30 resultados para Sound recordings in ethnomusicology

em Universidad Politécnica de Madrid


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The default mode network (DMN) has received growing attention in recent years because it seems to be involved in the neuropathology of psychiatric and neurodegenerative disorders such as autism, schizophrenia and Alzheimer Disease. It has been defined as a task negative network, beca use the activity of all its brain regions is increased during the resting state and suspended during external or goal directed tasks.

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Within the regression framework, we show how different levels of nonlinearity influence the instantaneous firing rate prediction of single neurons. Nonlinearity can be achieved in several ways. In particular, we can enrich the predictor set with basis expansions of the input variables (enlarging the number of inputs) or train a simple but different model for each area of the data domain. Spline-based models are popular within the first category. Kernel smoothing methods fall into the second category. Whereas the first choice is useful for globally characterizing complex functions, the second is very handy for temporal data and is able to include inner-state subject variations. Also, interactions among stimuli are considered. We compare state-of-the-art firing rate prediction methods with some more sophisticated spline-based nonlinear methods: multivariate adaptive regression splines and sparse additive models. We also study the impact of kernel smoothing. Finally, we explore the combination of various local models in an incremental learning procedure. Our goal is to demonstrate that appropriate nonlinearity treatment can greatly improve the results. We test our hypothesis on both synthetic data and real neuronal recordings in cat primary visual cortex, giving a plausible explanation of the results from a biological perspective.

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Knowledge of the uncertainty of measurement of testing results is important when results have to be compared with limits and specifications. In the measurement of sound insulation following standards UNE EN ISO 140-4 the uncertainty of the final magnitude is mainly associated to the average sound pressure levels L1 and L2 measured. A parameter that allows us to quantify the spatial variation of the sound pressure level is the standard deviation of the pressure levels measured at different points of the room. In this work, for a wide number of measurements following standards UNE EN ISO 140-4 we analyzed qualitatively the behaviour of the standard deviation for L1 and L2. The study of sound fields in enclosed spaces is very difficult. There are a wide variety of rooms with different sound fields depending on factors as volume, geometry and materials. In general, we observe that the L1 and L2 standard deviations contain peaks and dips independent on characteristics of the rooms at single frequencies that could correspond to critical frequencies of walls, floors and windows or even to temporal alterations of the sound field. Also, in most measurements according to UNE EN ISO 140-4 a large similitude between L1 and L2 standard deviation is found. We believe that such result points to a coupled system between source and receiving rooms, mainly at low frequencies the shape of the L1 and L2 standard deviations is comparable to the velocity level standard deviation on a wall

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The ability to reproduce reduced gravity conditions for long periods is one of the reasons why the orbiting laboratory is so attractive. In this paper several fluid dynamics problem areas are reviewed in which zero-gravity conditions are of great importance. Although emphasis is placed on space processing, there are some older problems also in which gravity masks the phenomcna, impeding a reasonably simple approach to the solution. Three problems are considered: Thermal convection under reduced gravity. The dumping effect ofsurface gravity waves at the outset of convection induced by surface tractions is discussed in particular. The existence of convection is of concern for some satellite thermal control techniques presently used, and for most of the proposed manufacturing processes. Whereas convection should be normally avoided, problems related to the containerless stirring ofa melt constitute an exception. Secondly, gravity and chemical reactions. Although chemical reactions are independent of gravity because of the small mass of the molecules and atoms involved, in many cases the reaction rate dcpends on the arrival of the species to the reaction zone. When the arrival process is buoyancy-controlled, the net specd of the reaction will be affected by the gravity. Thirdly, two-phase flows under reduced gravity provkle interesting problems from boiling heat transfer to degasslng of melts. This part of the paper deals only with the measurement of sound veiocity in a liquid containing bubbles. It is suggested that such measurements should be mude under reduced gravity to provide reliable residís.

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This paper presents a new verification procedure for sound source coverage according to ISO 140?5 requirements. The ISO 140?5 standard applies to the measurement of façade insulation and requires a sound source able to achieve a sufficiently uniform sound field in free field conditions on the façade under study. The proposed method involves the electroacoustic characterisation of the sound source in laboratory free field conditions (anechoic room) and the subsequent prediction by computer simulation of the sound free field radiated on a rectangular surface equal in size to the façade being measured. The loudspeaker is characterised in an anechoic room under laboratory controlled conditions, carefully measuring directivity, and then a computer model is designed to calculate the acoustic free field coverage for different loudspeaker positions and façade sizes. For each sound source position, the method provides the maximum direct acoustic level differences on a façade specimen and therefore determines whether the loudspeaker verifies the maximum allowed level difference of 5 dB (or 10 dB for façade dimensions greater than 5 m) required by the ISO standard. Additionally, the maximum horizontal dimension of the façade meeting the standard is calculated and provided for each sound source position, both with the 5 dB and 10 dB criteria. In the last section of the paper, the proposed procedure is compared with another method used by the authors in the past to achieve the same purpose: in situ outdoor measurements attempting to recreate free field conditions. From this comparison, it is concluded that the proposed method is able to reproduce the actual measurements with high accuracy, for example, the ground reflection effect, at least at low frequencies, which is difficult to avoid in the outdoor measurement method, and it is fully eliminated with the proposed method to achieve the free field requisite.

