948 resultados para Signal processing-oriented solution
Resumo:
Intermittency phenomenon is a continuous route from regular to chaotic behaviour. Intermittency is an occurrence of a signal that alternates chaotic bursts between quasi-regular periods called laminar phases, driven by the so called reinjection probability density function (RPD). In this paper is introduced a new technique to obtain the RPD for type-II and III intermittency. The new RPD is more general than the classical one and includes the classical RPD as a particular case. The probabilities of the laminar length, the average laminar lengths and the characteristic relations are determined with and without lower bound of the reinjection in agreement with numerical simulations. Finally, it is analyzed the noise effect in intermittency. A method to obtain the noisy RPD is developed extending the procedure used in the noiseless case. The analytical results show a good agreement with numerical simulations.
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We consider the problem of developing efficient sampling schemes for multiband sparse signals. Previous results on multicoset sampling implementations that lead to universal sampling patterns (which guarantee perfect reconstruction), are based on a set of appropriate interleaved analog to digital converters, all of them operating at the same sampling frequency. In this paper we propose an alternative multirate synchronous implementation of multicoset codes, that is, all the analog to digital converters in the sampling scheme operate at different sampling frequencies, without need of introducing any delay. The interleaving is achieved through the usage of different rates, whose sum is significantly lower than the Nyquist rate of the multiband signal. To obtain universal patterns the sampling matrix is formulated and analyzed. Appropriate choices of the parameters, that is the block length and the sampling rates, are also proposed.
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Many problems in digital communications involve wideband radio signals. As the most recent example, the impressive advances in Cognitive Radio systems make even more necessary the development of sampling schemes for wideband radio signals with spectral holes. This is equivalent to considering a sparse multiband signal in the framework of Compressive Sampling theory. Starting from previous results on multicoset sampling and recent advances in compressive sampling, we analyze the matrix involved in the corresponding reconstruction equation and define a new method for the design of universal multicoset codes, that is, codes guaranteeing perfect reconstruction of the sparse multiband signal.
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Structural Health Monitoring (SHM) requires integrated "all in one" electronic devices capable of performing analysis of structural integrity and on-board damage detection in aircraft?s structures. PAMELA III (Phased Array Monitoring for Enhanced Life Assessment, version III) SHM embedded system is an example of this device type. This equipment is capable of generating excitation signals to be applied to an array of integrated piezoelectric Phased Array (PhA) transducers stuck to aircraft structure, acquiring the response signals, and carrying out the advanced signal processing to obtain SHM maps. PAMELA III is connected with a host computer in order to receive the configuration parameters and sending the obtained SHM maps, alarms and so on. This host can communicate with PAMELA III through an Ethernet interface. To avoid the use of wires where necessary, it is possible to add Wi-Fi capabilities to PAMELA III, connecting a Wi-Fi node working as a bridge, and to establish a wireless communication between PAMELA III and the host. However, in a real aircraft scenario, several PAMELA III devices must work together inside closed structures. In this situation, it is not possible for all PAMELA III devices to establish a wireless communication directly with the host, due to the signal attenuation caused by the different obstacles of the aircraft structure. To provide communication among all PAMELA III devices and the host, a wireless mesh network (WMN) system has been implemented inside a closed aluminum wingbox. In a WMN, as long as a node is connected to at least one other node, it will have full connectivity to the entire network because each mesh node forwards packets to other nodes in the network as required. Mesh protocols automatically determine the best route through the network and can dynamically reconfigure the network if a link drops out. The advantages and disadvantages on the use of a wireless mesh network system inside closed aerospace structures are discussed.
