17 resultados para instantaneous frequency measurement
em Universidad Politécnica de Madrid
Resumo:
A new method for measuring the linewidth enhancement factor (α-parameter) of semiconductor lasers is proposed and discussed. The method itself provides an estimation of the measurement error, thus self-validating the entire procedure. The α-parameter is obtained from the temporal profile and the instantaneous frequency (chirp) of the pulses generated by gain switching. The time resolved chirp is measured with a polarization based optical differentiator. The accuracy of the obtained values of the α-parameter is estimated from the comparison between the directly measured pulse spectrum and the spectrum reconstructed from the chirp and the temporal profile of the pulse. The method is applied to a VCSEL and to a DFB laser emitting around 1550 nm at different temperatures, obtaining a measurement error lower than ± 8%.
Resumo:
This paper presents an automatic modulation classifier for electronic warfare applications. It is a pattern recognition modulation classifier based on statistical features of the phase and instantaneous frequency. This classifier runs in a real time operation mode with sampling rates in excess of 1 Gsample/s. The hardware platform for this application is a Field Programmable Gate Array (FPGA). This AMC is subsidiary of a digital channelised receiver also implemented in the same platform.
Resumo:
We present an educational software addressed to the students of optical communication courses, for a simple visualization of the basic dynamic processes of semiconductor lasers. The graphic interface allows the user to choose the laser and the modulation parameters and it plots the laser power output and instantaneous frequency versus time. Additionally, the optical frequency variations are numerically shifted into the audible frequency range in order to produce a sound wave from the computer loudspeakers. Using the proposed software, the student can simultaneously see and hear how the laser intensity and frequency change, depending on the modulation and device parameters.
Resumo:
In this work, educational software for intuitive understanding of the basic dynamic processes of semiconductor lasers is presented. The proposed tool is addressed to the students of optical communication courses, encouraging self consolidation of the subjects learned in lectures. The semiconductor laser model is based on the well known rate equations for the carrier density, photon density and optical phase. The direct modulation of the laser is considered with input parameters which can be selected by the user. Different options for the waveform, amplitude and frequency of thpoint. Simulation results are plotted for carrier density and output power versus time. Instantaneous frequency variations of the laser output are numerically shifted to the audible frequency range and sent to the computer loudspeakers. This results in an intuitive description of the “chirp” phenomenon due to amplitude-phase coupling, typical of directly modulated semiconductor lasers. In this way, the student can actually listen to the time resolved spectral content of the laser output. By changing the laser parameters and/or the modulation parameters,consequent variation of the laser output can be appreciated in intuitive manner. The proposed educational tool has been previously implemented by the same authors with locally executable software. In the present manuscript, we extend our previous work to a web based platform, offering improved distribution and allowing its use to the wide audience of the web.
Resumo:
El objetivo de este trabajo fin de grado (TFG) consiste en estudiar algunas técnicas de análisis tiempo-frecuencia y aplicarlas a la detección de señales radar. Estas técnicas se incorporan en los actuales equipos de guerra electrónica radar, tales como los interceptadores digitales. La principal motivación de estos equipos consiste en detectar y localizar las fuentes radiantes enemigas e intentar obtener cierta información de las señales interceptadas, tal como, la dirección de llegada (DOA, Direction Of Arrival), el tiempo de llegada (TOA, Time Of Arrival), amplitud de pulso (PA, Pulse Amplitude), anchura de pulso (PW, Pulse Width), frecuencia instantánea (IF, Instantaneous Frequency) o modulación intrapulso. Se comenzará con un estudio detallado de la Short-Time Fourier Transform (STFT),dado su carácter lineal es la técnica más explotada actualmente. Este algoritmo presenta una mala resolución conjunta tiempo-frecuencia. Este hecho provoca el estudio complementario de una segunda técnica de análisis basada en la distribución de Wigner-Ville (WVD). Mediante este método se logra una resolución optima tiempo-frecuencia. A cambio, se obtienen términos cruzados indeseados debido a su carácter cuadrático. Uno de los objetivos de este TFG reside en calcular la sensibilidad de los sistemas de detección analizados a partir de las técnicas tiempo-frecuencia. Se hará uso del método de Monte Carlo para estimar ciertos parámetros estadísticos del sistema tales como la probabilidad de falsa alarma y de detección. Así mismo, se llevará a cabo el estudio completo de un receptor digital de guerra electrónica a fin de comprender el funcionamiento de todos los subsistemas que componen el conjunto (STFT/WVD, medidor instantáneo de frecuencias, procesamiento no coherente y generación de descriptores de pulso). Por último, se analizará su comportamiento frente a diferentes señales Radar (FM-lineal, BPSK, chirp o Barker). Se utilizará para ello la herramienta Matlab.
