Nuevas arquitecturas de transpondedores de microondas = New transponder architectures at microwave frequencies = Nouvelles architectures des transpondeurs d’hyperfréquences

En la última década, los sistemas de telecomunicación de alta frecuencia han evolucionado tremendamente. Las bandas de frecuencias, los anchos de banda del usuario, las técnicas de modulación y otras características eléctricas están en constante cambio de acuerdo a la evolución de la tecnología y la aparición de nuevas aplicaciones. Las arquitecturas de los transceptores modernos son diferentes de las tradicionales. Muchas de las funciones convencionalmente realizadas por circuitos analógicos han sido asignadas gradualmente a procesadores digitales de señal, de esta manera, las fronteras entre la banda base y las funcionalidades de RF se difuminan. Además, los transceptores inalámbricos digitales modernos son capaces de soportar protocolos de datos de alta velocidad, por lo que emplean una elevada escala de integración para muchos de los subsistemas que componen las diferentes etapas. Uno de los objetivos de este trabajo de investigación es realizar un estudio de las nuevas configuraciones en el desarrollo de demostradores de radiofrecuencia (un receptor y un transmisor) y transpondedores para fines de comunicaciones y militares, respectivamente. Algunos trabajos se han llevado a cabo en el marco del proyecto TECRAIL, donde se ha implementado un demostrador de la capa física LTE para evaluar la viabilidad del estándar LTE en el entorno ferroviario. En el ámbito militar y asociado al proyecto de calibración de radares (CALRADAR), se ha efectuado una actividad importante en el campo de la calibración de radares balísticos Doppler donde se ha analizado cuidadosamente su precisión y se ha desarrollado la unidad generadora de Doppler de un patrón electrónico para la calibración de estos radares. Dicha unidad Doppler es la responsable de la elevada resolución en frecuencia del generador de “blancos” radar construido. Por otro lado, se ha elaborado un análisis completo de las incertidumbres del sistema para optimizar el proceso de calibración. En una segunda fase se han propuesto soluciones en el desarrollo de dispositivos electro-ópticos para aplicaciones de comunicaciones. Estos dispositivos son considerados, debido a sus ventajas, tecnologías de soporte para futuros dispositivos y subsistemas de RF/microondas. Algunas demandas de radio definida por software podrían cubrirse aplicando nuevos conceptos de circuitos sintonizables mediante parámetros programables de un modo dinámico. También se ha realizado una contribución relacionada con el diseño de filtros paso banda con topología “Hairpin”, los cuales son compactos y se pueden integrar fácilmente en circuitos de microondas en una amplia gama de aplicaciones destinadas a las comunicaciones y a los sistemas militares. Como importante aportación final, se ha presentado una propuesta para ecualizar y mejorar las transmisiones de señales discretas de temporización entre los TRMs y otras unidades de procesamiento, en el satélite de última generación SEOSAR/PAZ. Tras un análisis exhaustivo, se ha obtenido la configuración óptima de los buses de transmisión de datos de alta velocidad basadas en una red de transceptores. ABSTRACT In the last decade, high-frequency telecommunications systems have extremely evolved. Frequency bands, user bandwidths, modulation techniques and other electrical characteristics of these systems are constantly changing following to the evolution of technology and the emergence of new applications. The architectures of modern transceivers are different from the traditional ones. Many of the functions conventionally performed by analog circuitry have gradually been assigned to digital signal processors. In this way, boundaries between baseband and RF functionalities are diffused. The design of modern digital wireless transceivers are capable of supporting high-speed data protocols. Therefore, a high integration scale is required for many of the components in the block chain. One of the goals of this research work is to investigate new configurations in the development of RF demonstrators (a receiver and a transmitter) and transponders for communications and military purposes, respectively. A LTE physical layer demonstrator has been implemented to assess the viability of LTE in railway scenario under the framework of the TECRAIL project. An important activity, related to the CALRADAR project, for the