This dissertation is concerned with the active control of local sound fields, as applied to an active headrest system. Using loudspeakers and microphones, an active headrest is capable of attenuating ambient noise and providing a comfortable acoustic environment for an occupant. A finite element (FE) model of an active headrest is built and analyzed such that the expected noise reduction levels could be quantified for various geometries as well as primary sound field conditions. Both plane wave and diffuse primary sound fields are considered and it is shown that the performance deteriorates for diffuse sound fields. It is then demonstrated that virtual sensing can greatly improve the spatial extent of the quiet zones as well as the attenuation levels. A prototype of the active headrest was constructed, with characteristics similar to those of the FE model, and tested in both anechoic and reverberant sound fields. Multichannel feedforward and feedback control architectures are implemented in real-time and it is shown that adaptive feedback systems are capable of attenuating band-limited disturbances. The spatial attenuation pattern surrounding the head is also measured by shifting the head to various positions and measuring the attenuation at the ears. Two virtual sensing techniques are compared in both feedback and feedforward architectures. The virtual microphone arrangement, which assumes that the primary sound field is equivalent at the physical and virtual locations, results in the best performance when used in a feedback system attenuating broadband disturbances. The remote microphone technique, which accounts for the transfer response between the physical and virtual locations, offers the best performance for tonal primary sound fields. In broadband sound fields, a causal relationship rarely exists between the physical and virtual microphones, resulting in poor performance. / PHD / Excessive noise and vibration levels in aircraft, rotorcraft, launch vehicles, and other aerospace vehicles may create harsh acoustic environments inside the vehicle. In some extreme cases, military applications being a prime example, hearing damage can occur due to the high noise levels associated with certain vehicles. Noise canceling headsets have been proven an effective solution to this problem, although in certain instances their use may not be safe or feasible. In this work, an active noise canceling headrest, or active headrest, is explored as an alternative solution to noise canceling headphones/headsets. An active headrest uses microphones and loudspeakers, typically located non-intrusively behind the head of the seat occupant, to reduce the ambient noise levels in the vicinity of the head and create a comfortable acoustic environment. A thorough investigation of the viability of such a system in a practical vehicle is assessed through the use of theoretical analysis, finite element modeling, and real-time performance experiments. Performance predictions generated using the finite element model were verified by performing real-time experiments, thus providing a level of confidence in additional predictions for alternative headrest geometries and configurations. Factors such as loudspeaker and microphone placement, head movements away from the nominal position, primary acoustic field characteristics, and choice of control strategy are all found to heavily influence the performance of an active headrest. Real-time experiments were performed in anechoic and reverberant sound fields and it is found that the noise canceling capability of the active headrest worsens in reverberant sound fields as compared to free field conditions.

In this thesis, the design and implementation of a Portable Feedback Active Noise Controller Headphone System, which is based on Texas Instruments TMS320VC5416PGE120 Fixed Point DSP, is described. Problems resulted from fixed-point implementation of LMS algorithm and delays existing in digital ANC implementation are determined. Effective solutions to overcome the aforementioned problems are proposed based on the literature survey. Design of the DSP based control card is explained and crucial points about analog-digital-mixed board design for noise sensitive applications are explained. Filtered input LMS algorithm, filtered input normalized LMS algorithm and filtered input sign-sign LMS algorithm are implemented as adaptation algorithms. The advantages and disadvantages of using modified LMS algorithms are indicated. The selection of the parameters of these algorithms is based on theoretical results and experiments. The real time performances of different adaptation algorithms are compared with each other as well as with a commercial analog ANC headphone under different types of artificial and natural noise signals. Moreover, practical conditions such as put on - put off case and dynamic range overflow case are handled with additional software implementations. It is shown that adaptive ANC systems improve the noise reduction significantly when the noise is within a narrow frequency range and this reduction can be applied to a wider frequency range. It is also shown that the problems of digitally implemented adaptive filters which are based on tracking capability, stability, dynamic range and portability can be fixed to challenge with the analog commercial ANC systems.

