Fundação Carlos Chagas Filho de Amparo a Pesquisa do Estado do Rio de Janeiro / Neste trabalho é descrita a teoria necessária para a obtenção da grandeza denominada intensidade supersônica, a qual tem por objetivo identificar as regiões de uma fonte de ruído que efetivamente contribuem para a potência sonora, filtrando, consequentemente, a parcela referente às ondas sonoras recirculantes e evanescentes. É apresentada a abordagem de Fourier para a obtenção da intensidade supersônica em fontes com geometrias separáveis e a formulação numérica existente para a obtenção de um equivalente à intensidade supersônica em fontes sonoras com geometrias arbitrárias. Este trabalho apresenta como principal contribuição original, uma técnica para o cálculo de um equivalente à intensidade supersônica, denominado aqui de intensidade acústica útil, capaz de identificar as regiões de uma superfície vibrante de geometria arbitrária que efetivamente contribuem para a potência sonora que será radiada. Ao contrário da formulação numérica existente, o modelo proposto é mais direto, totalmente formulado na superfície vibrante, onde a potência sonora é obtida através de um operador (uma matriz) que relaciona a potência sonora radiada com a distribuição de velocidade normal à superfície vibrante, obtida com o uso do método de elementos finitos. Tal operador, chamado aqui de operador de potência, é Hermitiano, fato crucial para a obtenção da intensidade acússtica útil, após a aplicação da decomposição em autovalores e autovetores no operador de potência, e do critério de truncamento proposto. Exemplos de aplicações da intensidade acústica útil em superfícies vibrantes com a geometria de uma placa, de um cilindro com tampas e de um silenciador automotivo são apresentados, e os resultados são comparados com os obtidos via intensidade supersônica (placa) e via técnica numérica existente (cilindro), evidenciando que a intensidade acústica útil traz, como benefício adicional, uma redução em relação ao tempo computacional quando comparada com a técnica numérica existente. / This work describes the theory necessary to obtain the greatness called supersonic intensity, which aims to identify the regions of a sound source that effectively contribute to the sound power radiated, filtering recirculating and evanescent sound waves. The Fourier approach to obtain the supersonic intensity in sources having separable geometries, and the existent numerical formulation to obtain an equivalent to supersonic intensity on noise sources with arbitrary geometry. This work presents a new numeric technique for the computation of the numerical equivalent to the supersonic acoustic intensity. The technique provides the identification of the regions of a noise source with arbitrary geometry that effectively contribute to the sound power radiated to the far field by filtering recirculating and evanescent sound waves. The proposed technique is entirely formulated on the vibrating surface. The acoustic power radiated is obtained through a numerical operator that relates it with the distribution of superficial normal velocity, which is obtained by the boundary element method. Such power operator, possesses the property of being Hermitian. The advantage of this characteristic is that their eigenvalues are real and their eigenvectors form an orthonormal set for the velocity distribution. It is applied to the power operator the decomposition in eigenvalues and eigenvectors, becoming possible to compute the numerical equivalent to the supersonic intensity, called here useful intensity, after applying a cutoff criterion to remove the non propagating components. Some numerical tests confirming the effectiveness of the convergence criterions are presented. Examples of the application of the useful intensity technique in vibrating surfaces such as a plate, a cylinder with flat caps and an automotive muffler are presented and the numerical results are discussed, showing that the useful intensity brings, as additional benefit, a reduction of the computational effort, when compared to existent numerical technique.