calibration of Doppler radars with extremely high precision has been performed. The contribution is the Doppler unit of the radar target generator developed that reveals a high frequency resolution. In order to assure the accuracy of radar calibration process, a complete analysis of the uncertainty in the above mentioned procedure has been carried out. Another important research topic has been the development of photonic devices that are considered enabling technologies for future RF and microwave devices and subsystems. Some Software Defined Radio demands are addressed by the proposed novel circuit concepts based on photonically tunable elements with dynamically programmable parameters. A small contribution has been made in the field of Hairpin-line bandpass filters. These filters are compact and can also be easily integrated into microwave circuits finding a wide range of applications in communication and military systems. In this research field, the contributions made have been the improvements in the design and the simulations of wideband filters. Finally, an important proposal to balance and enhance transmissions of discrete timing signals between TRMs and other processing units into the state of the art SEOSAR/PAZ Satellite has been carried out obtaining the optimal configuration of the high-speed data transmission buses based on a transceiver network. RÉSUMÉ Les systèmes d'hyperfréquence dédiés aux télécommunications ont beaucoup évolué dans la dernière décennie. Les bandes de fréquences, les bandes passantes par utilisateur, les techniques de modulation et d'autres caractéristiques électriques sont en constant changement en fonction de l'évolution des technologies et l'émergence de nouvelles applications. Les architectures modernes des transcepteurs sont différentes des traditionnelles. Un grand nombre d’opérations normalement effectuées par les circuits analogiques a été progressivement alloué à des processeurs de signaux numériques. Ainsi, les frontières entre la bande de base et la fonctionnalité RF sont floues. Les transcepteurs sans fils numériques modernes sont capables de transférer des données à haute vitesse selon les différents protocoles de communication utilisés. C'est pour cette raison qu’un niveau élevé d'intégration est nécessaire pour un grand nombre de composants qui constitue les différentes étapes des systèmes. L'un des objectifs de cette recherche est d'étudier les nouvelles configurations dans le développement des démonstrateurs RF (récepteur et émetteur) et des transpondeurs à des fins militaire et de communication. Certains travaux ont été réalisés dans le cadre du projet TECRAIL, où un démonstrateur de la couche physique LTE a été mis en place pour évaluer la faisabilité de la norme LTE dans l'environnement ferroviaire. Une contribution importante, liée au projet CALRADAR, est proposée dans le domaine des systèmes d’étalonnage de radar Doppler de haute précision. Cette contribution est le module Doppler de génération d’hyperfréquence intégré dans le système électronique de génération de cibles radar virtuelles que présente une résolution de fréquence très élevée. Une analyse complète de l'incertitude dans l'étalonnage des radars Doppler a été effectuée, afin d'assurer la précision du calibrage. La conception et la mise en oeuvre de quelques dispositifs photoniques sont un autre sujet important du travail de recherche présenté dans cette thèse. De tels dispositifs sont considérés comme étant des technologies habilitantes clés pour les futurs dispositifs et sous-systèmes RF et micro-ondes grâce à leurs avantages. Certaines demandes de radio définies par logiciel pourraient être supportées par nouveaux concepts de circuits basés sur des éléments dynamiquement programmables en utilisant des paramètres ajustables. Une petite contribution a été apportée pour améliorer la conception et les simulations des filtres passe-bande Hairpin à large bande. Ces filtres sont compacts et peuvent également être intégrés dans des circuits à micro-ondes compatibles avec un large éventail d'applications dans les systèmes militaires et de communication. Finalement, une proposition a été effectuée visant à équilibrer et améliorer la transmission des signaux discrets de synchronisation entre les TRMs et d'autres unités de traitement dans le satellite SEOSAR/PAZ de dernière génération et permettant l’obtention de la configuration optimale des bus de transmission de données à grande vitesse basés sur un réseau de transcepteurs.