This thesis presents the practical and novel work in the area of optimal placement of actuators and sensors for sound-structure interaction systems. The work has been done by the author during his PhD candidature. The research is concentrated in systems with non-ideal boundary conditions as in the case in practical engineering applications. An experimental acoustic cavity with five walls of timber and a thin aluminium sheet fixed tightly on the cavity mouth is chosen in this thesis as a good representation of general sound-structure interaction systems. The sheet is intentionally so fixed that it does not satisfy ideal boundary conditions. The existing methods for obtaining optimal sensor-actuator location using analytic models with ideal boundary conditions are of limited use for such problem with non-ideal boundary conditions. The method presented in this thesis for optimal placement of actuators and sensors is motivated by energy based approach and model uncertainty inclusion. The optimal placement of actuator and sensor for the experimental acoustic cavity is used to construct a robust feedback controller based on minimax LQG control design method. The controller is aimed to reduce acoustic potential energy in the cavity. This energy is due to the structure-borne sound inside the sound-structure interaction system. Practical aspects of the method for optimal placement of actuator and sensors are highlighted by experimental vibration and acoustic noise attenuation for arbitrary disturbance using feedback controllers with optimal placement of actuator and sensor. The disturbance is experimentally set to enter the system via a spatial location different from the controller input as would be in any practical applications of standard feedback disturbance rejections. Experimental demonstration of the novel methods presented in this thesis attenuate structural vibration up to 13 dB and acoustic noise up to 5 dB for broadband frequency range of interest. This attenuation is achieved without the explicit knowledge of the model of the disturbance.

Achieving a low noise level is of great interest to the forest machine industry. Traditionally this is obtained by using passive noise reduction, i.e. by using materials for sound isolation and sound absorption. Especially designs to attenuate low frequency noise tend to be bulky and impractical from an installation point of view. An alternative solution to the problem is to use active noise control (ANC). The basic principle of ANC is to generate an anti-noise signal designed to destructively interfere with the unwanted noise. In this thesis two algorithms (Feedback FxLMS and Feedforward FxLMS) are implemented and evaluated for use in the ANC-system. The ANC-system is tuned to the specific environment in the driver’s cabin of a Komatsu forest machine. The algorithms are first tested in a simulated environment and then in real-time inside a forest machine. Simulations are made both in Matlab and in C using both generated signals and recorded signals. The C code is implemented on the Analog Devices Blackfin DSP card BF526. The result showed a significantly reduction of the sound pressure level (SPL) in the driver’s cabin. The noise attenuation obtained using the Feedback FxLMS was approximately 14 dB for a tonal 100 Hz signal and 11 dB using recorded engine noise from a forest machine at 850 rpm.

The design of a battery-powered, portable headphone active noise control system with TI TMS320C5416 DSP is described. The preliminary implementation of the system on a C5416 DSK is also explained. The problems of fixed-point implementation are described and solutions are proposed. Sign-sign Fx-LMS algorithm with a dead-zone is introduced and used as the adaptation algorithm. Effective use of dynamic range to improve the accuracy in filtering operations is discussed. Details of the designed battery-powered DSP board are given and board software development process is explained. The DSK system and designed portable system is compared against two commercially available analog systems under three different types of noises / composition of tones, drill noise and propeller plane cabin noise. The results reveal that adaptive system has better overall performance.