Des réseaux de capteurs acoustiques sont actuellement mis en place dans plusieurs grandes villes afin d’obtenir une description plus fine de l’environnement sonore urbain. Un des défis à relever est celui de réussir,à partir d’enregistrements sonores, à estimer des indicateurs utiles tels que le niveau sonore du trafic routier. Cette tâche n’est en rien triviale en raison de la multitude de sources sonores qui composent cet environnement. Pour cela, la Factorisation en Matrices Non-négatives (NMF) est considérée et appliquée sur deux corpus de mixtures sonores urbaines simulés. L’intérêt de simuler de tels mélanges est la possibilité de connaitre toutes les caractéristiques de chaque classe de son dont le niveau sonore exact du trafic routier. Le premier corpus consiste en 750 scènes de 30 secondes mélangeant une composante de trafic routier dont le niveau sonore est calibré et une classe de son plus générique. Les différents résultats ont notamment permis de proposer une nouvelle approche, appelée « NMF initialisée seuillée », qui se révèle être la plus performante. Le deuxième corpus créé permet de simuler des mixtures sonores plus représentatives des enregistrements effectués en villes, dont leur réalisme a été validé par un test perceptif. Avec une erreur moyenne d’estimation du niveau sonore inférieure à 1,2 dB, la NMF initialisée seuillée se révèle, là encore, la méthode la plus adaptée aux différents environnements sonores urbains. Ces résultats ouvrent alors la voie vers l’utilisation de cette méthode à d’autres sources sonores, celles que les voix et les sifflements d’oiseaux, qui pourront mener, à terme, à la réalisation de cartes de bruits multi-sources. / Acoustic sensor networks are being set up in several major cities in order to obtain a more detailed description of the urban sound environment. One challenge is to estimate useful indicators such as the road traffic noise level on the basis of sound recordings. This task is by no means trivial because of the multitude of sound sources that composed this environment. For this, Non-negative Matrix Factorization (NMF) is considered and applied on two corpuses of simulated urban sound mixtures. The interest of simulating such mixtures is the possibility of knowing all the characteristics of each sound class including the exact road traffic noise level. The first corpus consists of 750 30-second scenes mixing a road traffic component with a calibrated sound level and a more generic sound class. The various results have notably made it possible to propose a new approach, called ‘Thresholded Initialized NMF', which is proving to be the most effective. The second corpus created makes it possible to simulate sound mixtures more representatives of recordings made in cities whose realism has been validated by a perceptual test. With an average noise level estimation error of less than 1.3 dB, the Thresholded Initialized NMF stays the most suitable method for the different urban noise environments. These results open the way to the use of this method for other sound sources, such as birds' whistling and voices, which can eventually lead to the creation of multi-source noise maps.

Fundação Carlos Chagas Filho de Amparo a Pesquisa do Estado do Rio de Janeiro / Neste trabalho é descrita a teoria necessária para a obtenção da grandeza denominada intensidade supersônica, a qual tem por objetivo identificar as regiões de uma fonte de ruído que efetivamente contribuem para a potência sonora, filtrando, consequentemente, a parcela referente às ondas sonoras recirculantes e evanescentes. É apresentada a abordagem de Fourier para a obtenção da intensidade supersônica em fontes com geometrias separáveis e a formulação numérica existente para a obtenção de um equivalente à intensidade supersônica em fontes sonoras com geometrias arbitrárias. Este trabalho apresenta como principal contribuição original, uma técnica para o cálculo de um equivalente à intensidade supersônica, denominado aqui de intensidade acústica útil, capaz de identificar as regiões de uma superfície vibrante de geometria arbitrária que efetivamente contribuem para a potência sonora que será radiada. Ao contrário da formulação numérica existente, o modelo proposto é mais direto, totalmente formulado na superfície vibrante, onde a potência sonora é obtida através de um operador (uma matriz) que relaciona a potência sonora radiada com a distribuição de velocidade normal à superfície vibrante, obtida com o uso do método de elementos finitos. Tal operador, chamado aqui de operador de potência, é Hermitiano, fato crucial para a obtenção da intensidade acússtica útil, após a aplicação da decomposição em autovalores e autovetores no operador de potência, e do critério de truncamento proposto. Exemplos de aplicações da intensidade acústica útil em superfícies vibrantes com a geometria de uma placa, de um cilindro com tampas e de um silenciador automotivo são apresentados, e os resultados são comparados com os obtidos via intensidade supersônica (placa) e via técnica numérica existente (cilindro), evidenciando que a intensidade acústica útil traz, como benefício adicional, uma redução em relação ao tempo computacional quando comparada com a técnica numérica existente. / This work describes the theory necessary to obtain the greatness called supersonic intensity, which aims to identify the regions of a sound source that effectively contribute to the sound power radiated, filtering recirculating and evanescent sound waves. The Fourier approach to obtain the supersonic intensity in sources having separable geometries, and the existent numerical formulation to obtain an equivalent to supersonic intensity on noise sources with arbitrary geometry. This work presents a new numeric technique for the computation of the numerical equivalent to the supersonic acoustic intensity. The technique provides the identification of the regions of a noise source with arbitrary geometry that effectively contribute to the sound power radiated to the far field by filtering recirculating and evanescent sound waves. The proposed technique is entirely formulated on the vibrating surface. The acoustic power radiated is obtained through a numerical operator that relates it with the distribution of superficial normal velocity, which is obtained by the boundary element method. Such power operator, possesses the property of being Hermitian. The advantage of this characteristic is that their eigenvalues are real and their eigenvectors form an orthonormal set for the velocity distribution. It is applied to the power operator the decomposition in eigenvalues and eigenvectors, becoming possible to compute the numerical equivalent to the supersonic intensity, called here useful intensity, after applying a cutoff criterion to remove the non propagating components. Some numerical tests confirming the effectiveness of the convergence criterions are presented. Examples of the application of the useful intensity technique in vibrating surfaces such as a plate, a cylinder with flat caps and an automotive muffler are presented and the numerical results are discussed, showing that the useful intensity brings, as additional benefit, a reduction of the computational effort, when compared to existent numerical technique.

