation of computational complexity to allow the practical implementation on an embedded processor. One chapter comprehensively describes the room acoustic environment, including the behavior of the sound field in rooms. It addresses the speaker room microphone (SRM) system which is expressed in the time domain as the room impulse response (RIR). The convolution of the RIR with the clean speech signal yields the reverberant signal at the microphone. A thorough analysis proposes that the degree of the distortion caused by reverberation is dependent on two parameters, the reverberation time T60 and the speaker-to-microphone distance (SMD). To evaluate the dependency of the recognition rate on the degree of distortion, a number of experiments has been successfully conducted, confirming the above mentioned dependency of the two parameters, T60 and SMD. Further experiments have shown that ASR is barely affected by high-frequency reverberation, whereas low frequency reverberation has a detrimental effect on the recognition rate. A literature survey concludes that, although several approaches exist which claim significant improvements, none of them fulfils the above mentioned practical implementation criteria. Within this thesis, a new approach entitled 'harmonicity-based feature analysis' (HFA) is proposed. It is based on three ideas that are derived in former chapters. Experimental results prove that HFA is able to enhance the recognition rate in reverberant environments. Even practical applicable results are achieved when HFA is combined with reverberant training. The method is further evaluated against three other approaches from the literature. Also combinations of methods are tested. In a last chapter the two base technologies fundamental frequency (F0) estimation and voiced unvoiced decision (VUD) are evaluated in reverberant environments, since they are necessary to run HFA. This evaluation aims to find one optimal method for each of these technologies. The results show that all F0 estimation methods and also the VUD methods have a strong decreasing performance in reverberant environments. Nevertheless it is shown that HFA is able to deal with uncertainties of these base technologies as such that the recognition performance still improves.

In this Thesis a study on the subject on how to improve the overall sound quality within a room using signal processing, played back using a loudspeaker, was conducted. This is a subject that has gained attention during the recent years, with more and more consumer and professional products including it. The objective was to find techniques that offered perceptually good audio quality covering most of the room, while being robust and stable. The solution was to design a correction system which fulfilled these requirements and took advantage of today’s computing technology. This problem and its solution, as included in this Thesis, expose the reader to an introduction to loudspeaker system design and reproduction, room acoustics, psychoacoustics (how humans perceive sound), signal extraction (pre-processing) and filter design as well as design considerations for all of these components. Different ways that this system can be developed further were also discussed. This thesis was mainly based on the theory explained in Immersive Audio Signal Processing av S. Bharitkar and C. Kyriakakis [1]. The results of experiments show that a well-performing room correction system can be realized using a microphone with a known response and a computer. In most cases the improvement in both audible and measurable audio quality is considerable, with only a few cases where an improvement was not made. Using multiple measurement positions, positions of the microphone, led to a further improvement. On the other hand, it was also shown that having two well-positioned microphones was shown to be close to as performant as covering the whole room, even if a combination measurements over the whole listening area was the best performing approach. / I den här examensuppsatsen utfördes en studie på hur man kan förbättra ljudupplevelsen i ett rum, när ljud spelas upp på en högtalare, genom att använda signalbehanlindning. Detta är ett ämne som blivit mer relevant, med mer och mer avancerade och prisvärda ljudsystem på marknaden. Målet för projektet var att hitta tekniker som gav en förbättring av ljudupplevelsen som både var robust och täckte en större yta av rummet. Lösningen var att designa ett korrektionssystem som uppfyllde kraven och tog vara på de stora beräkningsresurserna som dagens datorer erbjuder. Problemet och dess lösning förklaras tillsammans med en introduktion av varje ämne som påverkar ljuduppspelningen samt vad man kan göra för att motverka de oönskade sidoeffekterna. Det inkluderar områden såsom högtalarsystemkonstruktion, rumsaksustik, signalbearbetning och filterdesign, samt exempel och en diskussion på vidare utvecklingar av projektet. Projektet baserades till stor del på boken Immersive Audio Signal Processing av S. Bharitkar and C. Kyriakakis [1] som beskriver hur man skapar en inneslutande ljudupplevelse via rumskorrigering. Slutresultaten visade att det går att med några få steg bygga ett högtalar- och rumskorrigeringssystem som uppfyller de satta villkoren med mycket god ljudkvalitet. Även de enklare systemen, som bara använder en enstaka mätpunkt, kan korrigera för uppspelningen i ett helt rum med goda resultat. Genom att gå vidare med att undersöka att kombinera flera mätpunkter visades det att bara två välplacerade punkter kan prestera likvärdigt med att mäta över hela lyssningsytan. Däremot visas det att en kombination av mätningar över lyssningytan alltid presterar bäst.