Noise pollution from road traffic is one of the greatest environmental issues in modern day, and the social cost for road traffic noise was estimated to over 16 billion SEK per year in Sweden in2014. Passive or active control methods can be used to reduce the noise. Active control methods or active noise control is more suitable for attenuating noise in lower frequencies. Active noise control reduces noise by eliminating the noise with a secondary source. There are different control strategies to construct an active noise control system, where the update of the secondary sourceis controlled by an algorithm. There are several different algorithms that are possible to use, and one option is to use a Feedforward Filtered-X Least-Mean-Square (FXLMS) algorithm. It uses control positions where the noise is meant to be reduced and reference signals that measure the noise upstream prior the secondary source. FXLMS also uses a model of the secondary source path to the control position in order to ensure convergence of the algorithm. Although the use of multiple reference signals increases the accuracy of the algorithm, it also increases the convergence time and the practical cost of such an installation. Unfortunately, it can require many reference signals to obtain a sufficient noise reduction when the unwanted noise source is complex and has multiple propagation paths.This study investigates the possibility of producing a new, reduced set of reference signals with a linear combination of the original reference signals that still contain the majority of information needed for suficient noise reduction. This new set of reference signals are sometimes called virtual reference signals. Three different methods of virtual reference signals are analysed; first a constant method using singular-value decomposition on the covariance of the reference signals, second another constant method using singular-value decomposition on the covariance of response estimate from each corresponding reference signal, third an adaptive algorithm updating the linear combination to adapt for incoming data. The different strategies are tested on road test measurements at three different constant speeds, 40km=h; 80km=h and 120km=h, and on data generated from a numerical vehicle model in COMSOL.The results from the analysis indicates that the virtual reference signals could sufficiently reproduce information from the original reference signals to obtain a similar noise reduction with fewer reference signals. However, the virtual reference signals with the adaptive algorithm could not manage to track a transient system where the signal amplitudes are varying over time. Further work is needed to analyse the limits and requirements to obtain virtual reference signals that can represent and track a system even for transient events.

[EN] The field of audio signal processing has undergone a major development in recent years. Both the consumer and professional marketplaces continue to show growth in audio applications such as immersive audio schemes that offer optimal listening experience, intelligent noise reduction in cars or improvements in audio teleconferencing or hearing aids. The development of these applications has a common interest in increasing or improving the number of discrete audio channels, the quality of the audio or the sophistication of the algorithms. This often gives rise to problems of high computational cost, even when using common signal processing algorithms, mainly due to the application of these algorithms to multiple signals with real-time requirements. The field of High Performance Computing (HPC) based on low cost hardware elements is the bridge needed between the computing problems and the real multimedia signals and systems that lead to user's applications. In this sense, the present thesis goes a step further in the development of these systems by using the computational power of General Purpose Graphics Processing Units (GPGPUs) to exploit the inherent parallelism of signal processing for multichannel audio applications. The increase of the computational capacity of the processing devices 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 and using parallel processing. The Graphics Processing Units (GPUs), which have now thousands of computing cores, are a representative example. The GPUs were traditionally used to graphic or image processing, but new releases in the GPU programming environments such as CUDA have allowed the use of GPUS for general processing applications. Hence, the use of GPUs is being extended to a wide variety of intensive-computation applications among which audio processing is included. However, the data transactions between the CPU and the GPU and viceversa have questioned the viability of the use of GPUs for audio applications in which real-time interaction between microphones and loudspeakers is required. This is the case of the adaptive filtering applications, where an efficient use of parallel computation in not