Resumo:
When aqueous suspensions of gold nanorods are irradiated with a pulsing laser (808 nm), pressure waves appear even at low frequencies (pulse repetition rate of 25 kHz). We found that the pressure wave amplitude depends on the dynamics of the phenomenon. For fixed concentration and average laser current intensity, the amplitude of the pressure waves shows a trend of increasing with the pulse slope and the pulse maximum amplitude.We postulate that the detected ultrasonic pressure waves are a sort of shock waves that would be generated at the beginning of each pulse, because the pressure wave amplitude would be the result of the positive interference of all the individual shock waves.
Resumo:
This paper presents a new methodology for measurement of the instantaneous average exhaust mass flow rate in reciprocating internal combustion engines to be used to determinate real driving emissions on light duty vehicles, as part of a Portable Emission Measurement System (PEMS). Firstly a flow meter, named MIVECO flow meter, was designed based on a Pitot tube adapted to exhaust gases which are characterized by moisture and particle content, rapid changes in flow rate and chemical composition, pulsating and reverse flow at very low engine speed. Then, an off-line methodology was developed to calculate the instantaneous average flow, considering the ?square root error? phenomenon. The paper includes the theoretical fundamentals, the developed flow meter specifications, the calibration tests, the description of the proposed off-line methodology and the results of the validation test carried out in a chassis dynamometer, where the validity of the mass flow meter and the methodology developed are demonstrated.
Resumo:
This paper presents a theoretical analysis of possible jitter impact in application of numeric criterion for fastmeasurement of frequency by coincidence principle. The primary goal is the generation of a signal containing a known amount of each jitter components. This signal was used for testing signals with regular pulse trains. Initially, jitter components are analyzed and modeled individually. Next, sequences for combining different kinds of jitter are modeled, simulated and evaluated. Jitter model simulation in Matlab is utilized to show the independence of frequencymeasurement results on the total jitter present in the reference and desired pulse trains independently. A good agreement between previously introduced theory of fastmeasurement of frequency and simulation in jitter presence is verified; these results allows to engineers use the numeric criterion for fastmeasurement of frequency in spite to interactions among jitter components in various applications for frequency domain sensors.
Resumo:
Computing the modal parameters of structural systems often requires processing data from multiple non-simultaneously recorded setups of sensors. These setups share some sensors in common, the so-called reference sensors, which are fixed for all measurements, while the other sensors change their position from one setup to the next. One possibility is to process the setups separately resulting in different modal parameter estimates for each setup. Then, the reference sensors are used to merge or glue the different parts of the mode shapes to obtain global mode shapes, while the natural frequencies and damping ratios are usually averaged. In this paper we present a new state space model that processes all setups at once. The result is that the global mode shapes are obtained automatically, and only a value for the natural frequency and damping ratio of each mode is estimated. We also investigate the estimation of this model using maximum likelihood and the Expectation Maximization algorithm, and apply this technique to simulated and measured data corresponding to different structures.
Resumo:
This paper presents the analysis of the reflections in two kind of spherical far field ranges: one if the classical acquisition where the AUT is rotated and the second one corresponds to the systems where the AUT is fixed and the antenna probe is rotated. In large far field systems this is not possible, but this can be used to the measurement of small antennas, for instance, with the SATIMO StarGate system. In both cases, it is assumed that only one frequency is acquired and the results should be improved cut by cut, in order not to lose the advantages or far field measurements. Finally, some practical results are studied using measurements of one antenna in the outdoor far field facility of LIT INPE in Brazil.
Resumo:
This paper explains the progress accomplished in the WP03 of the Terasense Project (TERAHERTZ TECHNOLOGY FOR ELECTROMAGNETIC SENSING APPLICATIONS) approved in the 2008 CONSOLIDERINGENIO program (project CSD2008-0068). The Radiation and Sensor Measurement Lab (RSMLab) is a laboratory based in the existing antenna measurement laboratories at UPM, UC3 and UNiOvi and the new capacities to extend the measurement range from the millimetre wave to the THz region. This laboratory is intended to be shared in more than one place and with more than one institution, in such a way that we could take advantage of other research financial sources and contributions from other institutions with interest in the same field of measurements. One important task will be the international links between the RSMLab and other European and international institutions dedicated to the antenna and sensor measurement in the same frequency range.