Optical signal impairment study of cascaded optical filters in 40 Gbps DQPSK and 100 Gbps PM-DQPSK systems

Optical filters are crucial elements in optical communications. The influence of cascaded filters in the optical signal will affect the communications quality seriously. In this paper we will study and simulate the optical signal impairment caused by different kinds of filters which include Butterworth, Bessel, Fiber Bragg Grating (FBG) and Fabry-Perot (FP). Optical signal impairment is analyzed from an Eye Opening Penalty (EOP) and optical spectrum point of view. The simulation results show that when the center frequency of all filters aligns with the laser’s frequency, the Butterworth has the smallest influence to the signal while the F-P has the biggest. With a -1dB EOP, the amount of cascaded Butterworth optical filters with a bandwidth of 50 GHz is 18 in 40 Gbps NRZ-DQPSK systems and 12 in 100 Gbps PMNRZ- DQPSK systems. The value is reduced to 9 and 6 respectively for Febry-Perot optical filters. In the situation of frequency misalignment, the impairment caused by filters is more serious. Our research shows that with a frequency deviation of 5 GHz, only 12 and 9 Butterworth optical filters can be cascaded in 40 Gbps NRZ-DQPSK and 100 Gbps PM-NRZ-DQPSK systems respectively. We also study the signal impairment caused by different orders of the Butterworth filter model. Our study shows that although the higher-order has a smaller clipping effect in the transmission spectrum, it will introduce a more serious phase ripple which seriously affects the signal. Simulation result shows that the 2nd order Butterworth filter has the best performance.