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In this work, we present a study whose objective is to prove the influence of background noise produced inside university facilities on the brain waves related to attention processes. Recordings of background noise were carried out in study areas inside university facilities. Volunteers were asked to perform an attention test without any background noise but also while being exposed to the sound recordings, and their cerebral activity was recorded through electroencephalography (EEG). After the application of the test in both conditions, changes in the frequency bands related to attention processes (beta 13-30 Hz and theta 4-7 Hz) were studied. The results of this study show that when the students were performing the test while being exposed to background noise, both beta and theta frequency bands decreased statistically significantly. Because attentional improvement is related to increases of the beta and theta waves, we believe that those decreases are directly related to a lack of attention caused by the exposure to background noise. Nevertheless, the results do not allow us to conclude that background noise produced inside university facilities has an influence on the attentional processes.

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Una de las aplicaciones más interesantes de las tecnologías de Realidad Acústica Virtual es la posibilidad de recuperar el patrimonio cultural del sonido de los recintos históricos que se han perdido o cuyas características se han modificado a lo largo de los siglos. En este trabajo, las tecnologías de realidad acústica virtual se utilizan para tratar de reconstruir virtualmente el sonido producido en las actividades litúrgicas del antiguo rito hispánico. Bajo este nombre se conoce a la liturgia que celebraban los cristianos de la península ibérica hasta su prohibición, en favor del culto romano, a mediados del siglo XI. El rito hispánico -también conocido como rito visigótico-mozárabe- es una de las manifestaciones culturales más interesantes de la Alta Edad Media en occidente. Se trata de un patrimonio cultural desaparecido, tanto desde el punto de vista de las señales sonoras que lo integraban -ya que la mayoría de las melodías que conformaban el rito se han perdido- como desde el punto de vista de los espacios en los que se desarrollaba, debido a que las iglesias conservadas de la época han experimentado modificaciones a lo largo de los siglos que alteran sus condiciones acústicas con respecto a las que tenían en el periodo de vigencia de esta liturgia. Para llevar a cabo este proyecto, se han realizado modelos acústicos digitales de un grupo representativo de iglesias prerrománicas de la Península Ibérica en su estado primitivo. Se ha procurado que las iglesias seleccionadas representen la variedad de comportamientos acústicos esperables en este tipo de edificios. Con este objetivo, se han elegido cinco iglesias prerrománicas que presentan diferencias sustanciales en los parámetros que, a priori, van a influir en mayor manera a su comportamiento acústico: el volumen del recinto, la forma de la planta y el tipo de cubierta. El proceso de creación de los modelos acústicos digitales de las iglesias se ha dividido en dos fases: en la primera se han creado modelos de los edificios en su estado actual, que se han validado a partir de los datos obtenidos en mediciones acústicas realizadas in situ; a partir de los modelos validados de las iglesias en su estado actual, en la segunda fase se han generado los modelos acústicos digitales correspondientes al estado primitivo de las mismas, modificando los modelos anteriores de acuerdo con las hipótesis de reconstrucción propuestas en las investigaciones arqueológicas más recientes. Se han realizado grabaciones en cámara anecoica de una serie de piezas del repertorio original del canto mozárabe. Las grabaciones se han llevado a cabo con un array esférico compuesto por 32 micrófonos, con objeto de obtener información sobre la directividad de la emisión sonora de los cantantes, que se aplicará en las auralizaciones. Finalmente, se han realizado diversas auralizaciones, teniendo en cuenta diferentes configuraciones litúrgicas que eran usuales en este rito. ABSTRACT One of the most interesting applications of the Acoustic Virtual Reality technologies is the possibility to recover the cultural heritage of the sound of the historical sites that have been lost or whose characteristics have been modified through time. In this work, Acoustic Virtual Reality technologies are used to try to reconstruct virtually the sound produced in the liturgical activities of the Hispanic Rite. This is the name given to the liturgy celebrated by Christians of the Iberian Peninsula prior to the introduction of the Roman cult in the mid-eleventh century. The Hispanic Rite (also known as Visigothic or Mozarabic rite) is one of the most interesting cultural manifestations of the Middle Ages. It is a lost cultural heritage, both from the point of view of the sound signals that were used since the majority of the melodies that conformed the rite have been lost, and from the point of view of the spaces in which this liturgy was celebrated, because the churches preserved from that era have changed over the centuries altering its acoustic conditions respect to which they were in the period in which this liturgy was in effect. To carry out this project, acoustic models of a representative group of the pre-Romanesque churches in the Iberian Peninsula have been made in his primitive state. To select the sample of buildings to study, efforts have been made to ensure that the selected churches were representative of the range of expected acoustic behaviors in this type of buildings. Five churches have been selected, showing differences in the parameters that have the greatest influence on their acoustic behavior: the enclosure volume, the shape of the floor plan and the type of roof. The process of creating digital acoustic models of the churches has been divided into two phases. In the first phase acoustic models of the churches in its current state have been created. These models have been validated with the data obtained from in situ acoustic measurements. From the validated models of churches in its current state, in the second phase, changes in the acoustic models have been performed in order to represent the primitive state of the churches, according to the historical reconstruction hypothesis that have been proposed in the most recent archaeological investigations. Anechoic recordings of a series of pieces of the original Mozarabic Chant repertoire have been recorded. The recordings were made with a spherical array composed of 32 microphones, in order to obtain information on the directivity of the sound emission of the singers. These directivity data will be used to simulate the directional radiation of the sound sources in the auralizations. Finally, auralizations were produced corresponding to different liturgical configurations which were common in this rite.