The aim of this thesis was to make a practical tool for low frequency analysis in room acoustics.The need arises from Acad’s experience that their results from simulations using raytracing software deviate in the lower frequencies when compared to field measurements inrooms. The tool was programmed in Matlab and utilizes the Finite Difference Time Domain (FDTD) method, which is a form of rapid finite element analysis in the time domain.A number of tests have been made to investigate the practical limitations of the FDTD method, such as numerical errors caused by sound sources, discretization and simulation time. Boundary conditions, with and without frequency dependence, have been analysed bycomparing results from simulations of a virtual impedance tube and reverberation room to analytical solutions. These tests show that the use of the FDTD method appears well suited for the purpose of the tool.A field test was made to verify that the tool enables easy and relatively quick simulations of real rooms, with results well in line with measured acoustic parameters. Comparisons of the results from using the FDTD method, ray-tracing and finite elements (FEM) showed goodcorrelation. This indicates that the deviations Acad experience between simulated results and field measurements are most likely caused by uncertainties in the sound absorption data used for low frequencies rather than by limitations in the ray-tracing software. The FDTDtool might still come in handy for more complex models, where edge diffraction is a more important factor, or simply as a means for a “second opinion” to ray-tracing - in general FEM is too time consuming a method to be used on a daily basis.Auxiliary tools made for importing models, providing output data in the of room acoustic parameters, graphs and audio files are not covered in detail here, as these lay outside the scope of this thesis. / Målet för detta examensarbete var att undersöka möjligheten att programmera ett praktisktanvändbart verktyg för lågfrekvensanalys inom rumsakustik. Behovet uppstår från Acadserfarenhet att resultat från simuleringar med hjälp av strålgångsmjukvara avviker i lågfrekvensområdeti jämförelse med fältmätningar i färdigställda rum. Verktyget är programmerati Matlab och använder Finite Difference Time Domain (FDTD) metoden, vilket är en typav snabb finita elementanalys i tidsdomänen.En rad tester har genomförts för att se metodens praktiska begräsningar orsakade av numeriskafel vid val av ljudkälla, diskretisering och simuleringstid. Randvillkor, med och utanfrekvensberoende, har analyserats genom jämförelser av simulerade resultat i virtuella impedansröroch efterklangsrum mot analytiska beräkningar. Testerna visar att FDTD-metodentycks fungerar väl för verktygets tilltänkta användningsområde.Ett fälttest genomfördes för att verifiera att det med verktyget är möjligt att enkelt och relativtsnabbt simulera resultat som väl matcher uppmätta rumsakustiska parametrar. Jämförelsermellan FDTD-metoden och resultat beräknade med strålgångsanalys och finita elementmetoden(FEM) visade även på god korrelation. Detta indikerar att de avvikelser Acaderfar mellan simulerade resultat och fältmätningar troligen orsakas av osäkerheter i den ingåendeljudabsorptionsdata som används för låga frekvenser, snarare än av begränsningar istrålgångsmjukvaran. Verktyget kan fortfarande komma till användning för mer komplexamodeller, där kantdiffraktion är en viktigare faktor, eller helt enkelt som ett sätt att få ett”andra utlåtande” till resultaten från strålgångsmjukvaran då FEM-analys generellt är en förtidskrävande metod för att användas på daglig basis.Kringverktyg skapade för t.ex. import av modeller, utdata i form av rumsakustiska parametrar,grafer och ljudfiler redovisas inte i detalj i denna rapport eftersom dessa ligger utanförexamensarbetet.