Resumo:
Computing the modal parameters of large structures in Operational Modal Analysis often requires to process data from multiple non simultaneously recorded setups of sensors. These setups share some sensors in common, the so-called reference sensors that are fixed for all the measurements, while the other sensors are moved from one setup to the next. One possibility is to process the setups separately what result in different modal parameter estimates for each setup. Then the reference sensors are used to merge or glue the different parts of the mode shapes to obtain global modes, while the natural frequencies and damping ratios are usually averaged. In this paper we present a state space model that can be used to process all setups at once so the global mode shapes are obtained automatically and subsequently only a value for the natural frequency and damping ratio of each mode is computed. We also present how this model can be estimated using maximum likelihood and the Expectation Maximization algorithm. We apply this technique to real data measured at a footbridge.
Resumo:
El requerimiento de proveer alta frecuencia de datos en los modernos sistema de comunicación inalámbricos resulta en complejas señales moduladas de radio-frequencia (RF) con un gran ancho de banda y alto ratio pico-promedio (PAPR). Para garantizar la linealidad del comportamiento, los amplificadores lineales de potencia comunes funcionan típicamente entre 4 y 10 dB de back-o_ desde la máxima potencia de salida, ocasionando una baja eficiencia del sistema. La eliminación y restauración de la evolvente (EER) y el seguimiento de la evolvente (ET) son dos prometedoras técnicas para resolver el problema de la eficiencia. Tanto en EER como en ET, es complicado diseñar un amplificador de potencia que sea eficiente para señales de RF de alto ancho de banda y alto PAPR. Una propuesta común para los amplificadores de potencia es incluir un convertidor de potencia de muy alta eficiencia operando a frecuencias más altas que el ancho de banda de la señal RF. En este caso, la potencia perdida del convertidor ocasionado por la alta frecuencia desaconseja su práctica cuando el ancho de banda es muy alto. La solución a este problema es el enfoque de esta disertación que presenta dos arquitecturas de amplificador evolvente: convertidor híbrido-serie con una técnica de evolvente lenta y un convertidor multinivel basado en un convertidor reductor multifase con control de tiempo mínimo. En la primera arquitectura, una topología híbrida está compuesta de una convertidor reductor conmutado y un regulador lineal en serie que trabajan juntos para ajustar la tensión de salida para seguir a la evolvente con precisión. Un algoritmo de generación de una evolvente lenta crea una forma de onda con una pendiente limitada que es menor que la pendiente máxima de la evolvente original. La salida del convertidor reductor sigue esa forma de onda en vez de la evolvente original usando una menor frecuencia de conmutación, porque la forma de onda no sólo tiene una pendiente reducida sino también un menor ancho de banda. De esta forma, el regulador lineal se usa para filtrar la forma de onda tiene una pérdida de potencia adicional. Dependiendo de cuánto se puede reducir la pendiente de la evolvente para producir la forma de onda, existe un trade-off entre la pérdida de potencia del convertidor reductor relacionada con la frecuencia de conmutación y el regulador lineal. El punto óptimo referido a la menor pérdida de potencia total del amplificador de evolvente es capaz de identificarse con la ayuda de modelo preciso de pérdidas que es una combinación de modelos comportamentales y analíticos de pérdidas. Además, se analiza el efecto en la respuesta del filtro de salida del convertidor reductor. Un filtro de dampeo paralelo extra es necesario para eliminar la oscilación resonante del filtro de salida porque el convertidor reductor opera en lazo abierto. La segunda arquitectura es un amplificador de evolvente de seguimiento de tensión multinivel. Al contrario que los convertidores que usan multi-fuentes, un convertidor reductor multifase se emplea para generar la tensión multinivel. En régimen permanente, el convertidor reductor opera en puntos del ciclo de trabajo con cancelación completa del rizado. El número de niveles de tensión es igual al número de fases de acuerdo a las características del entrelazamiento del convertidor reductor. En la transición, un control de tiempo mínimo (MTC) para convertidores multifase es novedosamente propuesto y desarrollado para cambiar la tensión de salida del convertidor reductor entre diferentes niveles. A diferencia de controles convencionales de tiempo mínimo para convertidores multifase con inductancia equivalente, el propuesto MTC considera el rizado de corriente por cada fase basado en un desfase fijo que resulta en diferentes esquemas de control entre las fases. La ventaja de este control es que todas las corrientes vuelven a su fase en régimen permanente después de la transición para que la siguiente transición pueda empezar muy pronto, lo que es muy favorable para la aplicación de seguimiento de tensión multinivel. Además, el control es independiente de la carga y no es afectado por corrientes de fase desbalanceadas. Al igual que en la primera arquitectura, hay una etapa lineal con la misma función, conectada en serie con el convertidor reductor multifase. Dado que