Temporally flat top pulse generation from gain switched semiconductor lasers based on a polarization interferometer with variable transfer function

We propose the use of a polarization based interferometer with variable transfer function for the generation of temporally flat top pulses from gain switched single mode semiconductor lasers. The main advantage of the presented technique is its flexibility in terms of input pulse characteristics, as pulse duration, spectral bandwidth and operating wavelength. Theoretical predictions and experimental demonstrations are presented and the proposed technique is applied to two different semiconductor laser sources emitting in the 1550 nm region. Flat top pulses are successfully obtained with input seed pulses with duration ranging from 40 ps to 100 ps.

Diseño de una interfaz de representación gráfica de la base de datos de HRTF

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.

Wavelength monitoring with semiconductor laser amplifiers

The semiconductor laser diodes that are typically used in applications of optical communications, when working as amplifiers, present under certain conditions optical bistability, which is characterized by abruptly switching between two different output states and an associated hysteresis cycle. This bistable behavior is strongly dependent on the frequency detuning between the frequency of the external optical signal that is injected into the semiconductor laser amplifier and its own emission frequency. This means that small changes in the wavelength of an optical signal applied to a laser amplifier causes relevant changes in the characteristics of its transfer function in terms of the power requirements to achieve bistability and the width of the hysteresis. This strong dependence in the working characteristics of semiconductor laser amplifiers on frequency detuning suggest the use of this kind of devices in optical sensing applications for optical communications, such as the detection of shifts in the emission wavelength of a laser, or detect possible interference between adjacent channels in DWDM (Dense Wavelength Division Multiplexing) optical communication networks

Characteristic study, its identification and self-tuned approach to control hydro-power plants

The water time constant and mechanical time constant greatly influences the power and speed oscillations of hydro-turbine-generator unit. This paper discusses the turbine power transients in response to different nature and changes in the gate position. The work presented here analyses the characteristics of hydraulic system with an emphasis on changes in the above time constants. The simulation study is based on mathematical first-, second-, third- and fourth-order transfer function models. The study is further extended to identify discrete time-domain models and their characteristic representation without noise and with noise content of 10 & 20 dB signal-to-noise ratio (SNR). The use of self-tuned control approach in minimising the speed deviation under plant parameter changes and disturbances is also discussed.

Algoritmos genéticos en control activo de ruido

Este proyecto se centra en la implementación de un sistema de control activo de ruido mediante algoritmos genéticos. Para ello, se ha tenido en cuenta el tipo de ruido que se quiere cancelar y el diseño del controlador, parte fundamental del sistema de control. El control activo de ruido sólo es eficaz a bajas frecuencias, hasta los 250 Hz, justo para las cuales los elementos pasivos pierden efectividad, y en zonas o recintos de pequeñas dimensiones y conductos. El controlador ha de ser capaz de seguir todas las posibles variaciones del campo acústico que puedan producirse (variaciones de fase, de frecuencia, de amplitud, de funciones de transferencia electro-acústicas, etc.). Su funcionamiento está basado en algoritmos FIR e IIR adaptativos. La elección de un tipo de filtro u otro depende de características tales como linealidad, causalidad y número de coeficientes. Para que la función de transferencia del controlador siga las variaciones que surgen en el entorno acústico de cancelación, tiene que ir variando el valor de los coeficientes del filtro mediante un algoritmo adaptativo. En este proyecto se emplea como algoritmo adaptativo un algoritmo genético, basado en la selección biológica, es decir, simulando el comportamiento evolutivo de los sistemas biológicos. Las simulaciones se han realizado con dos tipos de señales: ruido de carácter aleatorio (banda ancha) y ruido periódico (banda estrecha). En la parte final del proyecto se muestran los resultados obtenidos y las conclusiones al respecto. Summary. This project is focused on the implementation of an active noise control system using genetic algorithms. For that, it has been taken into account the noise type wanted to be canceled and the controller design, a key part of the control system. The active noise control is only effective at low frequencies, up to 250 Hz, for which the passive elements lose effectiveness, and in small areas or enclosures and ducts. The controller must be able to follow all the possible variations of the acoustic field that might be produced (phase, frequency, amplitude, electro-acoustic transfer functions, etc.). It is based on adaptive FIR and IIR algorithms. The choice of a kind of filter or another depends on characteristics like linearity, causality and number of coefficients. Moreover, the transfer function of the controller has to be changing filter coefficients value thought an adaptive algorithm. In this project a genetic algorithm is used as adaptive algorithm, based on biological selection, simulating the evolutionary behavior of biological systems. The simulations have been implemented with two signal types: random noise (broadband) and periodic noise (narrowband). In the final part of the project the results and conclusions are shown.