straightforward. For these reasons, up to the beginning of this thesis, very few publications had dealt with the GPU implementation of real-time acoustic applications based on adaptive filtering. Therefore, this thesis aims to demonstrate that GPUs are totally valid tools to carry out audio applications based on adaptive filtering that require high computational resources. To this end, different adaptive applications in the field 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. / [ES] El campo de procesado de señales de audio ha experimentado un desarrollo importante en los últimos años. Tanto el mercado de consumo como el profesional siguen mostrando un crecimiento en aplicaciones de audio, tales como: los sistemas de audio inmersivo que ofrecen una experiencia de sonido óptima, los sistemas inteligentes de reducción de ruido en coches o las mejoras en sistemas de teleconferencia o en audífonos. El desarrollo de estas aplicaciones tiene un propósito común de aumentar o mejorar el número de canales de audio, la propia calidad del audio o la sofisticación de los algoritmos. Estas mejoras suelen dar lugar a sistemas de alto coste computacional, incluso usando algoritmos comunes de procesado de señal. Esto se debe principalmente a que los algoritmos se suelen aplicar a sistemas multicanales con requerimientos de procesamiento en tiempo real. El campo de la Computación de Alto Rendimiento basado en elementos hardware de bajo coste es el puente necesario entre los problemas de computación y los sistemas multimedia que dan lugar a aplicaciones de usuario. En este sentido, la presente tesis va un paso más allá en el desarrollo de estos sistemas mediante el uso de la potencia de cálculo de las Unidades de Procesamiento Gráfico (GPU) en aplicaciones de propósito general. Con ello, aprovechamos la inherente capacidad de paralelización que poseen las GPU para procesar señales de audio y obtener aplicaciones de audio multicanal. El aumento de la capacidad computacional de los dispositivos de procesado ha estado vinculado históricamente al número de transistores que había en un chip. Sin embargo, hoy en día, las mejoras en la capacidad computacional se dan principalmente por el aumento del número de unidades de procesado y su uso para el procesado en paralelo. Las GPUs son un ejemplo muy representativo. Hoy en día, las GPUs poseen hasta miles de núcleos de computación. Tradicionalmente, las GPUs se han utilizado para el procesado de gráficos o imágenes. Sin embargo, la aparición de entornos sencillos de programación GPU, como por ejemplo CUDA, han permitido el uso de las GPU para aplicaciones de procesado general. De ese modo, el uso de las GPU se ha extendido a una amplia variedad de aplicaciones que requieren cálculo intensivo. Entre esta gama de aplicaciones, se incluye el procesado de señales de audio. No obstante, las transferencias de datos entre la CPU y la GPU y viceversa pusieron en duda la viabilidad de las GPUs para aplicaciones de audio en las que se requiere una interacción en tiempo real entre micrófonos y altavoces. Este es el caso de las aplicaciones basadas en filtrado adaptativo, donde el uso eficiente de la computación en paralelo no es sencillo. Por estas razones, hasta el comienzo de esta tesis, había muy pocas publicaciones que utilizaran la GPU para implementaciones en tiempo real de aplicaciones acústicas basadas en filtrado adaptativo. A pesar de todo, esta tesis pretende demostrar que las GPU son herramientas totalmente válidas para llevar a cabo aplicaciones de audio basadas en filtrado adaptativo que requieran elevados recursos computacionales. Con este fin, la presente tesis ha estudiado y desarrollado varias aplicaciones adaptativas de procesado de audio utilizando una GPU como procesador. Además, también analiza y resuelve las posibles limitaciones de cada aplicación tanto desde el punto de vista acústico como desde el punto de vista computacional. / [CA] El camp del processament de senyals d'àudio ha experimentat un desenvolupament important als últims anys. Tant el mercat de consum com el professional segueixen mostrant un creixement en aplicacions d'àudio, com ara: els sistemes d'àudio immersiu que ofereixen una experiència de so òptima, els sistemes intel·ligents de reducció de soroll en els cotxes o les millores en sistemes de teleconferència o en audiòfons. El desenvolupament d'aquestes aplicacions té un propòsit comú d'augmentar o millorar el nombre de canals d'àudio, la pròpia qualitat de l'àudio o la sofisticació dels algorismes que s'utilitzen. Això, sovint dóna lloc a sistemes d'alt cost computacional, fins i tot quan es fan servir algorismes comuns de processat de senyal. Això es deu principalment al fet que els algorismes se solen aplicar a sistemes multicanals amb requeriments de processat en temps real. El camp de la Computació d'Alt Rendiment basat en elements hardware de baix cost és el pont necessari entre els problemes de computació i els sistemes multimèdia que donen lloc a aplicacions d'usuari. En