tanto el régimen permanente como el estado de transición del convertidor no están fuertemente relacionados con la frecuencia de conmutación, la frecuencia de conmutación puede ser reducida para el alto ancho de banda de la evolvente, la cual es la principal consideración de esta arquitectura. La optimización de la segunda arquitectura para más alto anchos de banda de la evolvente es presentada incluyendo el diseño del filtro de salida, la frecuencia de conmutación y el número de fases. El área de diseño del filtro está restringido por la transición rápida y el mínimo pulso del hardware. La rápida transición necesita un filtro pequeño pero la limitación del pulso mínimo del hardware lleva el diseño en el sentido contrario. La frecuencia de conmutación del convertidor afecta principalmente a la limitación del mínimo pulso y a las pérdidas de potencia. Con una menor frecuencia de conmutación, el ancho de pulso en la transición es más pequeño. El número de fases relativo a la aplicación específica puede ser optimizado en términos de la eficiencia global. Otro aspecto de la optimización es mejorar la estrategia de control. La transición permite seguir algunas partes de la evolvente que son más rápidas de lo que el hardware puede soportar al precio de complejidad. El nuevo método de sincronización de la transición incrementa la frecuencia de la transición, permitiendo que la tensión multinivel esté más cerca de la evolvente. Ambas estrategias permiten que el convertidor pueda seguir una evolvente con un ancho de banda más alto que la limitación de la etapa de potencia. El modelo de pérdidas del amplificador de evolvente se ha detallado y validado mediante medidas. El mecanismo de pérdidas de potencia del convertidor reductor tiene que incluir las transiciones en tiempo real, lo cual es diferente del clásico modelos de pérdidas de un convertidor reductor síncrono. Este modelo estima la eficiencia del sistema y juega un papel muy importante en el proceso de optimización. Finalmente, la segunda arquitectura del amplificador de evolvente se integra con el amplificador de clase F. La medida del sistema EER prueba el ahorro de energía con el amplificador de evolvente propuesto sin perjudicar la linealidad del sistema. ABSTRACT The requirement of delivering high data rates in modern wireless communication systems results in complex modulated RF signals with wide bandwidth and high peak-to-average ratio (PAPR). In order to guarantee the linearity performance, the conventional linear power amplifiers typically work at 4 to 10 dB back-off from the maximum output power, leading to low system efficiency. The envelope elimination and restoration (EER) and envelope tracking (ET) are two promising techniques to overcome the efficiency problem. In both EER and ET, it is challenging to design efficient envelope amplifier for wide bandwidth and high PAPR RF signals. An usual approach for envelope amplifier includes a high-efficiency switching power converter operating at a frequency higher than the RF signal's bandwidth. In this case, the power loss of converter caused by high switching operation becomes unbearable for system efficiency when signal bandwidth is very wide. The solution of this problem is the focus of this dissertation that presents two architectures of envelope amplifier: a hybrid series converter with slow-envelope technique and a multilevel converter based on a multiphase buck converter with the minimum time control. In the first architecture, a hybrid topology is composed of a switched buck converter and a linear regulator in series that work together to adjust the output voltage to track the envelope with accuracy. A slow envelope generation algorithm yields a waveform with limited slew rate that is lower than the maximum slew rate of the original envelope. The buck converter's output follows this waveform instead of the original envelope using lower switching frequency, because the waveform has not only reduced slew rate but also reduced bandwidth. In this way, the linear regulator used to filter the waveform has additional power loss. Depending on how much reduction of the slew rate of envelope in order to obtain that waveform, there is a trade-off between the power loss of buck converter related to the switching frequency and the power loss of linear regulator. The optimal point referring to the lowest total power loss of this envelope amplifier is identified with the help of a precise power loss model that is a combination of behavioral and analytic loss model. In addition, the output filter's effect on the response is analyzed. An extra parallel damping filter is needed to eliminate the resonant oscillation of output filter L and C, because the buck converter operates in open loop. The second architecture is a multilevel voltage tracking envelope amplifier. Unlike the converters using multi-sources, a multiphase buck converter is employed to generate the multilevel voltage. In the steady state, the buck converter operates at complete ripple cancellation points of duty cycle. The number of the voltage levels is equal to the number of phases according the characteristics of interleaved buck converter. In the transition, a minimum time control (MTC) for multiphase converter is originally proposed and developed for changing the output voltage of buck converter between different levels. As opposed to conventional minimum time control for multiphase