Acoustic Sound Source localization using signal processing is required in order to estimate the direction from where a particular acoustic source signal is coming and it is also important in order to find a soluation for hands free communication. Video conferencing, hand free communications are different applications requiring acoustic sound source localization. This applications need a robust algorithm which can reliably localize and position the acoustic sound sources. The Steered Response Power Phase Transform (SRP-PHAT) is an important and roubst algorithm to localilze acoustic sound sources. However, the algorithm has a high computational complexity thus making the algorithm unsuitable for real time applications. This thesis focuses on describe the implementation of the SRP-PHAT algorithm as a function of source type, reverberation levels and ambient noise. The main objective of this thesis is to present different approaches of the SRP-PHAT to verify the algorithm in terms of acoustic enviroment, microphone array configuration, acoustic source position and levels of reverberation and noise.

Multichannel acoustic signal processing has undergone major development in recent years due to the increased complexity of current audio processing applications. People want to collaborate through communication with the feeling of being together and sharing the same environment, what is considered as Immersive Audio Schemes. In this phenomenon, several acoustic e ects are involved: 3D spatial sound, room compensation, crosstalk cancelation, sound source localization, among others. However, high computing capacity is required to achieve any of these e ects in a real large-scale system, what represents a considerable limitation for real-time applications. The increase of the computational capacity has been historically linked to the number of transistors in a chip. However, nowadays the improvements in the computational capacity are mainly given by increasing the number of processing units, i.e expanding parallelism in computing. This is the case of the Graphics Processing Units (GPUs), that own now thousands of computing cores. GPUs were traditionally related to graphic or image applications, but new releases in the GPU programming environments, CUDA or OpenCL, allowed that most applications were computationally accelerated in elds beyond graphics. This thesis aims to demonstrate that GPUs are totally valid tools to carry out audio applications that require high computational resources. To this end, di erent applications in the eld of audio processing are studied and performed using GPUs. This manuscript also analyzes and solves possible limitations in each GPU-based implementation both from the acoustic point of view as from the computational point of view. In this document, we have addressed the following problems: Most of audio applications are based on massive ltering. Thus, the rst implementation to undertake is a fundamental operation in the audio processing: the convolution. It has been rst developed as a computational kernel and afterwards used for an application that combines multiples convolutions concurrently: generalized crosstalk cancellation and equalization. The proposed implementation can successfully manage two di erent and common situations: size of bu ers that are much larger than the size of the lters and size of bu ers that are much smaller than the size of the lters. Two spatial audio applications that use the GPU as a co-processor have been developed from the massive multichannel ltering. First application deals with binaural audio. Its main feature is that this application is able to synthesize sound sources in spatial positions that are not included in the database of HRTF and to generate smoothly movements of sound sources. Both features were designed after di erent tests (objective and subjective). The performance regarding number of sound source that could be rendered in real time was assessed on GPUs with di erent GPU architectures. A similar performance is measured in a Wave Field Synthesis system (second spatial audio application) that is composed of 96 loudspeakers. The proposed GPU-based implementation is able to reduce the room e ects during the sound source rendering. A well-known approach for sound source localization in noisy and reverberant environments is also addressed on a multi-GPU system. This is the case of the Steered Response Power with Phase Transform (SRPPHAT) algorithm. Since localization accuracy can be improved by using high-resolution spatial grids and a high number of microphones, accurate acoustic