Resumo:
A fully integrated on-board electronic system that can perform in-situ structural health monitoring (SHM) of aircraft?s structures using specifically designed equipment for SHM based on guided wave ultrasonic method or Lamb waves? method is introduced. This equipment is called Phased Array Monitoring for Enhanced Life Assessment (PAMELA III) and is an essential part of overall PAMELA SHM? system. PAMELA III can generate any kind of excitation signals, acquire the response signals that propagate throughout the structure being tested, and perform the signal processing for damage detection directly on the structure without need to send the huge amount of raw signals but only the final SHM maps. It monitors the structure by means of an array of integrated Phased Array (PhA) transducers preferably bonded onto the host structure. The PAMELA III hardware for SHM mapping has been designed, built and subjected to laboratory tests, using aluminum and CFRP structures. The 12 channel system has been designed to be low weight (265 grams only), to have a small form factor, to be directly mounted above the integrated PhA transducers without need for cables and to be EMI protected so that the equipment can be taken on board an aircraft to perform required SHM analyses by use of embedded SHM algorithms. Moreover, the autonomous, automatic and on real-time working procedure makes it suitable for the avionic field, sending the corresponding alerts, maps and reports to external equipment.
Resumo:
n recent years, the development of advanced driver assistance systems (ADAS) – mainly based on lidar and cameras – has considerably improved the safety of driving in urban environments. These systems provide warning signals for the driver in the case that any unexpected traffic circumstance is detected. The next step is to develop systems capable not only of warning the driver but also of taking over control of the car to avoid a potential collision. In the present communication, a system capable of autonomously avoiding collisions in traffic jam situations is presented. First, a perception system was developed for urban situations—in which not only vehicles have to be considered, but also pedestrians and other non-motor-vehicles (NMV). It comprises a differential global positioning system (DGPS) and wireless communication for vehicle detection, and an ultrasound sensor for NMV detection. Then, the vehicle's actuators – brake and throttle pedals – were modified to permit autonomous control. Finally, a fuzzy logic controller was implemented capable of analyzing the information provided by the perception system and of sending control commands to the vehicle's actuators so as to avoid accidents. The feasibility of the integrated system was tested by mounting it in a commercial vehicle, with the results being encouraging.
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MPEG-M is a suite of ISO/IEC standards (ISO/IEC 23006) that has been developed under the auspices of Moving Picture Experts Group (MPEG). MPEG-M, also known as Multimedia Service Platform Technologies (MSPT), facilitates a collection of multimedia middleware APIs and elementary services as well as service aggregation so that service providers can offer users a plethora of innovative services by extending current IPTV technology toward the seamless integration of personal content creation and distribution, e-commerce, social networks and Internet distribution of digital media.
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In current communication systems, there are many new challenges like various competitive standards, the scarcity of frequency resource, etc., especially the development of personal wireless communication systems result the new system update faster than ever before, the conventional hardware-based wireless communication system is difficult to adapt to this situation. The emergence of SDR enabled the third revolution of wireless communication which from hardware to software and build a flexible, reliable, upgradable, reusable, reconfigurable and low cost platform. The Universal Software Radio Peripheral (USRP) products are commonly used with the GNU Radio software suite to create complex SDR systems. GNU Radio is a toolkit where digital signal processing blocks are written in C++, and connected to each other with Python. This makes it easy to develop more sophisticated signal processing systems, because many blocks already written by others and you can quickly put them together to create a complete system. Although the main function of GNU Radio is not be a simulator, but if there is no RF hardware components,it supports to researching the signal processing algorithm based on pre-stored and generated data by signal generator. This thesis introduced SDR platform from hardware (USRP) and software(GNU Radio), as well as some basic modulation techniques in wireless communication system. Based on the examples provided by GNU Radio, carried out some