Análisis, modelado e igualación no lineal de arrays paramétricos acústicos

La presente Tesis analiza y desarrolla metodología específica que permite la caracterización de sistemas de transmisión acústicos basados en el fenómeno del array paramétrico. Este tipo de estructuras es considerado como uno de los sistemas más representativos de la acústica no lineal con amplias posibilidades tecnológicas. Los arrays paramétricos aprovechan la no linealidad del medio aéreo para obtener en recepción señales en el margen sónico a partir de señales ultrasónicas en emisión. Por desgracia, este procedimiento implica que la señal transmitida y la recibida guardan una relación compleja, que incluye una fuerte ecualización así como una distorsión apreciable por el oyente. Este hecho reduce claramente la posibilidad de obtener sistemas acústicos de gran fidelidad. Hasta ahora, los esfuerzos tecnológicos dirigidos al diseño de sistemas comerciales han tratado de paliar esta falta de fidelidad mediante técnicas de preprocesado fuertemente dependientes de los modelos físicos teóricos. Estos están basados en la ecuación de propagación de onda no lineal. En esta Tesis se propone un nuevo enfoque: la obtención de una representación completa del sistema mediante series de Volterra que permita inferir un sistema de compensación computacionalmente ligero y fiable. La dificultad que entraña la correcta extracción de esta representación obliga a desarrollar una metodología completa de identificación adaptada a este tipo de estructuras. Así, a la hora de aplicar métodos de identificación se hace indispensable la determinación de ciertas características iniciales que favorezcan la parametrización del sistema. En esta Tesis se propone una metodología propia que extrae estas condiciones iniciales. Con estos datos, nos encontramos en disposición de plantear un sistema completo de identificación no lineal basado en señales pseudoaleatorias, que aumenta la fiabilidad de la descripción del sistema, posibilitando tanto la inferencia de la estructura basada en bloques subyacente, como el diseño de mecanismos de compensación adecuados. A su vez, en este escenario concreto en el que intervienen procesos de modulación, factores como el punto de trabajo o las características físicas del transductor, hacen inviables los algoritmos de caracterización habituales. Incluyendo el método de identificación propuesto. Con el fin de eliminar esta problemática se propone una serie de nuevos algoritmos de corrección que permiten la aplicación de la caracterización. Las capacidades de estos nuevos algoritmos se pondrán a prueba sobre un prototipo físico, diseñado a tal efecto. Para ello, se propondrán la metodología y los mecanismos de instrumentación necesarios para llevar a cabo el diseño, la identificación del sistema y su posible corrección, todo ello mediante técnicas de procesado digital previas al sistema de transducción. Los algoritmos se evaluarán en términos de error de modelado a partir de la señal de salida del sistema real frente a la salida sintetizada a partir del modelo estimado. Esta estrategia asegura la posibilidad de aplicar técnicas de compensación ya que éstas son sensibles a errores de estima en módulo y fase. La calidad del sistema final se evaluará en términos de fase, coloración y distorsión no lineal mediante un test propuesto a lo largo de este discurso, como paso previo a una futura evaluación subjetiva. ABSTRACT This Thesis presents a specific methodology for the characterization of acoustic transmission systems based on the parametric array phenomenon. These structures are well-known representatives of the nonlinear acoustics field and display large technological opportunities. Parametric arrays exploit the nonlinear behavior of air to obtain sonic signals at the receptors’side, which were generated within the ultrasonic range. The underlying physical process redunds in a complex relationship between the transmitted and received signals. This includes both a strong equalization and an appreciable distortion for a human listener. High fidelity, acoustic equipment based on this phenomenon is therefore difficult to design. Until recently, efforts devoted to this enterprise have focused in fidelity enhancement based on physically-informed, pre-processing schemes. These derive directly from the nonlinear form of the wave equation. However, online limited enhancement has been achieved. In this Thesis we propose a novel approach: the evaluation of a complete representation of the system through its projection onto the Volterra series, which allows the posterior inference of a computationally light and reliable compensation scheme. The main difficulty in the derivation of such representation strives from the need of a complete identification methodology, suitable for this particular type of structures. As an example, whenever identification techniques are involved, we require preliminary estimates on certain parameters that contribute to the correct parameterization of the system. In this Thesis we propose a methodology to derive such initial values from simple measures. Once these information is made available, a complete identification scheme is required for nonlinear systems based on pseudorandom signals. These contribute to the robustness and fidelity of the resulting model, and facilitate both the inference of the underlying structure, which we subdivide into a simple block-oriented construction, and the design of the corresponding compensation structure. In a scenario such as this where frequency modulations occur, one must control exogenous factors such as devices’ operation point and the physical properties of the transducer. These may conflict with the principia behind the standard identification procedures, as it is the case. With this idea in mind, the Thesis includes a series of novel correction algorithms that facilitate the application of the characterization results onto the system compensation. The proposed algorithms are tested on a prototype that was designed and built for this purpose. The methodology and instrumentation required for its design, the identification of the overall acoustic system and its correction are all based on signal processing techniques, focusing on the system front-end, i.e. prior to transduction. Results are evaluated in terms of input-output modelling error, considering a synthetic construction of the system. This criterion ensures that compensation techniques may actually be introduced, since these are highly sensible to estimation errors both on the envelope and the phase of the signals involved. Finally, the quality of the overall system will be evaluated in terms of phase, spectral color and nonlinear distortion; by means of a test protocol specifically devised for this Thesis, as a prior step for a future, subjective quality evaluation.