aquest sentit, aquesta tesi va un pas més enllà en el desenvolupament d'aquests sistemes mitjançant l'ús de la potència de càlcul de les Unitats de Processament Gràfic (GPU) en aplicacions de propòsit general. Amb això, s'aprofita la inherent capacitat de paral·lelització que posseeixen les GPUs per processar senyals d'àudio i obtenir aplicacions d'àudio multicanal. L'augment de la capacitat computacional dels dispositius de processat ha estat històricament vinculada al nombre de transistors que hi havia en un xip. No obstant, avui en dia, les millores en la capacitat computacional es donen principalment per l'augment del nombre d'unitats de processat i el seu ús per al processament en paral·lel. Un exemple molt representatiu són les GPU, que avui en dia posseeixen milers de nuclis de computació. Tradicionalment, les GPUs s'han utilitzat per al processat de gràfics o imatges. No obstant, l'aparició d'entorns senzills de programació de la GPU com és CUDA, han permès l'ús de les GPUs per a aplicacions de processat general. D'aquesta manera, l'ús de les GPUs s'ha estès a una àmplia varietat d'aplicacions que requereixen càlcul intensiu. Entre aquesta gamma d'aplicacions, s'inclou el processat de senyals d'àudio. No obstant, les transferències de dades entre la CPU i la GPU i viceversa van posar en dubte la viabilitat de les GPUs per a aplicacions d'àudio en què es requereix la interacció en temps real de micròfons i altaveus. Aquest és el cas de les aplicacions basades en filtrat adaptatiu, on l'ús eficient de la computació en paral·lel no és senzilla. Per aquestes raons, fins al començament d'aquesta tesi, hi havia molt poques publicacions que utilitzessin la GPU per implementar en temps real aplicacions acústiques basades en filtrat adaptatiu. Malgrat tot, aquesta tesi pretén demostrar que les GPU són eines totalment vàlides per dur a terme aplicacions d'àudio basades en filtrat adaptatiu que requereixen alts recursos computacionals. Amb aquesta finalitat, en la present tesi s'han estudiat i desenvolupat diverses aplicacions adaptatives de processament d'àudio utilitzant una GPU com a processador. A més, aquest manuscrit també analitza i resol les possibles limitacions de cada aplicació, tant des del punt de vista acústic, com des del punt de vista computacional. / Lorente Giner, J. (2015). Adaptive signal processing for multichannel sound using high performance computing [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/58427

É cada vez maior a preocupação com a poluição sonora gerada pelos transformadores de potência de subestações elétricas. Atualmente, o controle desse tipo de ruído é feito utilizando-se métodos passivos, que são caros e dificultam a manutenção dos transformadores. Uma alternativa para os métodos passivos é o controle ativo de ruído (ANC). Apesar de extensas pesquisas realizadas nas últimas três décadas, ainda não existem soluções comercialmente viáveis para o ANC de transformadores. As dificuldades para a aplicação bem sucedida do ANC para transformadores foram investigadas por meio de simulações e de testes com protótipo. Os três maiores obstáculos identificados foram: o posicionamento dos transdutores eletroacústicos; a obtenção de atenuação em uma região longe do transformador, usando um número reduzido de fontes de controle e de sensores de erro, os últimos colocados ainda na região de campo acústico próximo; e a identificação robusta do caminho secundário com baixa razão sinal/ruído. Os dois primeiros problemas foram abordados, analisando-se algumas alternativas de soluções. Algoritmos genéticos (GA) foram utilizados para a otimização da posição dos transdutores do sistema ANC. O desempenho desses algoritmos depende fortemente da modelagem acústica realizada e verificou-se que o método de Usry, escolhido para modelar o campo primário do transformador, não forneceu estimativa adequada. Usando um modelo mais simples de fonte primária, constatou-se a importância da função de mérito para o desempenho do GA. Também foi verificado que a otimização conjunta das posições dos transdutores fornece o mesmo resultado, e em menor tempo, que a otimização das posições das fontes de controle e dos sensores de erro separadamente. Simulações realizadas com uma nova estratégia de sensores virtuais (baseada no janelamento das fontes de controle) mostra que é possível aumentar o nível de atenuação longe do transformador, mesmo com um número pequeno de fontes de controle e sensores de erro. Testes com um protótipo de sistema ANC foram feitos em laboratório e em campo e os resultados desses testes são discutidos detalhadamente. / Concern regarding noise pollution caused by power transformers in electrical substations is increasing. Nowadays, this kind of noise is controlled using passive methods, which are expensive and make transformer maintenance more difficult. An alternative to passive methods is active noise control (ANC). However, despite extensive research undertaken in the last three decades, there is still no viable