H Διδακτορική Διατριβή μελετά το πρόβλημα της ψηφιακής ισοστάθμισης,σκοπεύοντας στην ανάπτυξη αποτελεσματικών μεθόδων εξάλειψης των ηχητικών παραμορφώσεων, που εισάγονται κατά την ηχητική αναπαραγωγή εξαιτίας της απόκρισης, είτε των ηχείων (ανηχωική ισοστάθμιση), είτε των χώρων ακρόασης (εξάλειψη αντήχησης). Αναπτύσσονται πρωτότυπες μέθοδοι που αφενός εξασφαλίζουν ακριβείς μετρήσεις των ανηχωικών ηλεκτρακουστικών αποκρίσεων μέσα σε μη ανηχωικούς χώρους, αφετέρου πετυχαίνουν κατάλληλη εξομάλυνση των πολύπλοκων αποκρίσεων των ακουστικών συστημάτων για χρήση στην ψηφιακή ισοστάθμιση αλλά και για χρήση σε άλλες εφαρμογές της ακουστικής χώρων που απαιτούν ανάλυση συγκεκριμένων ιδιοτήτων αυτών των συστημάτων. Η συστηματική μελέτη της μεθόδου εξάλειψης αντήχησης που βασίζεται στην ιδανική αντιστροφή των αποκρίσεων χώρων οδηγεί στο πρωτότυπο συμπέρασμα ότι τα ακουστά οφέλη από την εφαρμογή της μεθόδου σε πραγματικό χρόνο είναι σημαντικά υποδεέστερα από τα αναμενόμενα που προκύπτουν από τα αντίστοιχα πειράματα εξομοίωσης αυτής της μεθόδου. Το πρόβλημα της εξάλειψης αντήχησης αντιμετωπίζεται για πρώτη φορά με έναν πρακτικά βιώσιμο τρόπο, με την εισαγωγή πρωτότυπης μεθόδου ισοστάθμισης που βασίζεται στην Μιγαδική Εξομάλυνση των αποκρίσεων χώρων. / The dissertation studies the digital audio equalization problem, in order to develop methods that would effectively eliminate the audio distortions being introduced during the sound reproduction by either the loudspeakers(anechoic equalization) or the room response (dereverberation). Novel methods are introduced that ensure precise measurements of anechoic electracoustic responses inside reverberant enclosures and on the other hand, achieve appropriately smoothed acoustic responses, for use in digital equalization and also in other applications of room acoustics that require analysis of concrete properties of these systems. Novel conclusions have been drawn by the analytic study of the room acoustics dereverberation based on ideal inverse filtering, indicating that the application of such a method in real time yields a significantly degraded performance compared to that achieved by the corresponding simulated dereverberation experiments. The problem of dereverberation is faced with a practically viable solution, with the introduction of a novel method based on the room response Complex Smoothing.

Esta tesis está centrada en el modelado de la acústica de salas en espacios cerrados mediante el uso de una ecuación de transferencia radiativa y una ecuación de difusión En este trabajo se investiga cómo a través de estos modelos teóricos se pueden simular el campo sonoro en espacios complejos. Recientemente, el modelo de la ecuación de fusión ha sido prppuesto para ser utilizado en el modelado de la acústica de salas con superficies que reflejan el sonido de forma totalmente difusa. Este enfoque del uso de la ecuación de la disusión de sido intensamente investigado en los últimos años, ya que proporciona una alta eficiencia y flexibilidad para simular las distribuciones del campo sonoro en diferentes tipos de salas; sin embargo, sólo se han realizado unas pocas investigaciones con el objetivo de indagar sobre la precisión y las limitaciones de este método alternativo. Por lo tanto, en primer lugar se presenta un modelo basado en la ecuación de transferencia por radiación siendo meta principal el unificar una amplia gama de métodos geométricos de modelado de acústica de salas. Además, esta tesis está especialmente dedicada a establecer las bases y suposiciones que permitan obtener un modelo de difusión acústica como particularización del modelo de transferencia radiativa con el objetivo de conseguir una descripción clara y adecuada de sus ventajas y limitaciones desde el punto de vista teórico. Este trabajo permite enlazar directamente al modelo de la ecuación de difusión con el grupo de métodos de la acústica geométrica reforzando sus características y permitiendo una adecuada comparación con estos métodos ampliamente reconocidos. Una vez realizado este análisis teórico, esta tesis también se dedica a cuestiones relativas a la implementación numérica del modelo acústico de la ecuación de difusión . En este trabajo, se modela el campo sonoro a través de esquemas en diferencias finitas. Los resultados de este estudio proporcionan soluciones simples y practicas que muestran unos requerimientos computacionales bajos tanto de consumo de memoria como de tiempo. / Navarro Ruiz, JM. (2012). Discrete-time modelling of diffusion processes for room acoustics simulation and analysis [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/14861 / Palancia

Most acoustic measurements and parameters provided by a manufacturer of acoustic elements, which are offering additional solutions to room acoustics as well as acoustic construction works, are mainly limited to the parameters associated with absorption of individual elements. Until now, these diffusional elements have been neglected. Diffusi- onal panels are used to e.g. eliminate direct reflection of sound waves to the listener or reflection of sound waves concentrated at one point. Combination of absorptive acoustic panels and diffusion elements results in a space that is customised to the submitter’s needs.