localization systems require high computational power. The solutions implemented in this thesis are evaluated both from localization and from computational performance points of view, taking into account different acoustic environments, and always from a real-time implementation perspective. Finally, This manuscript addresses also massive multichannel ltering when the lters present an In nite Impulse Response (IIR). Two cases are analyzed in this manuscript: 1) IIR lters composed of multiple secondorder sections, and 2) IIR lters that presents an allpass response. Both cases are used to develop and accelerate two di erent applications: 1) to execute multiple Equalizations in a WFS system, and 2) to reduce the dynamic range in an audio signal. / Belloch Rodríguez, JA. (2014). PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/40651 / TESIS / Premios Extraordinarios de tesis doctorales

The PAL (Practise and Learn) Original is a ball board consisting of three wooden boards placed in a triangle, developed to practise football players’ passing ability and first touch. The former Swedish international footballer Jessica Landström observed that these ball boards can, if they are improved, help footballers to develop even more skills while practicing alone. Landstr¨om’s idea was to put lamps on top off the ball boards which light up when a certain ball board expects to receive a pass. This would force the player to look up instead of looking at the ball and hence improve their vision. We concluded that speaking also is important within football. So our objective became to follow up on the development of the simple PAL Original to a ball board which rotates towards a sound source. We wanted to achieve this without configuring the PAL Original’s construction. With the purpose of executing the idea we needed to estimate the angle between a sound source and a face of the ball board, rotate the ball board with an electric motor, communicate wirelessly between units and detect a ball hit when the ball board receives a pass. The final prototype consists of two systems, one system executing the sound source localization and rotation and the other system executing the detection of ball hit and wireless communication. The first system uses time difference of arrival (TDOA) between incoming sound for three sound sensors to calculate an angle, which in turn is communicated to a DC motorthat executes the rotation. The other system combines an LED to light up when a pass is expected, an accelerometer to detect a pass, and radio transceivers to communicate with each other. When at least three of these devices are used a randomizing algorithm decides which one should light its LED next when the first one detects a pass. / PAL (Practice and Learn) Original är ett bollplank bestående av tre träskivor placerade i en triangel, utvecklad för att träna fotbollsspelares passningsförmåga och första touch. Jessica Landström, landslagsmeriterad fotbollsspelare, insåg att dessa bollplank kan utvecklas till att hjälpa fotbollsspelare att träna ännu fler områden vid indviduell träning. Landströms ursprungliga idé var att placera en lampa på bollplanket som lyser upp när den förväntar sig en passning, detta för att tvinga spelaren att titta upp istället för att titta på bollen, och därigenom träna spelarens spelförståelse. Vi drog slutsatsen att det också är mycket viktigt med kommunikation i fotboll. Vårt mål blev därför att vidareutveckla PAL Original till ett bollplank som roterar en sida mot en ljudkälla. Vi ville uppnå detta så att det är kompatibelt med PAL Original utan att ändra dess konstruktion. För att genomföra detta behövde vi alltså uppskatta vinkeln mellan en ljudkälla och en sida av det triangulära bollplanket, rotera bollplanket med en motor, kommunicera trådlöst och detektera när bollplanket mottar en passning. Den slutliga prototypen består av två system, ett system som utför lokalisering av ljudkälla samt rotation och ett system som utför detektering av bollträff samt hanterar trådlös kommunikation. Det första systemet utnyttjar tidsskillnad för ankomst, TDOA (Time Difference of Arrival), mellan inkommande ljud till tre ljudsensorer för att beräkna en vinkel, som i sin tur kommuniceras till en likströmsmotor som utför rotationen. Det andra systemet kombinerar en lysdiod som lyser när en passning förväntas, en accelerometer för att detektera att passning mottagits och radiosändare samt mottagare för trådlös kommunikation. När minst tre sådana enheter används, bestämmer en slumpgenerator vilken enhet som ska tända sin lysdiod när den första detekterar en passning.