commercial solution for the active control of transformer noise. The difficulties for a successful implementation of an ANC solution in the case of power transformer noise are investigated through simulations and tests with a prototype. The three main obstacles found were: the positioning of the electro-acoustic transducers; the achievement of sufficient attenuation in a region far from the transformer, using a small number of control sources and error sensors (when the latter are positioned on the region of acoustic near-field); and the robust identification of the secondary path in a low signal/noise situation. The two former problems were dealt with, and some alternative solutions were analyzed. Genetic algorithms (GA) were used for the optimization of the transducers\' position. The performance of these algorithms is strongly related to the acoustical model used and it was verified that the Usry method, used for modelling the transformers primary field, did not result in an adequate estimate. Using a simplified model for the primary source, the importance of the cost function in the GA\'s performance was made evident. It was also verified that the joint optimization of transducers\' position provides the same result, and in shorter time, as the independent optimization of control source and error sensor positions. Simulations with a new virtual sensor strategy (based on windowing the control sources) show that it is possible to increase attenuation levels in a region far from the transformer, even with a small number of control sources and error sensors. Laboratory and field tests with an ANC system prototype were undertaken and the results of these tests are thoroughly discussed.

Cette thèse porte sur le contrôle acoustique actif (ANC) dans une cavité. L’objectif est d’atténuer l’effet d’une onde sonore perturbatrice en des points ou dans un volume. Ceci est réalisé à l’aide d’un contre-bruit généré, par exemple, par un haut-parleur. Cette étude requiert l’utilisation de modèles dynamiques rendant compte de l’évolution des pressions aux points d’intérêt en fonction des bruits exogènes. Ce modèle peut être obtenu par une identification fréquentielle des réponses point-à-point ou en utilisant le modèle physique sous jacent (équation des ondes). Dans ce dernier cas, la recherche d'un modèle de dimension finie est souvent un préalable à l’étude conceptuelle d'un système d’ANC. Les contributions de cette thèse portent donc sur l’élaboration de différents modèles simplifiés paramétrés par la position pour les systèmes acoustiques et sur la conception de lois de commande pour l’ANC. Le premier volet de la thèse est dédié à l’élaboration de différents modèles simplifiés de système de propagation acoustique au sein d’une cavité. Pour cela, les simplifications envisagées peuvent être de nature spatiale autant que fréquentielle. Nous montrons notamment qu'il est possible, sous certaines conditions, d’approximer le système 3D par un système 1D. Ceci a été mis en évidence expérimentalement sur le banc d’essai LS2NBox. Le second volet porte sur la conception de lois de commande. En premier lieu, les stratégies de commandes couramment utilisées pour l’ANC sont comparées. L'effet dela commande multi-objectif H en différents points voisins des points d'atténuation est analysé. La possibilité d’une annulation parfaitedu bruit en un point est aussi discutée. / This thesis deals with active noise control (ANC) in a cavity. The aim is to mitigate the effect of a disturbing sound wave at some points or in a volume. This is achieved using ananti-noise generated, for example, by a loudspeaker. This study requires the use of dynamic models that report changes in pressure at points of interest in response to exogenous noises. Such models can be obtained by frequency identification of point-to-point responses or by using the underlying physical model (wave equation). In the latter case, the search for a low-complexity model (finite dimensional model) is often a prerequisite for the conceptual study of an active control system. The contributions of this thesis concern the development of different simplified models parameterized by the spatial position for acoustic systems, and the design of control laws for noise attenuation. The first part of the thesis is dedicated to the development of various simplified models of acoustic propagation system within a cavity. For that, the simplifications envisaged can be of spatial nature as much as frequential. We show in particular that it is possible, under certain conditions, to approximate the 3D system by a 1D system. This has been demonstrated experimentally on the prototype system, LS2NBox. The second part of the thesis deals with the design of control laws. First, the control strategies commonly used for ANC are compared. The effect of multi-objective H control at different spatial positions close to the attenuation points is analyzed. The possibility of perfect noise cancellation at one point is also discussed.