related experiments, for example GSM scanning and FM radio station receiving on USRP. And make a certain degree of improvement based on the experience of some investigators to observe OFDM spectrum and simulate real-time video transmission. GNU Radio combine with USRP hardware proved to be a valuable lab platform for implementing complex radio system prototypes in a short time. RESUMEN. Software Defined Radio (SDR) es una tecnología emergente que está creando un impacto revolucionario en la tecnología de radio convencional. Un buen ejemplo de radio software son los sistemas de código abierto llamados GNU Radio que emplean un kit de herramientas de desarrollo de software libre. En este trabajo se ha empleado un kit de desarrollo comercial (Ettus Research) que consiste en un módulo de procesado de señal y un hardaware sencillo. El módulo emplea un software de desarrollo basado en Linux sobre el que se pueden implementar aplicaciones de radio software muy variadas. El hardware de desarrollo consta de un microprocesador de propósito general, un dispositivo programable (FPGA) y un interfaz de radiofrecuencia que cubre de 50 a 2200MHz. Este hardware se conecta al PC por medio de un interfaz USB de 8Mb/s de velocidad. Sobre la plataforma de Ettus se pueden ejecutar aplicaciones GNU radio que utilizan principalmente lenguaje de programación Python para implementarse. Sin embargo, su módulo de procesado de señal está construido en C + + y emplea un microprocesador con aritmética de coma flotante. Por lo tanto, los desarrolladores pueden rápida y fácilmente construir aplicaciones en tiempo real sistemas de comunicación inalámbrica de alta capacidad. Aunque su función principal no es ser un simulador, si no puesto que hay componentes de hardware RF, Radio GNU sirve de apoyo a la investigación del algoritmo de procesado de señales basado en pre-almacenados y generados por los datos del generador de señal. En este trabajo fin de máster se ha evaluado la plataforma de hardware de DEG (USRP) y el software (GNU Radio). Para ello se han empleado algunas técnicas de modulación básicas en el sistema de comunicación inalámbrica. A partir de los ejemplos proporcionados por GNU Radio, hemos realizado algunos experimentos relacionados, por ejemplo, escaneado del espectro, demodulación de señales de FM empleando siempre el hardware de USRP. Una vez evaluadas aplicaciones sencillas se ha pasado a realizar un cierto grado de mejora y optimización de aplicaciones complejas descritas en la literatura. Se han empleado aplicaciones como la que consiste en la generación de un espectro de OFDM y la simulación y transmisión de señales de vídeo en tiempo real. Con estos resultados se está ahora en disposición de abordar la elaboración de aplicaciones complejas.
Resumo:
Este proyecto fin de carrera trata del sistema de grabación y reproducción sonora ambiofónico, destacar que este sistema y la tecnología que emplea es de dominio público. La ambiofonía se basa en un amalgama de investigaciones recientes y de los ya bien sabidos principios psicoacústicos y binaurales. Estos avances han expandido nuevas fronteras en lo concerniente a la grabación y reproducción de audio, así como de presentar al oyente un campo sonoro a la entrada de sus oídos lo más parecido posible al campo sonoro al que se expondría al oyente en el momento y lugar de la toma de sonido, es decir, reconstruye un campo sonora binaural. Este sistema ha podido desarrollarse, de una manera bastante satisfactoria, gracias a todos los estudios y textos anteriores en materia de psicoacústica y del mecanismo de escucha humano. Otro factor gracias al cual es posible y asequible, tanto el desarrollo como el disfrute de esta tecnología, es el hecho que en nuestros días es muy económico disponer de ordenadores lo suficientemente potentes y rápidos para realizar el procesado de señales que se requiere de una manera bastante rápida. Los desarrolladores de dicha tecnología han publicado diversos documentos y archivos descargables de la red con aplicaciones para la implementación de sistemas ambiofónicos de manera gratuita para uso privado. El sistema ambiofónico se basa en la combinación de factores psicoacústicos ignorados o subestimados y lo ya sabido sobre las propiedades acústicas de salas, tanto de salas en las que tienen lugar las ejecuciones musicales (auditorios, teatros, salas de conciertos...), como de salas de escucha (salones de domicilios, controles de estudios...). En la parte práctica del proyecto se van a realizar una serie de grabaciones musicales empleando tanto técnicas estereofónicas tradicionales como ambiofónicas de grabación con el fin de describir y comparar ambas técnicas microfónicas. También servirá para estudiar hasta que punto es favorable subjetivamente para el oyente el hecho de realizar la toma de sonido teniendo en cuenta las propiedades del sistema de reproducción ambiofónico. Esta comparación nos dará una idea de hasta donde se puede llegar, en cuanto a sensación de