Cancelación activa del ruido utilizando el KIT TMS320C5515 EZDSP

El control, o cancelación activa de ruido, consiste en la atenuación del ruido presente en un entorno acústico mediante la emisión de una señal igual y en oposición de fase al ruido que se desea atenuar. La suma de ambas señales en el medio acústico produce una cancelación mutua, de forma que el nivel de ruido resultante es mucho menor al inicial. El funcionamiento de estos sistemas se basa en los principios de comportamiento de los fenómenos ondulatorios descubiertos por Augustin-Jean Fresnel, Christiaan Huygens y Thomas Young entre otros. Desde la década de 1930, se han desarrollado prototipos de sistemas de control activo de ruido, aunque estas primeras ideas eran irrealizables en la práctica o requerían de ajustes manuales cada poco tiempo que hacían inviable su uso. En la década de 1970, el investigador estadounidense Bernard Widrow desarrolla la teoría de procesado adaptativo de señales y el algoritmo de mínimos cuadrados LMS. De este modo, es posible implementar filtros digitales cuya respuesta se adapte de forma dinámica a las condiciones variables del entorno. Con la aparición de los procesadores digitales de señal en la década de 1980 y su evolución posterior, se abre la puerta para el desarrollo de sistemas de cancelación activa de ruido basados en procesado de señal digital adaptativo. Hoy en día, existen sistemas de control activo de ruido implementados en automóviles, aviones, auriculares o racks de equipamiento profesional. El control activo de ruido se basa en el algoritmo fxlms, una versión modificada del algoritmo LMS de filtrado adaptativo que permite compensar la respuesta acústica del entorno. De este modo, se puede filtrar una señal de referencia de ruido de forma dinámica para emitir la señal adecuada que produzca la cancelación. Como el espacio de cancelación acústica está limitado a unas dimensiones de la décima parte de la longitud de onda, sólo es viable la reducción de ruido en baja frecuencia. Generalmente se acepta que el límite está en torno a 500 Hz. En frecuencias medias y altas deben emplearse métodos pasivos de acondicionamiento y aislamiento, que ofrecen muy buenos resultados. Este proyecto tiene como objetivo el desarrollo de un sistema de cancelación activa de ruidos de carácter periódico, empleando para ello electrónica de consumo y un kit de desarrollo DSP basado en un procesador de muy bajo coste. Se han desarrollado una serie de módulos de código para el DSP escritos en lenguaje C, que realizan el procesado de señal adecuado a la referencia de ruido. Esta señal procesada, una vez emitida, produce la cancelación acústica. Empleando el código implementado, se han realizado pruebas que generan la señal de ruido que se desea eliminar dentro del propio DSP. Esta señal se emite mediante un altavoz que simula la fuente de ruido a cancelar, y mediante otro altavoz se emite una versión filtrada de la misma empleando el algoritmo fxlms. Se han realizado pruebas con distintas versiones del algoritmo, y se han obtenido atenuaciones de entre 20 y 35 dB medidas en márgenes de frecuencia estrechos alrededor de la frecuencia del generador, y de entre 8 y 15 dB medidas en banda ancha. ABSTRACT. Active noise control consists on attenuating the noise in an acoustic environment by emitting a signal equal but phase opposed to the undesired noise. The sum of both signals results in mutual cancellation, so that the residual noise is much lower than the original. The operation of these systems is based on the behavior principles of wave phenomena discovered by Augustin-Jean Fresnel, Christiaan Huygens and Thomas Young. Since the 1930’s, active noise control system prototypes have been developed, though these first ideas were practically unrealizable or required manual adjustments very often, therefore they were unusable. In the 1970’s, American researcher Bernard Widrow develops the adaptive signal processing theory and the Least Mean Squares algorithm (LMS). Thereby, implementing digital filters whose response adapts dynamically to the variable environment conditions, becomes possible. With the emergence of digital signal processors in the 1980’s and their later evolution, active noise cancellation systems based on adaptive signal processing are attained. Nowadays active noise control systems have been successfully implemented on automobiles, planes, headphones or racks for professional equipment. Active noise control is based on the fxlms algorithm, which is actually a modified version of the LMS adaptive filtering algorithm that allows compensation for the acoustic response of the environment. Therefore it is possible to dynamically filter a noise reference signal to obtain the appropriate cancelling signal. As the noise cancellation space is limited to approximately one tenth of the wavelength, noise attenuation is only viable for low frequencies. It is commonly accepted the limit of 500 Hz. For mid and high frequencies, conditioning and isolating passive techniques must be used, as they produce very good results. The objective of this project is to develop a noise cancellation system for periodic noise, by using consumer electronics and a DSP development kit based on a very-low-cost processor. Several C coded modules have been developed for the DSP, implementing the appropriate signal processing to the noise reference. This processed signal, once emitted, results in noise cancellation. The developed code has been tested by generating the undesired noise signal in the DSP. This signal is emitted through a speaker simulating the noise source to be removed, and another speaker emits an fxlms filtered version of the same signal. Several versions of the algorithm have been tested, obtaining attenuation levels around 20 – 35 dB measured in a tight bandwidth around the generator frequency, or around 8 – 15 dB measured in broadband.

What does voice-processing technology support today?

This paper describes the state of the art in applications of voice-processing technologies. In the first part, technologies concerning the implementation of speech recognition and synthesis algorithms are described. Hardware technologies such as microprocessors and DSPs (digital signal processors) are discussed. Software development environment, which is a key technology in developing applications software, ranging from DSP software to support software also is described. In the second part, the state of the art of algorithms from the standpoint of applications is discussed. Several issues concerning evaluation of speech recognition/synthesis algorithms are covered, as well as issues concerning the robustness of algorithms in adverse conditions.

Classificação das amostras do ensaio de Baumann através do processamento digital de imagens.

O presente trabalho apresenta uma alternativa ao processo de classificação do defeito da segregação central em amostras de aço, utilizando as imagens digitais que são geradas durante o ensaio de Baumann. O algoritmo proposto tem como objetivo agregar as técnicas de processamento digital de imagens e o conhecimento dos especialistas sobre o defeito da segregação central, visando a classificação do defeito de referência. O algoritmo implementado inclui a identificação e a segmentação da linha segregada por meio da aplicação da transformada de Hough e limiar adaptativo. Adicionalmente, o algoritmo apresenta uma proposta para o mapeamento dos atributos da segregação central nos diferentes graus de severidade do defeito, em função dos critérios de continuidade e intensidade. O mapeamento foi realizado por meio da análise das características individuais, como comprimento, largura e área, dos elementos segmentados que compõem a linha segregada. A avaliação do desempenho do algoritmo foi realizada em dois momentos específicos, de acordo com sua fase de implementação. Para a realização da avaliação, foram analisadas 255 imagens de amostras reais, oriundas de duas usinas siderúrgicas, distribuídas nos diferentes graus de severidade. Os resultados da primeira fase de implementação mostram que a identificação da linha segregada apresenta acurácia de 93%. As classificações oriundas do mapeamento realizado para as classes de criticidade do defeito, na segunda fase de implementação, apresentam acurácia de 92% para o critério de continuidade e 68% para o critério de intensidade.