É cada vez maior a preocupação com a poluição sonora gerada pelos transformadores de potência de subestações elétricas. Atualmente, o controle desse tipo de ruído é feito utilizando-se métodos passivos, que são caros e dificultam a manutenção dos transformadores. Uma alternativa para os métodos passivos é o controle ativo de ruído (ANC). Apesar de extensas pesquisas realizadas nas últimas três décadas, ainda não existem soluções comercialmente viáveis para o ANC de transformadores. As dificuldades para a aplicação bem sucedida do ANC para transformadores foram investigadas por meio de simulações e de testes com protótipo. Os três maiores obstáculos identificados foram: o posicionamento dos transdutores eletroacústicos; a obtenção de atenuação em uma região longe do transformador, usando um número reduzido de fontes de controle e de sensores de erro, os últimos colocados ainda na região de campo acústico próximo; e a identificação robusta do caminho secundário com baixa razão sinal/ruído. Os dois primeiros problemas foram abordados, analisando-se algumas alternativas de soluções. Algoritmos genéticos (GA) foram utilizados para a otimização da posição dos transdutores do sistema ANC. O desempenho desses algoritmos depende fortemente da modelagem acústica realizada e verificou-se que o método de Usry, escolhido para modelar o campo primário do transformador, não forneceu estimativa adequada. Usando um modelo mais simples de fonte primária, constatou-se a importância da função de mérito para o desempenho do GA. Também foi verificado que a otimização conjunta das posições dos transdutores fornece o mesmo resultado, e em menor tempo, que a otimização das posições das fontes de controle e dos sensores de erro separadamente. Simulações realizadas com uma nova estratégia de sensores virtuais (baseada no janelamento das fontes de controle) mostra que é possível aumentar o nível de atenuação longe do transformador, mesmo com um número pequeno de fontes de controle e sensores de erro. Testes com um protótipo de sistema ANC foram feitos em laboratório e em campo e os resultados desses testes são discutidos detalhadamente. / Concern regarding noise pollution caused by power transformers in electrical substations is increasing. Nowadays, this kind of noise is controlled using passive methods, which are expensive and make transformer maintenance more difficult. An alternative to passive methods is active noise control (ANC). However, despite extensive research undertaken in the last three decades, there is still no viable commercial solution for the active control of transformer noise. The difficulties for a successful implementation of an ANC solution in the case of power transformer noise are investigated through simulations and tests with a prototype. The three main obstacles found were: the positioning of the electro-acoustic transducers; the achievement of sufficient attenuation in a region far from the transformer, using a small number of control sources and error sensors (when the latter are positioned on the region of acoustic near-field); and the robust identification of the secondary path in a low signal/noise situation. The two former problems were dealt with, and some alternative solutions were analyzed. Genetic algorithms (GA) were used for the optimization of the transducers\' position. The performance of these algorithms is strongly related to the acoustical model used and it was verified that the Usry method, used for modelling the transformers primary field, did not result in an adequate estimate. Using a simplified model for the primary source, the importance of the cost function in the GA\'s performance was made evident. It was also verified that the joint optimization of transducers\' position provides the same result, and in shorter time, as the independent optimization of control source and error sensor positions. Simulations with a new virtual sensor strategy (based on windowing the control sources) show that it is possible to increase attenuation levels in a region far from the transformer, even with a small number of control sources and error sensors. Laboratory and field tests with an ANC system prototype were undertaken and the results of these tests are thoroughly discussed.