The PAL (Practise and Learn) Original is a ball boardconsisting of three wooden boards placed in a triangle, developedto practise football players’ passing ability and firsttouch. The former Swedish international footballer JessicaLandstr¨om observed that these ball boards can, if they areimproved, help footballers to develop even more skills whilepracticing alone. Landstr¨om’s idea was to put lamps on topoff the ball boards which light up when a certain ball boardexpects to receive a pass. This would force the player tolook up instead of looking at the ball and hence improvetheir vision.We concluded that speaking also is important withinfootball. So our objective became to follow up on the developmentof the simple PAL Original to a ball board whichrotates towards a sound source. We wanted to achieve thiswithout configuring the PAL Original’s construction.With the purpose of executing the idea we needed toestimate the angle between a sound source and a face ofthe ball board, rotate the ball board with an electric motor,communicate wirelessly between units and detect a ball hitwhen the ball board receives a pass.The final prototype consists of two systems, one systemexecuting the sound source localization and rotation andthe other system executing the detection of ball hit andwireless communication.The first system uses time difference of arrival (TDOA)between incoming sound for three sound sensors to calculatean angle, which in turn is communicated to a DC motorthat executes the rotation.The other system combines an LED to light up whena pass is expected, an accelerometer to detect a pass, andradio transceivers to communicate with each other. Whenat least three of these devices are used a randomizing algorithmdecides which one should light its LED next whenthe first one detects a pass. / PAL (Practice and Learn) Original är ett bollplankbestående av tre träskivor placerade i en triangel, utveckladför att träna fotbollsspelares passningsförmåga och förstatouch. Jessica Landström, landslagsmeriterad fotbollsspelare,insåg att dessa bollplank kan utvecklas till att hjälpafotbollsspelare att träna ännu fler områden vid indviduellträning. Landströms ursprungliga idé var att placeraen lampa på bollplanket som lyser upp när den förväntarsig en passning, detta för att tvinga spelaren att titta uppistället för att titta på bollen, och därigenom träna spelarensspelförståelse.Vi drog slutsatsen att det också är mycket viktigt medkommunikation i fotboll. Vårt mål blev därför att vidareutvecklaPAL Original till ett bollplank som roterar en sidamot en ljudkälla. Vi ville uppnå detta så att det är kompatibeltmed PAL Original utan att ändra dess konstruktion.För att genomföra detta behövde vi alltså uppskattavinkeln mellan en ljudkälla och en sida av det triangulärabollplanket, rotera bollplanket med en motor, kommuniceratrådlöst och detektera när bollplanket mottar en passning.Den slutliga prototypen består av två system, ett systemsom utför lokalisering av ljudkälla samt rotation ochett system som utför detektering av bollträff samt hanterartrådlös kommunikation.Det första systemet utnyttjar tidsskillnad för ankomst,TDOA (Time Difference of Arrival), mellan inkommandeljud till tre ljudsensorer för att beräkna en vinkel, som i sintur kommuniceras till en likströmsmotor som utför rotationen.Det andra systemet kombinerar en lysdiod som lysernär en passning förväntas, en accelerometer för att detekteraatt passning mottagits och radiosändare samt mottagareför trådlös kommunikation. När minst tre sådana enheteranvänds, bestämmer en slumpgenerator vilken enhet somska tända sin lysdiod när den första detekterar en passning.

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / Em engenharia, a modelagem computacional desempenha um papel importante na concepção de produtos e no desenvolvimento de técnicas de atenuação de ruído. Nesse contexto, esta tese investiga a intensidade acústica gerada pela radiação sonora de superfícies vibrantes. De modo específico, a pesquisa enfoca a identificação das regiões de uma fonte sonora que contribuem efetivamente para potência sonora radiada para o campo afastado, quando a frequência de excitação ocorre abaixo da frequência crítica de coincidência. São descritas as fundamentações teóricas de duas diferentes abordagens. A primeira delas, denominada intensidade supersônica (analítica) é calculada via transformadas de Fourier para fontes sonoras com geometrias separáveis. A segunda, denominada intensidade útil (numérica) é calculada através do método dos elementos de contorno clássico para fontes com geometrias arbitrárias. Em ambas, a identificação das regiões é feita pela filtragem das ondas não propagantes (evanescentes). O trabalho está centrado em duas propostas, a saber. A primeira delas, é a apresentação implementação e análise de uma nova técnica numérica para o cálculo da grandeza intensidade útil. Essa técnica constitui uma variante do método dos elementos de contorno (MEC), tendo como base o fato de as aproximações para as variáveis acústicas pressão e velocidade normal serem tomadas como constantes em cada elemento. E também no modo peculiar de obter a velocidade constante através da média de um certo número de velocidades interiores a cada elemento. Por esse motivo, a técnica recebe o nome de método de elemento de contorno com velocidade média (AVBEMAverage Velocity Boundary Element Method). A segunda, é a obtenção da solução forma fechada do campo de velocidade normal para placas retangulares com oito diferentes combinações de condições contorno clássicas. Então, a intensidade supersônica é estimada e comparada à intensidade acústica. Nos ensaios numéricos, a comparação da intensidade útil obtida via MEC clássico e via AVBEM é mostrada para ilustrar a eficiência computacional da técnica aqui proposta, que traz como benefício adicional o fato de poder ser utilizada uma malha