The Riesz transform and simultaneous representations of phase, energy and orientation in spatial vision

To represent the local orientation and energy of a 1-D image signal, many models of early visual processing employ bandpass quadrature filters, formed by combining the original signal with its Hilbert transform. However, representations capable of estimating an image signal's 2-D phase have been largely ignored. Here, we consider 2-D phase representations using a method based upon the Riesz transform. For spatial images there exist two Riesz transformed signals and one original signal from which orientation, phase and energy may be represented as a vector in 3-D signal space. We show that these image properties may be represented by a Singular Value Decomposition (SVD) of the higher-order derivatives of the original and the Riesz transformed signals. We further show that the expected responses of even and odd symmetric filters from the Riesz transform may be represented by a single signal autocorrelation function, which is beneficial in simplifying Bayesian computations for spatial orientation. Importantly, the Riesz transform allows one to weight linearly across orientation using both symmetric and asymmetric filters to account for some perceptual phase distortions observed in image signals - notably one's perception of edge structure within plaid patterns whose component gratings are either equal or unequal in contrast. Finally, exploiting the benefits that arise from the Riesz definition of local energy as a scalar quantity, we demonstrate the utility of Riesz signal representations in estimating the spatial orientation of second-order image signals. We conclude that the Riesz transform may be employed as a general tool for 2-D visual pattern recognition by its virtue of representing phase, orientation and energy as orthogonal signal quantities.

Towards a unified understanding of event-related changes in the EEG:the Firefly model of synchronization through cross-frequency phase modulation

Although event-related potentials (ERPs) are widely used to study sensory, perceptual and cognitive processes, it remains unknown whether they are phase-locked signals superimposed upon the ongoing electroencephalogram (EEG) or result from phase-alignment of the EEG. Previous attempts to discriminate between these hypotheses have been unsuccessful but here a new test is presented based on the prediction that ERPs generated by phase-alignment will be associated with event-related changes in frequency whereas evoked-ERPs will not. Using empirical mode decomposition (EMD), which allows measurement of narrow-band changes in the EEG without predefining frequency bands, evidence was found for transient frequency slowing in recognition memory ERPs but not in simulated data derived from the evoked model. Furthermore, the timing of phase-alignment was frequency dependent with the earliest alignment occurring at high frequencies. Based on these findings, the Firefly model was developed, which proposes that both evoked and induced power changes derive from frequency-dependent phase-alignment of the ongoing EEG. Simulated data derived from the Firefly model provided a close match with empirical data and the model was able to account for i) the shape and timing of ERPs at different scalp sites, ii) the event-related desynchronization in alpha and synchronization in theta, and iii) changes in the power density spectrum from the pre-stimulus baseline to the post-stimulus period. The Firefly Model, therefore, provides not only a unifying account of event-related changes in the EEG but also a possible mechanism for cross-frequency information processing.

Multi-wavelength regeneration of phase encoded signals based on phase sensitive amplifiers

Future high capacity optical links will have to make use of frequent signal regeneration to enable long distance transmission. In this respect, the role of all-optical signal processing becomes increasingly important because of its potential to mitigate signal impairments at low cost and power consumption. More substantial benefits are expected if regeneration is achieved simultaneously on a multiple signal band. Until recently, this had been achieved only for on-off keying modulation formats. However, as in future transmission links the information will be encoded also in the phase for enhancing the spectral efficiency, novel subsystem concepts will be needed for multichannel processing of such advanced signal formats. In this paper we show that phase sensitive amplifiers can be an ideal technology platform for developing such regenerators and we discuss our recent demonstration of the first multi-channel regenerator for phase encoded signals.

High stability multiplexed fiber interferometer and its application on absolute displacement measurement and on-line surface metrology

We propose a self-reference multiplexed fibre interferometer (MFI) by using a tunable laser and fibre Bragg grating (FBG). The optical measurement system multiplexes two Michelson fibre interferometers with shared optical path in the main part of optical system. One fibre optic interferometer is used as a reference interferometer to monitor and control the high accuracy of the measurement system under environmental perturbations. The other is used as a measurement interferometer to obtain information from the target. An active phase tracking homodyne (APTH) technique is applied for signal processing to achieve high resolution. MFI can be utilised for high precision absolute displacement measurement with different combination of wavelengths from the tuneable laser. By means of Wavelength-Division-Multiplexing (WDM) technique, MFI is also capable of realising on-line surface measurement, in which traditional stylus scanning is replaced by spatial light-wave scanning so as to greatly improve the measurement speed and robustness.