On sonic anemometer measurement theory

In this paper a model for the measuring process of sonic anemometers (ultrasound pulse based) is presented. The differential equations that describe the travel of ultrasound pulses are solved in the general case of non-steady, non-uniform atmospheric flow field. The concepts of instantaneous line-average and travelling pulse-referenced average are established and employed to explain and calculate the differences between the measured turbulent speed (travelling pulse-referenced average) and the line-averaged one. The limit k1l=1 established by Kaimal in 1968, as the maximum value which permits the neglect of the influence of the sonic measuring process on the measurement of turbulent components is reviewed here. Three particular measurement cases are analysed: A non-steady, uniform flow speed field, a steady, non-uniform flow speed field and finally an atmospheric flow speed field. In the first case, for a harmonic time-dependent flow field, Mach number, M (flow speed to sound speed ratio) and time delay between pulses have revealed themselves to be important parameters in the behaviour of sonic anemometers, within the range of operation. The second case demonstrates how the spatial non-uniformity of the flow speed field leads to an influence of the finite transit time of the pulses (M≠0) even in the absence of non-steady behaviour of the wind speed. In the last case, a model of the influence of the sonic anemometer processes on the measurement of wind speed spectral characteristics is presented. The new solution is compared to the line-averaging models existing in the literature. Mach number and time delay significantly distort the measurement in the normal operational range. Classical line averaging solutions are recovered when Mach number and time delay between pulses go to zero in the new proposed model. The results obtained from the mathematical model have been applied to the calculation of errors in different configurations of practical interest, such as an anemometer located on a meteorological mast and the transfer function of a sensor in an atmospheric wind. The expressions obtained can be also applied to determine the quality requirements of the flow in a wind tunnel used for ultrasonic anemometer calibrations.

Adaptive control system based on lineal control theory for the path-following problem of a car-like mobile robot

The objective of this paper is to design a path following control system for a car-like mobile robot using classical linear control techniques, so that it adapts on-line to varying conditions during the trajectory following task. The main advantages of the proposed control structure is that well known linear control theory can be applied in calculating the PID controllers to full control requirements, while at the same time it is exible to be applied in non-linear changing conditions of the path following task. For this purpose the Frenet frame kinematic model of the robot is linearised at a varying working point that is calculated as a function of the actual velocity, the path curvature and kinematic parameters of the robot, yielding a transfer function that varies during the trajectory. The proposed controller is formed by a combination of an adaptive PID and a feed-forward controller, which varies accordingly with the working conditions and compensates the non-linearity of the system. The good features and exibility of the proposed control structure have been demonstrated through realistic simulations that include both kinematics and dynamics of the car-like robot.

Modelling of Topographic Irregularities for Seismic Site Response

Seismic evaluation methodology is applied to an existing viaduct in the south of Spain, near Granada, which is a medium seismicity region. The influence of both geology and topography in the spatial variability of ground motion are studied as well as seismic hazard analysis and ground motion characterization. Artificial hazard-consistent ground motion records are synthesised applying seismic hazard analysis and site effects are estimated through a diffraction study. Direct BEM is used to calculate the valley displacement response to vertically propagating SV waves and transfer functions are generated allowing the transformation of free field motion to motion at each support. A closed formulae is used to estimate these transfer function. Finally, the results obtained are compared.

Analysis of a thermal system through remote laboratories

This paper describes the experiences using remote laboratories for thorough analysis of a thermal system, including disturbances. Remote laboratories for education in subjects of control, is a common resorted method, used by universities. This method is applied to offer a flexible service in schedules so as to obtain greater and better results of available resources. Remote laboratories have been used for controlling physical devices remotely. Furthermore, remote labs have been used for transfer function identification of real equipment. Nevertheless, remote analyses of disturbances have not been done. The aim of this contribution is thereby to apply the experience of remote laboratories in the study of disturbances. Some experiments are carried out to demonstrate the effectiveness in using remote laboratories for complete analysis of a thermal system. Considering the remote access to thermal system, “Sistema de Laboratorios a Distancia” (SLD) was used.

Realización de un procedimiento para la medida del coeficiente de absorción sonora normal

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.