converter with equivalent inductance, the proposed MTC considers the current ripple of each phase based on the fixed phase shift resulting in different control schemes among the phases. The advantage of this control is that all the phase current return to the steady state after the transition so that the next transition can be triggered very soon, which is very favorable for the application of multilevel voltage tracking. Besides, the control is independent on the load condition and not affected by the unbalance of phase current. Like the first architecture, there is also a linear stage with the same function, connected in series with the multiphase buck converter. Since both steady state and transition state of the converter are not strongly related to the switching frequency, it can be reduced for wide bandwidth envelope which is the main consideration of this architecture. The optimization of the second architecture for wider bandwidth envelope is presented including the output filter design, switching frequency and the number of phases. The filter design area is restrained by fast transition and the minimum pulse of hardware. The fast transition needs small filter but the minimum pulse of hardware limitation pushes the filter in opposite way. The converter switching frequency mainly affects the minimum pulse limitation and the power loss. With lower switching frequency, the pulse width in the transition is smaller. The number of phases related to specific application can be optimized in terms of overall efficiency. Another aspect of optimization is improving control strategy. Transition shift allows tracking some parts of envelope that are faster than the hardware can support at the price of complexity. The new transition synchronization method increases the frequency of transition, allowing the multilevel voltage to be closer to the envelope. Both control strategies push the converter to track wider bandwidth envelope than the limitation of power stage. The power loss model of envelope amplifier is detailed and validated by measurements. The power loss mechanism of buck converter has to include the transitions in real time operation, which is different from classical power loss model of synchronous buck converter. This model estimates the system efficiency and play a very important role in optimization process. Finally, the second envelope amplifier architecture is integrated with a Class F amplifier. EER system measurement proves the power saving with the proposed envelope amplifier without disrupting the linearity performance.
Resumo:
We report on an experimental study on the spin-waves relaxation rate in two series of nanodisks of diameter ϕ=300 , 500, and 700 nm, patterned out of two systems: a 20 nm thick yttrium iron garnet (YIG) film grown by pulsed laser deposition either bare or covered by 13 nm of Pt. Using a magnetic resonance force microscope, we measure precisely the ferromagnetic resonance linewidth of each individual YIG and YIG|Pt nanodisks. We find that the linewidth in the nanostructure is sensibly smaller than the one measured in the extended film. Analysis of the frequency dependence of the spectral linewidth indicates that the improvement is principally due to the suppression of the inhomogeneous part of the broadening due to geometrical confinement, suggesting that only the homogeneous broadening contributes to the linewidth of the nanostructure. For the bare YIG nano-disks, the broadening is associated to a damping constant α=4 × 10−4 . A threefold increase of the linewidth is observed for the series with Pt cap layer, attributed to the spin pumping effect. The measured enhancement allows to extract the spin mixing conductance found to be G↑↓=1.55 × 1014 Ω−1 m−2 for our YIG(20nm)|Pt interface, thus opening large opportunities for the design of YIG based nanostructures with optimized magnetic losses.
Resumo:
In this paper, a model of the measuring process of sonic anemometers with more than one measuring path is presented. The main hypothesis of the work is that the time variation of the turbulent speed field during the sequence of pulses that produces a measure of the wind speed vector affects the measurement. Therefore, the previously considered frozen flow, or instantaneous averaging, condition is relaxed. This time variation, quantified by the mean Mach number of the flow and the time delay between consecutive pulses firings, in combination with both the full geometry of sensors (acoustic path location and orientation) and the incidence angles of the mean with speed vector, give rise to significant errors in the measurement of turbulence which are not considered by models based on the hypothesis of instantaneous line averaging. The additional corrections (relative to the ones proposed by instantaneous line-averaging models) are strongly dependent on the wave number component parallel to the mean wind speed, the time delay between consecutive pulses, the Mach number of the flow, the geometry of the sensor and the incidence angles of mean wind speed vector. Kaimal´s limit k W1=1/l (where k W1 is the wave number component parallel to mean wind speed and l is the path length) for the maximum wave numbers from which the sonic process affects the measurement of turbulence is here generalized as k W1=C l /l, where C l is usually lesser than unity and depends on all the new parameters taken into account by the present model.