menos refinada para as simulações e, consequentemente, economia significativa de recursos computacionais. / In engineering computational modeling plays an important role in product design and development techniques for noise attenuation. In such context, this thesis investigates the acoustic intensity generated by sound radiation from vibrating surfaces. Specifically, the research focuses on the identification that effectively contribute to the radiated sound power into the far-field, especially when the driven frequency occurs below the critical coincidence frequency. The theoretical formulations of two different approaches are described. The first one, called supersonic intensity (analytical) is calculated via Fourier transforms for noise sources with separable geometries. The second, called useful intensity (numerical) is calculated by the method of boundary elements to classic fonts with arbitrary geometries. In both, the identification of regions is done by filtering the non propagating waves (evanescent). The work is focused on the presentation, implementation and analysis of a new numerical technique for calculating the magnitude useful intensity. This technique is a variant of the method-boundary element (BEM), based on the fact that the approximations for the acoustic pressure and normal speed variables are taken as constant in each element. Also in particular way to obtain constant speed by mean of a number of speeds interior of each element. For this reason, the technique is called the Average Velocity Boundary Element Method . The closed form solution of the normal velocity field for rectangular plates in eight cases with distinct combinations of classical boundary conditions is also obtained. Then, the supersonic intensity is estimated and compared to the acoustic intensity. Numerical experiments are performed comparing to the useful intensity via the conventional BEM with theAVBEM in order to illustrate the positive computational features of the method. The numerical results indicate that the proposed method is much more computationally effcient than its standard BEM counterpart, it enabling the use of a coarser mesh.

Em vista da tentência de se criarem intefaces entre humanos e máquinas que cada vez mais permitam meios simples de interação, é natural que sejam realizadas pesquisas em técnicas que procuram simular o meio mais convencional de comunicação que os humanos usam: a fala. No sistema auditivo humano, a voz é automaticamente processada pelo cérebro de modo efetivo e fácil, também comumente auxiliada por informações visuais, como movimentação labial e localizacão dos locutores. Este processamento realizado pelo cérebro inclui dois componentes importantes que a comunicação baseada em fala requere: Detecção de Atividade de Voz (Voice Activity Detection - VAD) e Localização de Fontes Sonoras (Sound Source Localization - SSL). Consequentemente, VAD e SSL também servem como ferramentas mandatórias de pré-processamento em aplicações de Interfaces Humano-Computador (Human Computer Interface - HCI), como no caso de reconhecimento automático de voz e identificação de locutor. Entretanto, VAD e SSL ainda são problemas desafiadores quando se lidando com cenários acústicos realísticos, particularmente na presença de ruído, reverberação e locutores simultâneos. Neste trabalho, são propostas abordagens para tratar tais problemas, para os casos de uma e múltiplas fontes sonoras, através do uso de informações audiovisuais, explorando-se variadas maneiras de se fundir as modalidades de áudio e vídeo. Este trabalho também emprega um arranjo de microfones para o processamento de som, o qual permite que as informações espaciais dos sinais acústicos sejam exploradas através do algoritmo estado-da-arte SRP (Steered Response Power). Por consequência adicional, uma eficiente implementação em GPU do SRP foi desenvolvida, possibilitando processamento em tempo real do algoritmo. Os experimentos realizados mostram uma acurácia média de 95% ao se efetuar VAD de até três locutores simultâneos, e um erro médio de 10cm ao se localizar tais locutores. / Given the tendency of creating interfaces between human and machines that increasingly allow simple ways of interaction, it is only natural that research effort is put into techniques that seek to simulate the most conventional mean of communication humans use: the speech. In the human auditory system, voice is automatically processed by the brain in an effortless and effective way, also commonly aided by visual cues, such as mouth movement and location of the speakers. This processing done by the brain includes two important components that speech-based communication require: Voice Activity Detection (VAD) and Sound Source Localization (SSL). Consequently, VAD and SSL also serve as mandatory preprocessing tools for high-end Human Computer Interface (HCI) applications in a computing environment, as the case of automatic speech recognition and speaker identification. However, VAD and SSL are still challenging problems when dealing with realistic acoustic scenarios, particularly in the presence of noise, reverberation and multiple simultaneous speakers. In this work we propose some approaches for tackling these problems using audiovisual information, both for the single source and the competing sources scenario, exploiting distinct ways of fusing the audio and video modalities. Our work also employs a microphone array for the audio processing, which allows the spatial information of the acoustic signals to be explored through the stateof- the art method Steered Response Power (SRP). As an additional consequence, a very fast GPU version of the SRP is developed, so that real-time processing is achieved. Our experiments show an average accuracy of 95% when performing VAD of up to three simultaneous speakers and an average error of 10cm when locating such speakers.