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Traditional Text-To-Speech (TTS) systems have been developed using especially-designed non-expressive scripted recordings. In order to develop a new generation of expressive TTS systems in the Simple4All project, real recordings from the media should be used for training new voices with a whole new range of speaking styles. However, for processing this more spontaneous material, the new systems must be able to deal with imperfect data (multi-speaker recordings, background and foreground music and noise), filtering out low-quality audio segments and creating mono-speaker clusters. In this paper we compare several architectures for combining speaker diarization and music and noise detection which improve the precision and overall quality of the segmentation.

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Noise maps are usually represented as contour or isolines maps describing the sound levels in a region. Using this kind of representation the user can easily find the noise level assigned to every location in the map. But the acoustic calculations behind the map are not performed for every single location on it; they are only performed in a grid of receivers. The results in this calculation grid are interpolated to draw the isolines or contours. Therefore, the resolution of the calculation grid and the way it was created (rectangular, triangulated, random…) have an effect on the resulting map. In this paper we describe a smart iterative procedure to optimize the quality of the map at a really low additional computational cost, using self-adaptive grids for the acoustic calculations. These self-adaptive grids add new receivers to the sampling grid in those locations where they are expected to be more useful, so that the performance at the output of the interpolator is enhanced. Self-adaptive sampling grids can be used for minimizing the overall error of the map (improving its quality), or for reducing calculation times, and can be also applied selectively to target areas or contour lines. This can be done by the user customizing the maximum number of iterations, the number of new receivers for each iteration, the target isolines, the target quality…