realidad para el oyente, al tener en cuenta durante el proceso de grabación efectos como la respuesta del pabellón auditivo del oyente, la cual es única, y que posteriormente la diafonía interaural va a ser cancelada mediante un procesado digital de señal. ABSTRACT. This final project is about the ambiophonic recording and playback system, note that this system and the technology it uses is of public domain. Ambiophonics is based on an amalgam of recent research and to the well known and binaural psychoacoustic principles. These advances have expanded new frontiers with regard to the audio recording and playback, as well as to present the listener a sound field at the entrance of their ears as close as possible to the sound field that would the listener be exposed to at the time and place of the mucial interpretation, so we can say that ambiophonics reconstructs a binaural sound field . This system has been developed, in a fairly satisfactory way, thanks to all the studies and previous texts on psychoacoustics and human listening mechanism. Another factor by which it is possible and affordable, both the development and the enjoyment of this technology, is the fact that in our days is inexpensive to usres to own computers that are powerful and fast enough to perform the signal processing that is required in a short time. The developers of this technology have published several documents and downloadable files on the network with applications for ambiophonics system implementation for free. Ambiophonics is based on a combination of factors ignored or underestimated psychoacousticly and what is already known about the acoustic properties of rooms, including rooms where musical performances take place (auditoriums, theaters, concert halls...), and listening rooms (concet halls, studios controls...). In the practical part of the project will be making a series of musical recordings using both traditional stereo recording techniques and recording techiniques compatible with ambiophonics in order to describe and compare both recording techniques. It will also examine to what extent is subjectively favorable for the listener the fact of considering the playback system properties of ambiophonics during the recording stage. This comparison will give us an idea of how far can we get, in terms of sense of reality to the listener, keeping in mind during the recording process the effects introduced by the response of the ear of the listener, which is unique, and that the subsequently interaural crosstalk will be canceled by a digital signal processing.
Resumo:
El objetivo principal del presente proyecto es proporcionar al ingeniero de telecomunicaciones una visión general de las técnicas que se utilizan en el modelado del sistema auditivo. El modelado del sistema auditivo se realiza con los siguientes objetivos: a) Interpretar medidas directas, b)unificar el entendimiento de diferentes fenómenos, c) guiar estrategias de amplificación para suplir pérdidas auditivas y d) tener predicciones experimentalmente comprobables de comportamientos, con diferentes niveles de complejidad. En este trabajo se tratarán y explicarán brevemente las diferentes técnicas utilizadas para modelar las partes del sistema auditivo, desde las analogías electroacústicas, modelos biofísicos, binaurales, hasta la implementación de filtros auditivos mediante procesado de señal. Podemos concluir que el modelado mediante analogías electroacústicas permite una rápida implementación y entendimiento, pero tiene ciertas limitaciones. Las simulaciones mediante análisis numéricos son precisas y de gran utilidad tanto para del oído medio como para el interno. El procesado de señal es el procedimiento más completo y utilizado ya que permite modelar oído externo y medio además de permitir la implementación de filtros cocleares muy precisos y coherentes con la realidad incluyéndolos en modelos perceptivos. ABSTRACT. The main aim of the Project is to provide the Telecommunications Engineer an overview about the approaches for modelling the auditory system. The auditory system modelling is done for the next objectives: a) Interpret direct measures, b) Understand different phenomena c) get strategies of amplification for hearing impaired people and d) Obtain testable predictions experimentally about some behaviors with different complexity levels. Inside this document, several approaches about modeling of the auditory system parts will be explained: analog circuits, biophysics models, binaural models, and auditory filters made through signal processing. In conclusion, analog circuits are made quickly and they are easier to understand but they have many limitations. Simulations through numerical analysis are accurate and useful in middle and inner ear models. Signal processing is the more versatile approach because it lets to make a model of external and middle ear and then it allows to make complex auditory filters. Perceptive models can be made entirely through this method.