Contribución al desarrollo de técnicas CAD para el diseño de antenas impresas y dispositivos pasivos de microondas basadas en el método de elementos finitos

El objetivo principal de esta tesis es el desarrollo de herramientas numéricas basadas en técnicas de onda completa para el diseño asistido por ordenador (Computer-Aided Design,‘CAD’) de dispositivos de microondas. En este contexto, se desarrolla una herramienta numérica basada en el método de los elementos finitos para el diseño y análisis de antenas impresas mediante algoritmos de optimización. Esta técnica consiste en dividir el análisis de una antena en dos partes. Una parte de análisis 3D que se realiza sólo una vez en cada punto de frecuencia de la banda de funcionamiento donde se sustituye una superficie que contiene la metalización del parche por puertas artificiales. En una segunda parte se inserta entre las puertas artificiales en la estructura 3D la superficie soportando una metalización y se procede un análisis 2D para caracterizar el comportamiento de la antena. La técnica propuesta en esta tesis se puede implementar en un algoritmo de optimización para definir el perfil de la antena que permite conseguir los objetivos del diseño. Se valida experimentalmente dicha técnica empleándola en el diseño de antenas impresas de banda ancha para diferentes aplicaciones mediante la optimización del perfil de los parches. También, se desarrolla en esta tesis un procedimiento basado en el método de descomposición de dominio y el método de los elementos finitos para el diseño de dispositivos pasivos de microonda. Se utiliza este procedimiento en particular para el diseño y sintonía de filtros de microondas. En la primera etapa de su aplicación se divide la estructura que se quiere analizar en subdominios aplicando el método de descomposición de dominio, este proceso permite analizar cada segmento por separado utilizando el método de análisis adecuado dado que suele haber subdominios que se pueden analizar mediante métodos analíticos por lo que el tiempo de análisis es más reducido. Se utilizan métodos numéricos para analizar los subdominios que no se pueden analizar mediante métodos analíticos. En esta tesis, se utiliza el método de los elementos finitos para llevar a cabo el análisis. Además de la descomposición de dominio, se aplica un proceso de barrido en frecuencia para reducir los tiempos del análisis. Como método de orden reducido se utiliza la técnica de bases reducidas. Se ha utilizado este procedimiento para diseñar y sintonizar varios ejemplos de filtros con el fin de comprobar la validez de dicho procedimiento. Los resultados obtenidos demuestran la utilidad de este procedimiento y confirman su rigurosidad, precisión y eficiencia en el diseño de filtros de microondas. ABSTRACT The main objective of this thesis is the development of numerical tools based on full-wave techniques for computer-aided design ‘CAD’ of microwave devices. In this context, a numerical technique based on the finite element method ‘FEM’ for the design and analysis of printed antennas using optimization algorithms has been developed. The proposed technique consists in dividing the analysis of the antenna in two stages. In the first stage, the regions of the antenna which do not need to be modified during the CAD process are initially characterized only once from their corresponding matrix transfer function (Generalized Admittance matrix, ‘GAM’). The regions which will be modified are defined as artificial ports, precisely the regions which will contain the conducting surfaces of the printed antenna. In a second stage, the contour shape of the conducting surfaces of the printed antenna is iteratively modified in order to achieve a desired electromagnetic performance of the antenna. In this way, a new GAM of the radiating device which takes into account each printed antenna shape is computed after each iteration. The proposed technique can be implemented with a genetic algorithm to achieve the design objectives. This technique is validated experimentally and applied to the design of wideband printed antennas for different applications by optimizing the shape of the radiating device. In addition, a procedure based on the domain decomposition method and the finite element method has been developed for the design of microwave passive devices. In particular, this procedure can be applied to the design and tune of microwave filters. In the first stage of its implementation, the structure to be analyzed is divided into subdomains using the domain decomposition method; this process allows each subdomains can be analyzed separately using suitable analysis method, since there is usually subdomains that can be analyzed by analytical methods so that the time of analysis is reduced. For analyzing the subdomains that cannot be analyzed by analytical methods, we use the numerical methods. In this thesis, the FEM is used to carry out the analysis. Furthermore the decomposition of the domain, a frequency sweep process is applied to reduce analysis times. The reduced order model as the reduced basis technique is used in this procedure. This procedure is applied to the design and tune of several examples of microwave filters in order to check its validity. The obtained results allow concluding the usefulness of this procedure and confirming their thoroughness, accuracy and efficiency for the design of microwave filters.