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Este trabajo tiene la vocación de dibujar el mapa de los dominios de oportunidad que hay a disposición de un arquitecto, al imaginar el aspecto sonoro de la ciudad en relación con la campana. Muchas de las dimensiones señaladas, tanto conocidas como desconocidas, son sencillamente despertadas al estudio de lo que un diseñador del espacio sonoro, como Llorenç Barber, viene a decirnos, desde una disciplina aparentemente contactada, pero muchas veces paralela a la arquitectónica. Tomando los conciertos de campanas celebrados hasta la fecha como puestas en práctica real de la condición instrumental de más de doscientas ciudades de todo el mundo, el estudio analiza los hallazgos, éxitos y fracasos, obtenidos a lo largo de estos últimos veinticinco años de experiencia, como resultados explícitos de un trabajo sonoro en la ciudad, probado, contrastado y afinado, del que extraer indicaciones concretas susceptibles de ser aplicadas en el diseño del espacio urbano. Mediante un análisis triangular de vértices definidos: emisor-campana, medio-ciudad, receptor-ciudadano; la investigación aborda los aspectos relativos a la producción y recepción del fenómeno sonoro generado por la campana en la ciudad. En relación con la producción, una parte del trabajo se dedica al estudio de los efectos acústicos observados, ordenado por escalas graduales en las que se produce una distribución del sonido susceptible de diseño: el vaso de la campana, la sala de campanas, el campanario, el cuerpo del edificio, la ciudad y el territorio. A modo de cajas de resonancia y leídas como muñecas rusas, unas dentro de otras, los espacios identificados muestran cualidades sonoras específicas, definidas tanto por sus condiciones geométricas, formales, constructivas o de uso; como por las correspondientes a las subsiguientes cajas que alojan. A fin de esclarecer la influencia de tales cuestiones en el aspecto sonoro de la ciudad, se propone un modelo ordenado de conexión y desconexión de escalas, utilizando una paramétrica puramente técnica creada ex profeso, junto con variables metodológicas más tradicionales. Al mismo tiempo, y tratando de esclarecer cómo, al ser puesta en vibración, la ciudad es aprehendida, disfrutada y rememorada por el ciudadano-oyente, otra parte del trabajo se dedica al estudio de los aspectos relativos a su recepción aural en deambulación compartida. En este caso la investigación se propone reclamar otras dimensiones más subversivas que, si bien escapan a los modos habituales de trabajar del arquitecto, se revelan intensamente en la experiencia plurifocal, multiplicando los efectos a considerar: efectos funcionales y significativos, de reconocimiento, integración y pertenencia a un cuerpo territorial y social de coordenadas históricas y geográficas de nuevo significadas; efectos perceptuales de inmersión, ubicuidad, temporalidad o inestabilidad; efectos estéticos, de rememoración, interpretación simbólica y recreación poética; e incluso efectos políticos, descubriendo un espacio urbano en continua regeneración, lugar para la exposición en su doble acepción, para la exhibición y el peligro, o como contenedor situacional del más profundo sentido ciudadano. Para afinar la relevancia de lo obtenido en cada una de las dimensiones señaladas, el trabajo se articula en tres aproximaciones graduales: el corpus general de los conciertos celebrados hasta la fecha; los conciertos celebrados en España; los tres conciertos para Madrid: Magna Mater (1991), Festi Clamores (2000) y Aurea Catena (2007). Si bien el modelo propuesto nace a la luz de los conciertos de campanas de Llorenç Barber, a escala de una ciudad entera y con una intención compositiva individual, se entiende que sería útil para el uso ordenado de cualquier profesional interesado en el aspecto sonoro de la ciudad, faceta escasamente atendida, dicho sea de paso, desde la disciplina arquitectónica. ABSTRACT This work has the vocation of drawing out the numerous opportunities an architect has at his disposal, upon imagining the sonorous aspect of the city in relation to the bell. Many of the dimensions indicated, both known and unknown, are just awakened to the study of what a sound space designer, as Llorenç Barber, comes to tell us, from a discipline apparently contacted, but often parallel to the architecture. Taking the bell concerts held so far as actual implementation of the instrumental condition of over two hundred cities around the world, the study analyzes the findings, successes and failures, obtained over the last twenty years of experience, as explicit results of a sound work in the city, tested, verified and refined, from which to extract specific indications that can be applied in the design of urban space. By triangular analyzing of defined vertices: sender-bell, half-city, receptor-citizen; the research addresses issues relating to the production and reception of sound phenomenon generated by the bell in the city. In relation to production, part of the work is devoted to the study of observed acoustic effects, ordered gradual scale which produces a distribution of sound capable of design: the glass of the bell, the bell room, the bell tower and the body of the building, the city and territory. By way of sounding boards and read as if they were Russian dolls, one inside the other, the show spaces identified specific sound qualities, defined both for their geometric, formal, constructive use, such as those for hosting the subsequent boxes. In order to clarify the influence of such issues in the sound aspect of the city, we propose an ordered pattern of connection and disconnection of scales, using a purely parametric technique created on purpose, along with more traditional methodological variables. At the same time, and trying to clarify how, when set in vibration, the city is apprehended, enjoyed and remembered for the citizen-listener, another part of the work is devoted to the study of aspects of aural reception in shared ambulation. In this case the research aims to claim more subversive than other dimensions, but beyond the usual ways in which an architect works, the experience reveals intensely plurifocal multiplying effects to consider: functional effects and significant recognition and integration belonging to a body of territorial and social historical and geographical coordinates of new meaning and perceptual effects of immersion, ubiquity, timeliness or instability; aesthetic effects of recall, interpretation and recreation of symbolic poetic; and even political effects, revealing a continuous urban space regeneration site for the exhibition in its double meaning, for display and danger, or as a citizen sense container. To sharpen the relevance of what was obtained in each of the dimensions mentioned, the work is divided into three incremental approaches: the general corpus of the concerts held so far, the concerts in Spain, the three concerts for Madrid: Magna Mater (1991), Festi Clamores (2000) and Aurea Catena (2007). While the proposed model comes in the light of the bells concert Llorenç Barber, the scale of a whole city and individual compositional intent, it is understood that it would be useful for the orderly use of any professional interested in the sound aspect of the city, an aspect sparsely attended, incidentally, from the architectural discipline.