Resumo:
We report conditions on a switching signal that guarantee that solutions of a switched linear systems converge asymptotically to zero. These conditions are apply to continuous, discrete-time and hybrid switched linear systems, both those having stable subsystems and mixtures of stable and unstable subsystems.
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Stochastic model updating must be considered for quantifying uncertainties inherently existing in real-world engineering structures. By this means the statistical properties,instead of deterministic values, of structural parameters can be sought indicating the parameter variability. However, the implementation of stochastic model updating is much more complicated than that of deterministic methods particularly in the aspects of theoretical complexity and low computational efficiency. This study attempts to propose a simple and cost-efficient method by decomposing a stochastic updating process into a series of deterministic ones with the aid of response surface models and Monte Carlo simulation. The response surface models are used as surrogates for original FE models in the interest of programming simplification, fast response computation and easy inverse optimization. Monte Carlo simulation is adopted for generating samples from the assumed or measured probability distributions of responses. Each sample corresponds to an individual deterministic inverse process predicting the deterministic values of parameters. Then the parameter means and variances can be statistically estimated based on all the parameter predictions by running all the samples. Meanwhile, the analysis of variance approach is employed for the evaluation of parameter variability significance. The proposed method has been demonstrated firstly on a numerical beam and then a set of nominally identical steel plates tested in the laboratory. It is found that compared with the existing stochastic model updating methods, the proposed method presents similar accuracy while its primary merits consist in its simple implementation and cost efficiency in response computation and inverse optimization.
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Monte Carlo techniques, which require the generation of samples from some target density, are often the only alternative for performing Bayesian inference. Two classic sampling techniques to draw independent samples are the ratio of uniforms (RoU) and rejection sampling (RS). An efficient sampling algorithm is proposed combining the RoU and polar RS (i.e. RS inside a sector of a circle using polar coordinates). Its efficiency is shown in drawing samples from truncated Cauchy and Gaussian random variables, which have many important applications in signal processing and communications. RESUMEN. Método eficiente para generar algunas variables aleatorias de uso común en procesado de señal y comunicaciones (por ejemplo, Gaussianas o Cauchy truncadas) mediante la combinación de dos técnicas: "ratio of uniforms" y "rejection sampling".
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The constant development of digital systems in radio communications demands the adaptation of the current receiving equipment to the new technologies. In this context, a new Software Defined Radio based receiver is being implemented with the aim of carrying out different experiments to analyze the propagation of signals through the atmosphere from a satellite beacon. The receiver selected for this task is the PERSEUS SDR from the Italian company Microtelecom s.r.l. It is a software defined VLF-LF-MF-HF receiver based on an outstanding direct sampling digital architecture which features a 14 bit 80 MSamples/s analog-to-digital converter, a high-performance FPGA-based digital down-converter and a high-speed 480 Mbit/s USB2.0 PC interface. The main goal is to implement the related software and adapt the new receiver to the current working environment. In this paper, SDR technology guidelines are given and PERSEUS receiver digital signal processing is presented with the most remarkable results.
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Atrial fibrillation (AF) is a common heart disorder. One of the most prominent hypothesis about its initiation and maintenance considers multiple uncoordinated activation foci inside the atrium. However, the implicit assumption behind all the signal processing techniques used for AF, such as dominant frequency and organization analysis, is the existence of a single regular component in the observed signals. In this paper we take into account the existence of multiple foci, performing a spectral analysis to detect their number and frequencies. In order to obtain a cleaner signal on which the spectral analysis can be performed, we introduce sparsity-aware learning techniques to infer the spike trains corresponding to the activations. The good performance of the proposed algorithm is demonstrated both on synthetic and real data. RESUMEN. Algoritmo basado en técnicas de regresión dispersa para la extracción de las señales cardiacas en pacientes con fibrilación atrial (AF).