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / Em engenharia, a modelagem computacional desempenha um papel importante na concepção de produtos e no desenvolvimento de técnicas de atenuação de ruído. Nesse contexto, esta tese investiga a intensidade acústica gerada pela radiação sonora de superfícies vibrantes. De modo específico, a pesquisa enfoca a identificação das regiões de uma fonte sonora que contribuem efetivamente para potência sonora radiada para o campo afastado, quando a frequência de excitação ocorre abaixo da frequência crítica de coincidência. São descritas as fundamentações teóricas de duas diferentes abordagens. A primeira delas, denominada intensidade supersônica (analítica) é calculada via transformadas de Fourier para fontes sonoras com geometrias separáveis. A segunda, denominada intensidade útil (numérica) é calculada através do método dos elementos de contorno clássico para fontes com geometrias arbitrárias. Em ambas, a identificação das regiões é feita pela filtragem das ondas não propagantes (evanescentes). O trabalho está centrado em duas propostas, a saber. A primeira delas, é a apresentação implementação e análise de uma nova técnica numérica para o cálculo da grandeza intensidade útil. Essa técnica constitui uma variante do método dos elementos de contorno (MEC), tendo como base o fato de as aproximações para as variáveis acústicas pressão e velocidade normal serem tomadas como constantes em cada elemento. E também no modo peculiar de obter a velocidade constante através da média de um certo número de velocidades interiores a cada elemento. Por esse motivo, a técnica recebe o nome de método de elemento de contorno com velocidade média (AVBEMAverage Velocity Boundary Element Method). A segunda, é a obtenção da solução forma fechada do campo de velocidade normal para placas retangulares com oito diferentes combinações de condições contorno clássicas. Então, a intensidade supersônica é estimada e comparada à intensidade acústica. Nos ensaios numéricos, a comparação da intensidade útil obtida via MEC clássico e via AVBEM é mostrada para ilustrar a eficiência computacional da técnica aqui proposta, que traz como benefício adicional o fato de poder ser utilizada uma malha menos refinada para as simulações e, consequentemente, economia significativa de recursos computacionais. / In engineering computational modeling plays an important role in product design and development techniques for noise attenuation. In such context, this thesis investigates the acoustic intensity generated by sound radiation from vibrating surfaces. Specifically, the research focuses on the identification that effectively contribute to the radiated sound power into the far-field, especially when the driven frequency occurs below the critical coincidence frequency. The theoretical formulations of two different approaches are described. The first one, called supersonic intensity (analytical) is calculated via Fourier transforms for noise sources with separable geometries. The second, called useful intensity (numerical) is calculated by the method of boundary elements to classic fonts with arbitrary geometries. In both, the identification of regions is done by filtering the non propagating waves (evanescent). The work is focused on the presentation, implementation and analysis of a new numerical technique for calculating the magnitude useful intensity. This technique is a variant of the method-boundary element (BEM), based on the fact that the approximations for the acoustic pressure and normal speed variables are taken as constant in each element. Also in particular way to obtain constant speed by mean of a number of speeds interior of each element. For this reason, the technique is called the Average Velocity Boundary Element Method . The closed form solution of the normal velocity field for rectangular plates in eight cases with distinct combinations of classical boundary conditions is also obtained. Then, the supersonic intensity is estimated and compared to the acoustic intensity. Numerical experiments are performed comparing to the useful intensity via the conventional BEM with theAVBEM in order to illustrate the positive computational features of the method. The numerical results indicate that the proposed method is much more computationally effcient than its standard BEM counterpart, it enabling the use of a coarser mesh.