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Este proyecto de fin de carrera tiene como objetivo obtener una visión detallada de los sistemas y tecnologías de grabación y reproducción utilizadas para aplicaciones de audio 3D y entornos de realidad virtual, analizando las diferentes alternativas existentes, su funcionamiento, características, detalles técnicos y sus ámbitos de aplicación. Como punto de partida se estudiará la teoría psicoacústica y la localización de fuentes sonoras en el espacio, base para el estudio de los sistemas de audio 3D. Se estudiará tanto la espacialización sonora en un espacio real y la espacialización virtual (simulación mediante procesado de información de la localización de fuentes sonoras), en los que intervienen algunos fenómenos acústicos y psicoacústicos como ITD, o diferencia de tiempo que existe entre una señal acústica que llega a los pabellones auditivos, la ILD, o diferencia de intensidad o amplitud que hay entre la señal que llega a los pabellones auditivos y la localización espacial mediante otra serie de mecanismos biaurales. Tras una visión general de la teoría psicoacústica y la espacialización sonora, se analizarán con detalle los elementos de grabación y reproducción existentes para audio 3D. Concretamente, a lo largo del proyecto se profundizará en el funcionamiento del sistema estéreo, caracterizado por el posicionamiento sonoro mediante la utilización de dos canales; del sistema biaural, caracterizado por reconstruir campos sonoros mediante el uso de las HRTF; de los sistemas multicanal, detallando gran parte de las alternativas y configuraciones existentes; del sistema Ambiophonics, caracterizado por implementar filtros de cruce; del sistema Ambisonics, y sus diferentes formatos y técnicas de codificación y decodificación; y del sistema Wavefield Synthesis, caracterizado por recrear ambientes sonoros en grandes espacios. ABSTRACT This project aims to get a detailed view of recording and reproducing systems and technologies used to 3D audio applications and virtual reality environments, analyzing the different alternatives available, their functioning, features, technical details and their different scopes of applications. As a starting point, will be studied the psychoacoustic theory and the localization of sound sources in space, basis for the 3D audio study. Will be studied both the spacialization of sound sources in real space as virtual spatialization of sound sources (simulation by information processing of localization of sound sources), in which involves some acoustic and psychoacoustic phenomena like ITD (or the Interaural time difference), the ILD, (or the Interaural Level Difference) and spatial localization by another set of binaural mechanisms. After a general overview of the psychoacoustics theory and the sound spatialization, will be analyzed in detail existing methods of recording and reproducing for 3D audio. Specifically, during the project will analyze the characteristics of the stereo systems, characterized by sound positioning using two channels; the binaural systems, characterized by reconstructing sound fields by using the HRTF; the multichannel systems, detailing many of the existing alternatives and configurations; the Ambiophonics system, which is characterized by implementing crosstalk elimination techniques; the Ambiosonics system, and its various formats and encoding and decoding techniques; and the Wavefield Synthesis system, characterized by recreate soundscapes in large spaces.

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El objetivo de este trabajo es la elaboración de un procedimiento para la medida del coeficiente de absorción sonora normal en un tubo de impedancia. Para ello se han estudiado los fundamentos básicos de la ecuación de ondas y sus soluciones. Se han considerado las soluciones pertinentes que describen el comportamiento de una onda sonora dentro de un tubo rígido. Se ha considerado también la teoría básica de funciones de transferencia. Estas teorías son claves a la hora de poder desarrollar el procedimiento de medida, ya que el coeficiente de absorción acústica se obtendrá con la ayuda de un tubo de impedancias que mide las funciones de transferencia entre dos posiciones de micrófonos incorporados en una de las caras del tubo. La utilización de esta técnica tiene como principal ventaja, la necesidad de poco espacio en un laboratorio y el empleo de muestras pequeñas de material. La implementación de los visto teóricamente a su aplicación práctica se ha hecho a través de un procedimiento de medida que sigue la Norma UNE-EN ISO 10534-2 (2002) “Determinación del coeficiente de absorción sonoro y la impedancia en tubos de impedancia Parte 2: método función de transferencia”. El valor del coeficiente de absorción se puede obtener a través de una instrumentación específica y un programa computador. Para poder validar los cálculos que realiza el programa utilizado, se ha realizado una batería de medidas del coeficiente de absorción a diferentes tipos de materiales acústicos, y los cálculos se han hecho por la vía del programa y por la vía de una hoja de cálculo. Como parte del procedimiento de medida se ha calculado la incertidumbre en las medidas. En definitiva se pretende contribuir con este trabajo a establecer un procedimiento de medida del comportamiento acústico de diversos materiales. SUMMARY. The aim of this work is the development of a procedure for measuring the sound absorption coefficient normal of an impedance tube. To this end we have studied the basics of the wave equation and its solutions. We have considered the relevant solutions that describe the behavior of a sound wave in a rigid tube. It has also considered the basic theory of transfer functions. These theories are key when we want to develop the measurement method, since the absorption coefficient is obtained with the aid of an impedance tube measuring transfer functions between two positions of microphones incorporated into one side of the tube. The use of this technique has the main advantage, the need of little space on a laboratory and use of small samples of material. The implementation of theoretically seen to his practical application has been made through a measurement procedure following the UNE-EN ISO 10534-2 (2002) "Determination of sound absorption coefficient and impedance in impedance tubes Part 2 : transfer function method ". The value of the absorption coefficient can be obtained through a specific instrumentation and computer software. In order to validate the calculations performed by the program used, there has been realized a series of measures of the absorption coefficient at different types of acoustical materials, and calculations were made by means of the program and by means of a spreadsheet. As part of the measurement procedure has been estimated uncertainty in the measurements. Ultimately it’s tried to contribute with this work to establish a procedure measuring the acoustic behavior of various materials.