3-dimensional binaural sound rendering using acoustic measurements stored in the SOFA format

Several groups all over the world are researching in several ways to render 3D sounds. One way to achieve this is to use Head Related Transfer Functions (HRTFs). These measurements contain the Frequency Response of the human head and torso for each angle. Some years ago, was only possible to measure these Frequency Responses only in the horizontal plane. Nowadays, several improvements have made possible to measure and use 3D data for this purpose. The problem was that the groups didn't have a standard format file to store the data. That was a problem when a third part wanted to use some different HRTFs for 3D audio rendering. Every of them have different ways to store the data. The Spatially Oriented Format for Acoustics or SOFA was created to provide a solution to this problem. It is a format definition to unify all the previous different ways of storing any kind of acoustics data. At the moment of this project they have defined some basis for the format and some recommendations to store HRTFs. It is actually under development, so several changes could come. The SOFA[1] file format uses a numeric container called netCDF[2], specifically the Enhaced data model described in netCDF 4 that is based on HDF5[3]. The SoundScape Renderer (SSR) is a tool for real-time spatial audio reproduction providing a variety of rendering algorithms. The SSR was developed at the Quality and Usability Lab at TU Berlin and is now further developed at the Institut für Nachrichtentechnik at Universität Rostock [4]. This project is intended to be an introduction to the use of SOFA files, providing a C++ API to manipulate them and adapt the binaural renderer of the SSR for working with the SOFA format. RESUMEN. El SSR (SoundScape Renderer) es un programa que está siendo desarrollado actualmente por la Universität Rostock, y previamente por la Technische Universität Berlin. El SSR es una herramienta diseñada para la reproducción y renderización de audio 2D en tiempo real. Para ello utiliza diversos algoritmos, algunos orientados a sistemas formados por arrays de altavoces en diferentes configuraciones y otros algoritmos diseñados para cascos. El principal objetivo de este proyecto es dotar al SSR de la capacidad de renderizar sonidos binaurales en 3D. Este proyecto está centrado en el binaural renderer del SSR. Este algoritmo se basa en el uso de HRTFs (Head Related Transfer Function). Las HRTFs representan la función de transferencia del sistema formado por la cabeza y el torso del oyente. Esta función es medida desde diferentes ángulos. Con estos datos el binaural renderer puede generar audio en tiempo real simulando la posición de diferentes fuentes. Para poder incluir una base de datos con HRTFs en 3D se ha hecho uso del nuevo formato SOFA (Spatially Oriented Format for Acoustics). Este nuevo formato se encuentra en una fase bastante temprana de su desarrollo. Está pensado para servir como formato estándar para almacenar HRTFs y cualquier otro tipo de medidas acústicas, ya que actualmente cada laboratorio cuenta con su propio formato de almacenamiento y esto hace bastante difícil usar varias bases de datos diferentes en un mismo proyecto. El formato SOFA hace uso del contenedor numérico netCDF, que a su vez esta basado en un contenedor más básico llamado HRTF-5. Para poder incluir el formato SOFA en el binaural renderer del SSR se ha desarrollado una API en C++ para poder crear y leer archivos SOFA con el fin de utilizar los datos contenidos en ellos dentro del SSR.

UWB doublet generation employing cross-phase modulation in a semiconductor optical amplifier mach-zehnder interferometer

In this paper, a novel method to generate ultrawideband (UWB) doublets is proposed and experimentally demonstrated, which is based on exploiting the cross-phase modulation in a semiconductor optical amplifier (SOA). The key component is an integrated SOA Mach-Zehnder interferometer pumped with an optical carrier modulated by a Gaussian pulse. The transfer function of the nonlinear conversion process leads to the generation of UWB doublet pulses by tuning the SOA currents to different values.

UWB doublet generation employing non-linear effects of a Semiconductor Optical Amplifier Mach-Zehnder inteferometer

In this article, a novel method to generate an ultra-wideband (UWB) doublet using the cross-phase modulation (XPM) effect is proposed and experimentally demonstrated. The main component of the submitted architecture is a SOA-Mach-Zehnder interferometer (MZI) pumped with a modulated Gaussian pulse. Maximum and minimum conversion points are analyzed through the systems transfer function in order to determinate the most effective operation stage. By tuning different values for the SOAs currents, it is possible to identify a conversion step in which the input pulse is enough large to saturate the SOAMZI, leading to the generation of a UWB doublet pulse.

UWB doublet generation in an integrated semiconductor optical amplifier Mach-Zehnder interferometer

In this paper, we propose and experimentally demonstrate a novel technique to generate ultrawideband (UWB) doublet pulses by exploiting the cross-phase modulation (XPM) in a semiconductor optical amplifier (SOA). The key component in the proposed system consists on an integrated SOA Mach-Zehnder interferometer (MZI) pumped with a Gaussian pulse modulated optical carrier. The transfer function of the nonlinear conversion process leads to the generation of UWB doublet pulses through the control of the biasing point of the SOA-MZI.