Digital fiber nonlinearity compensation:toward 1-Tb/s transport

The world is connected by a core network of long-haul optical communication systems that link countries and continents, enabling long-distance phone calls, data-center communications, and the Internet. The demands on information rates have been constantly driven up by applications such as online gaming, high-definition video, and cloud computing. All over the world, end-user connection speeds are being increased by replacing conventional digital subscriber line (DSL) and asymmetric DSL (ADSL) with fiber to the home. Clearly, the capacity of the core network must also increase proportionally. © 1991-2012 IEEE.

Shannon capacity of nonlinear communication channels

The exponentially increasing demand on operational data rate has been met with technological advances in telecommunication systems such as advanced multilevel and multidimensional modulation formats, fast signal processing, and research into new different media for signal transmission. Since the current communication channels are essentially nonlinear, estimation of the Shannon capacity for modern nonlinear communication channels is required. This PhD research project has targeted the study of the capacity limits of different nonlinear communication channels with a view to enable a significant enhancement in the data rate of the currently deployed fiber networks. In the current study, a theoretical framework for calculating the Shannon capacity of nonlinear regenerative channels has been developed and illustrated on the example of the proposed here regenerative Fourier transform (RFT). Moreover, the maximum gain in Shannon capacity due to regeneration (that is, the Shannon capacity of a system with ideal regenerators – the upper bound on capacity for all regenerative schemes) is calculated analytically. Thus, we derived a regenerative limit to which the capacity of any regenerative system can be compared, as analogue of the seminal linear Shannon limit. A general optimization scheme (regenerative mapping) has been introduced and demonstrated on systems with different regenerative elements: phase sensitive amplifiers and the proposed here multilevel regenerative schemes: the regenerative Fourier transform and the coupled nonlinear loop mirror.

High stability multiplexed fibre interferometer and its application on absolute displacement measurement and on-line surface metrology

We propose a self-reference multiplexed fibre interferometer (MFI) by using a tunable laser and fibre Bragg grating (FBG). The optical measurement system multiplexes two Michelson fibre interferometers with shared optical path in the main part of optical system. One fibre optic interferometer is used as a reference interferometer to monitor and control the high accuracy of the measurement system under environmental perturbations. The other is used as a measurement interferometer to obtain information from the target. An active phase tracking homodyne (APTH) technique is applied for signal processing to achieve high resolution. MFI can be utilised for high precision absolute displacement measurement with different combination of wavelengths from the tuneable laser. By means of Wavelength-Division-Multiplexing (WDM) technique, MFI is also capable of realising on-line surface measurement, in which traditional stylus scanning is replaced by spatial light-wave scanning so as to greatly improve the measurement speed and robustness. © 2004 Optical Society of America.

Multi-wavelength regeneration of phase encoded signals based on phase sensitive amplifiers

Future high capacity optical links will have to make use of frequent signal regeneration to enable long distance transmission. In this respect, the role of all-optical signal processing becomes increasingly important because of its potential to mitigate signal impairments at low cost and power consumption. More substantial benefits are expected if regeneration is achieved simultaneously on a multiple signal band. Until recently, this had been achieved only for on-off keying modulation formats. However, as in future transmission links the information will be encoded also in the phase for enhancing the spectral efficiency, novel subsystem concepts will be needed for multichannel processing of such advanced signal formats. In this paper we show that phase sensitive amplifiers can be an ideal technology platform for developing such regenerators and we discuss our recent demonstration of the first multi-channel regenerator for phase encoded signals.

The impact of parametric noise amplification on long haul transmission throughput

Advanced signal processing, such as multi-channel digital back propagation and mid span optical phase conjugation, can compensate for inter channel nonlinear effects in point to point links. However, once such are effects are compensated, the interaction between the signal and noise fields becomes dominant. We will show that this interaction has a direct impact on the signal to noise ratio improvement, observing that ideal optical phase conjugation offers 1.5 dB more performance benefit than DSP based compensation.