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En primer lugar se realiza una breve introducción a la historia del refuerzo sonoro, en el cual veremos cómo a ido evolucionando hasta lo que se conoce hoy en día como Line Array, luego nos centraremos en los factores acústicos a tener en cuenta, en ellos repasaremos los conceptos de fase y la importancia de este factor en cuanto a la interacción de más de altavoz, en esta parte también repasaremos como afectan las condiciones climáticas como la atenuación del aire y la temperatura, a la propagación de sonido. A continuación nos centraremos en el diseño de arreglos de altavoces, veremos las diferentes configuraciones para diferentes rangos de trabajo, veremos sus ventajas y desventajas de cada arreglo y también se verá la forma de controlar la directividad de los arreglos para optimizar la propagación del sonido en el área a cubrir, para terminar esta parte profundizaremos en el diseño de los sistemas Line Array, analizando su estructura interna para entender su comportamiento directivo y eficaz en cuanto a la propagación de ondas. Por último se hará el análisis de un montaje real, en el cual tuve participación directa en el montaje ya que la empresa para la que trabajaba se encargó de hacer la gira del grupo español Amaral. Esta gira se realiza en el año 2008-2009, gracias a esta experiencia he podido llevar a cabo este proyecto donde también he podido comprobar algunos conceptos empleados en el diseño de arreglos. De esta Gira, se analizará la efectividad y el rendimiento del diseño de arreglo empleado, para esto se generará mediante software de predicción acústica, el mapa de presión sonora generado por el diseño empleado, una vez visto los resultados, se planteará una reorganización del arreglo de altavoces, para poder conseguir un mejor rendimiento en el área a cubrir. ABSTRACT. First is a brief introduction to the history of sound reinforcement, in which we will see how to have evolved into what is known today as Line Array, then we will focus on acoustic factors to consider in they will review phase concepts and the importance of this factor as to the interaction of the speaker more in this part also review such as climatic conditions affecting air attenuation and temperature, to the propagation of sound. Here we focus on the speaker array design, we see the different configurations for different ranges of work, we will see the advantages and disadvantages of each arrangement and also see how to control the directivity of the arrays to optimize sound propagation in the area to be covered, to finish this part will delve into the design of line array systems, analyzing its internal structure to understand its behavior management and effective in terms of wave propagation. Finally, we will analyze a real assembly, which had direct involvement in the assembly as the company for which he worked was commissioned to do the tour of the Spanish group Amaral. This tour takes place in the year 2008-2009, thanks to this experience I have been able to conduct this project where I have seen also some concepts used in the array design. In this tour, we analyze the effectiveness and performance of the array design used for this is generated by acoustical prediction software, the map of sound pressure generated by the design employed, once seen the results, he will consider reorganization under speaker, in order to achieve better performance in the area to be covered.

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El actual proyecto consiste en la creación de una interfaz gráfica de usuario (GUI) en entorno de MATLAB que realice una representación gráfica de la base de datos de HRTF (Head-Related Transfer Function). La función de transferencia de la cabeza es una herramienta muy útil en el estudio de la capacidad del ser humano para percibir su entorno sonoro, además de la habilidad de éste en la localización de fuentes sonoras en el espacio que le rodea. La HRTF biaural (terminología para referirse al conjunto de HRTF del oído izquierdo y del oído derecho) en sí misma, posee información de especial interés ya que las diferencias entre las HRTF de cada oído, conceden la información que nuestro sistema de audición utiliza en la percepción del campo sonoro. Por ello, la funcionalidad de la interfaz gráfica creada presenta gran provecho dentro del estudio de este campo. Las diferencias interaurales se caracterizan en amplitud y en tiempo, variando en función de la frecuencia. Mediante la transformada inversa de Fourier de la señal HRTF, se obtiene la repuesta al impulso de la cabeza, es decir, la HRIR (Head-Related Impulse Response). La cual, además de tener una gran utilidad en la creación de software o dispositivos de generación de sonido envolvente, se utiliza para obtener las diferencias ITD (Interaural Time Difference) e ILD (Interaural Time Difference), comúnmente denominados “parámetros de localización espacial”. La base de datos de HRTF contiene la información biaural de diferentes puntos de ubicación de la fuente sonora, formando una red de coordenadas esféricas que envuelve la cabeza del sujeto. Dicha red, según las medidas realizadas en la cámara anecoica de la EUITT (Escuela Universitaria de Ingeniería Técnica de Telecomunicación), presenta una precisión en elevación de 10º y en azimut de 5º. Los receptores son dos micrófonos alojados en el maniquí acústico llamado HATS (Hats and Torso Simulator) modelo 4100D de Brüel&Kjaer. Éste posee las características físicas que influyen en la percepción del entorno como son las formas del pabellón auditivo (pinna), de la cabeza, del cuello y del torso humano. Será necesario realizar los cálculos de interpolación para todos aquellos puntos no contenidos en la base de datos HRTF, este proceso es sumamente importante no solo para potenciar la capacidad de la misma sino por su utilidad para la comparación entre otras bases de datos existentes en el estudio de este ámbito. La interfaz gráfica de usuario está concebida para un manejo sencillo, claro y predecible, a la vez que interactivo. Desde el primer boceto del programa se ha tenido clara su filosofía, impuesta por las necesidades de un usuario que busca una herramienta práctica y de manejo intuitivo. Su diseño de una sola ventana reúne tanto los componentes de obtención de datos como los que hacen posible la representación gráfica de las HRTF, las HRIR y los parámetros de localización espacial, ITD e ILD. El usuario podrá ir alternando las representaciones gráficas a la vez que introduce las coordenadas de los puntos que desea visualizar, definidas por phi (elevación) y theta (azimut). Esta faceta de la interfaz es la que le otorga una gran facilidad de acceso y lectura de la información representada en ella. Además, el usuario puede introducir valores incluidos en la base de datos o valores intermedios a estos, de esta manera, se indica a la interfaz la necesidad de realizar la interpolación de los mismos. El método de interpolación escogido es el de la ponderación de la distancia inversa entre puntos. Dependiendo de los valores introducidos por el usuario se realizará una interpolación de dos o cuatro puntos, siendo éstos limítrofes al valor introducido, ya sea de phi o theta. Para añadir versatilidad a la interfaz gráfica de usuario, se ha añadido la opción de generar archivos de salida en forma de imagen de las gráficas representadas, de tal forma que el usuario pueda extraer los datos que le interese para cualquier valor de phi y theta. Se completa el presente proyecto fin de carrera con un trabajo de investigación y estudio comparativo de la función y la aplicación de las bases de datos de HRTF dentro del marco científico y de investigación. Esto ha hecho posible concentrar información relacionada a través de revistas científicas de investigación como la JAES (Journal of the Audio Engineering Society) o la ASA (Acoustical Society of America), además, del IEEE ( Institute of Electrical and Electronics Engineers) o la “Web of knowledge” entre otras. Además de realizar la búsqueda en estas fuentes, se ha optado por vías de información más comunes como Google Académico o el portal de acceso “Ingenio” a los todos los recursos electrónicos contenidos en la base de datos de la universidad. El estudio genera una ampliación en el conocimiento de la labor práctica de las HRTF. La mayoría de los estudios enfocan sus esfuerzos en mejorar la percepción del evento sonoro mediante su simulación en la escucha estéreo o multicanal. A partir de las HRTF, esto es posible mediante el análisis y el cálculo de datos como pueden ser las regresiones, siendo éstas muy útiles en la predicción de una medida basándose en la información de la actual. Otro campo de especial interés es el de la generación de sonido 3D. Mediante la base de datos HRTF es posible la simulación de una señal biaural. Se han diseñado algoritmos que son implementados en dispositivos DSP, de tal manera que por medio de retardos interaurales y de diferencias espectrales es posible llegar a un resultado óptimo de sonido envolvente, sin olvidar la importancia de los efectos de reverberación para conseguir un efecto creíble de sonido envolvente. Debido a la complejidad computacional que esto requiere, gran parte de los estudios coinciden en desarrollar sistemas más eficientes, llegando a objetivos tales como la generación de sonido 3D en tiempo real. ABSTRACT. This project involves the creation of a Graphic User Interface (GUI) in the Matlab environment which creates a graphic representation of the HRTF (Head-Related Transfer Function) database. The head transfer function is a very useful tool in the study of the capacity of human beings to perceive their sound environment, as well as their ability to localise sound sources in the area surrounding them. The binaural HRTF (terminology which refers to the HRTF group of the left and right ear) in itself possesses information of special interest seeing that the differences between the HRTF of each ear admits the information that our system of hearing uses in the perception of each sound field. For this reason, the functionality of the graphic interface created presents great benefits within the study of this field. The interaural differences are characterised in space and in time, varying depending on the frequency. By means of Fourier's transformed inverse of the HRTF signal, the response to the head impulse is obtained, in other words, the HRIR (Head-Related Impulse Response). This, as well as having a great use in the creation of software or surround sound generating devices, is used to obtain ITD differences (Interaural Time Difference) and ILD (Interaural Time Difference), commonly named “spatial localisation parameters”. The HRTF database contains the binaural information of different points of sound source location, forming a network of spherical coordinates which surround the subject's head. This network, according to the measures carried out in the anechoic chamber at the EUITT (School of Telecommunications Engineering) gives a precision in elevation of 10º and in azimuth of 5º. The receivers are two microphones placed on the acoustic mannequin called HATS (Hats and Torso Simulator) Brüel&Kjaer model 4100D. This has the physical characteristics which affect the perception of the surroundings which are the forms of the auricle (pinna), the head, neck and human torso. It will be necessary to make interpolation calculations for all those points which are not contained the HRTF database. This process is extremely important not only to strengthen the database's capacity but also for its usefulness in making comparisons with other databases that exist in the study of this field. The graphic user interface is conceived for a simple, clear and predictable use which is also interactive. Since the first outline of the program, its philosophy has been clear, based on the needs of a user who requires a practical tool with an intuitive use. Its design with only one window unites not only the components which obtain data but also those which make the graphic representation of the HRTFs possible, the hrir and the ITD and ILD spatial location parameters. The user will be able to alternate the graphic representations at the same time as entering the point coordinates that they wish to display, defined by phi (elevation) and theta (azimuth). The facet of the interface is what provides the great ease of access and reading of the information displayed on it. In addition, the user can enter values included in the database or values which are intermediate to these. It is, likewise, indicated to the interface the need to carry out the interpolation of these values. The interpolation method is the deliberation of the inverse distance between points. Depending on the values entered by the user, an interpolation of two or four points will be carried out, with these being adjacent to the entered value, whether that is phi or theta. To add versatility to the graphic user interface, the option of generating output files in the form of an image of the graphics displayed has been added. This is so that the user may extract the information that interests them for any phi and theta value. This final project is completed with a research and comparative study essay on the function and application of HRTF databases within the scientific and research framework. It has been possible to collate related information by means of scientific research magazines such as the JAES (Journal of the Audio Engineering Society), the ASA (Acoustical Society of America) as well as the IEEE (Institute of Electrical and Electronics Engineers) and the “Web of knowledge” amongst others. In addition to carrying out research with these sources, I also opted to use more common sources of information such as Academic Google and the “Ingenio” point of entry to all the electronic resources contained on the university databases. The study generates an expansion in the knowledge of the practical work of the HRTF. The majority of studies focus their efforts on improving the perception of the sound event by means of its simulation in stereo or multichannel listening. With the HRTFs, this is possible by means of analysis and calculation of data as can be the regressions. These are very useful in the prediction of a measure being based on the current information. Another field of special interest is that of the generation of 3D sound. Through HRTF databases it is possible to simulate the binaural signal. Algorithms have been designed which are implemented in DSP devices, in such a way that by means of interaural delays and wavelength differences it is possible to achieve an excellent result of surround sound, without forgetting the importance of the effects of reverberation to achieve a believable effect of surround sound. Due to the computational complexity that this requires, a great many studies agree on the development of more efficient systems which achieve objectives such as the generation of 3D sound in real time.

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La medida de la presión sonora es un proceso de extrema importancia para la ingeniería acústica, de aplicación en numerosas áreas de esta disciplina, como la acústica arquitectónica o el control de ruido. Sobre todo en esta última, es necesario poder efectuar medidas precisas en condiciones muy diversas. Por otra parte, la ubicuidad de los dispositivos móviles inteligentes (smartphones, tabletas, etc.), dispositivos que integran potencia de procesado, conectividad, interactividad y una interfaz intuitiva en un tamaño reducido, abre la posibilidad de su uso como sistemas de medida de calidad y de coste bajo. En este Proyecto se pretende utilizar las capacidades de entrada y salida, procesado, conectividad inalámbrica y geolocalización de los dispositivos móviles basados en iOS, en concreto el iPhone, para implementar un sistema de medidas acústicas que iguale o supere las prestaciones de los sonómetros existentes en el mercado. SonoPhone permitirá, mediante la conexión de un micrófono de medida adecuado, la realización de medidas de acuerdo a las normas técnicas en vigor, así como la posibilidad de programar, configurar y almacenar o trasmitir las medidas realizadas, que además estarán geolocalizadas con el GPS integrado en el dispositivo móvil. También se permitirá enviar los datos de la medida a un almacenamiento remoto en la nube. La aplicación tiene una estructura modular en la que un módulo de adquisición de datos lee la señal del micrófono, un back-end efectúa el procesado necesario, y otros módulos permiten la calibración del dispositivo y programar y configurar las medidas, así como su almacenamiento y transmisión en red. Una interfaz de usuario (GUI) permite visualizar las medidas y efectuar las configuraciones deseadas por el usuario, todo ello en tiempo real. Además de implementar la aplicación, se ha realizado una prueba de funcionamiento para determinar si el hardware del iPhone es adecuado para la medida de la presión acústica de acuerdo a las normas internacionales. Sound pressure measurement is an extremely important process in the field of acoustic engineering, with applications in numerous subfields, like for instance building acoustics and noise control, where it is necessary to be able to accurately measure sound pressure in very diverse (and sometimes adverse) conditions. On the other hand, the growing ubiquity of mobile devices such as smartphones or tablets, which combine processing power, connectivity, interactivity and an intuitive interface in a small size, makes it possible to use these devices as quality low-cost measurement systems. This Project aims to use the input-output capabilities of iOS-based mobile devices, in particular the iPhone, together with their processing power, wireless connectivity and geolocation features, to implement an acoustic measurement system that rivals the performance of existing devices. SonoPhone allows, with the addition of an adequate measurement microphone, to carry out measurements that comply with current technical regulations, as well as programming, configuring, storing and transmitting the results of the measurement. These measurements will be geolocated using the integrated GPS, and can be transmitted effortlessly to a remote cloud storage. The application is structured in modular fashion. A data acquisition module reads the signal from the microphone, while a back-end module carries out the necessary processing. Other modules permit the device to be calibrated, or control the configuration of the measurement and its storage or transmission. A Graphical User Interface (GUI) allows visual feedback on the measurement in progress, and provides the user with real-time control over the measurement parameters. Not only an application has been developed; a laboratory test was carried out with the goal of determining if the hardware of the iPhone permits the whole system to comply with international regulations regarding sound level meters.