Bloche-Maxwell treatment of amplification of high harmonic seed in soft x-ray laser amplifiers in both direct and chirped amplifications

Seeding plasma-based softx-raylaser (SXRL) demonstrated diffraction-limited, fully coherent in space and in time beam but with energy not exceeding 1 μJ per pulse. Quasi-steady-state (QSS) plasmas demonstrated to be able to store high amount of energy and then amplify incoherent SXRL up to several mJ. Using 1D time-dependant Bloch–Maxwell model including amplification of noise, we demonstrated that femtosecond HHG cannot be efficiently amplified in QSS plasmas. However, using Chirped Pulse Amplification concept on HHG seed allows to extract most of the stored energy, reaching up to 5 mJ in fully coherent pulses that can be compressed down to 130 fs.

Noise Figure Measurement of Differential Amplifiers Using non-Ideal Baluns

This paper analyzes the noise and gain measurement of microwave differential amplifiers using two passive baluns. A general model of the baluns is considered, including potential losses and phase/amplitude unbalances. This analysis allows de-embedding the actual gain and noise performance of the isolated amplifier by using single-ended measurements of the cascaded system and baluns. Finally, measured results from two amplifier prototypes are used to validate the theoretical principles.

Design methodology in multiphase buck converter based on minimum time control for high efficiency RF amplifiers

This paper proposes an interleaved multiphase buck converter with minimum time control strategy for envelope amplifiers in high efficiency RF power amplifiers. The solution of the envelope amplifier is to combine the proposed converter with a linear regulator in series. High system efficiency can be obtained through modulating the supply voltage of the envelope amplifier with the fast output voltage variation of the converter working with several particular duty cycles that achieve total ripple cancellation. The transient model for minimum time control is explained, and the calculation of transient times that are pre-calculated and inserted into a look-up table is presented. The filter design trade-off that limits capability of envelope modulation is also discussed. The experimental results verify the fast voltage transient obtained with a 4-phase buck prototype.

Envelope amplifier based on switching capacitors for high efficiency RF amplifiers

Modern transmitters usually have to amplify and transmit signals with simultaneous envelope and phase modulation. Due to this property of the transmitted signal, linear power amplifiers (class A, B, or AB) are usually used as a solution for the power amplifier stage. These amplifiers have high linearity, but suffer from low efficiency when the transmitted signal has high peak-to-average power ratio. The Kahn envelope elimination and restoration technique is used to enhance the efficiency of RF transmitters, by combining highly efficient, nonlinear RF amplifier (class E) with a highly efficient envelope amplifier in order to obtain a linear and highly efficient RF amplifier. This paper presents a solution for the envelope amplifier based on a multilevel converter in series with a linear regulator. The multilevel converter is implemented by employing voltage dividers based on switching capacitors. The implemented envelope amplifier can reproduce any signal with a maximum spectral component of 2 MHz and give instantaneous maximum power of 50 W. The efficiency measurements show that when the signals with low average value are transmitted, the implemented prototypes have up to 20% higher efficiency than linear regulators used as a conventional solution.

Modeling Reflective Bistability in Vertical-Cavity Semiconductor Optical Amplifiers

The characteristics of optical bistability in a vertical- cavity semiconductor optical amplifier (VCSOA) operated in reflection are reported. The dependences of the optical bistability in VCSOAs on the initial phase detuning and on the applied bias current are analyzed. The optical bistability is also studied for different numbers of superimposed periods in the top distributed bragg reflector (DBR) that conform the internal cavity of the device. The appearance of the X-bistable and the clockwise bistable loops is predicted theoretically in a VCSOA operated in reflection for the first time, to the best of our knowledge. Moreover, it is also predicted that the control of the VCSOA’s top reflectivity by the addition of new superimposed periods in its top DBR reduces by one order of magnitude the input power needed for the assessment of the X- and the clockwise bistable loop, compared to that required in in-plane semiconductor optical amplifiers. These results, added to the ease of fabricating two-dimensional arrays of this kind of device could be useful for the development of new optical logic or optical signal regeneration devices.

Neural Processing Unit based on the Optical Bistable Properties of Semiconductor Laser Amplifiers

This paper analyzes the behavior of a neural processing unit based on the optical bistable properties of semiconductor laser amplifiers. A similar unit to the reported here was previously employed in the simulation of the mammalian retina. The main advantages of the present cell are its larger fan-out and the possibility of different responses according to the light wavelength impinging onto the cell. These properties allow to work with larger structures as well as to obtain different behaviors according to the light characteristics. This new approach gives a possible modeling closer to the real biological configurations. Moreover, a more detailed analysis of the basic cell internal behavior is reported

All-optical logic gates with 1550nm Fabry-Perot and distributed feedback semiconductor laser amplifiers

The optical bistability occurring in laser diode amplifiers is used to design an all-optical logic gate capable to provide the whole set of logic functions. The structure of the reported logic gate is based on two connected 1550nm laser amplifiers (Fabry-Perot and distributed feedback laser amplifiers).

Correlación entre señales de audio multi-microfónicas

Este proyecto pretende mostrar los desfases existentes entre señales de audio obtenidas de la misma fuente en distintos puntos distanciados entre sí. Para ello nos basamos en el análisis de la correlación de las señales de audio multi-microfónicas, para determinar los retrasos entre dichas señales. Durante las de tres partes diferentes que conforman este proyecto, explicaremos el dónde, cómo y por qué se produce este efecto en este tipo de señales. En la primera se presentan algunos de los conceptos teóricos necesarios para entender el desarrollo posterior, tales como la coherencia y correlación entre señales, los retardos de fase y la importancia del micro-tiempo. Además se explican diversas técnicas microfónicas que se utilizarán en la tercera parte. A lo largo de la segunda, se presenta el software desarrollado para determinar y corregir el retraso entre las señales que se deseen analizar. Para ello se ha escogido la herramienta de programación Matlab, ya que ha sido la más utilizada en la mayoría de las asignaturas que componen la titulación y por ello se posee el suficiente dominio de la misma. Además de presentar el propio software, al final de esta parte hay un manual de usuario del mismo, en el que se explica el manejo para posibles usos futuros por parte de otras personas interesadas. En la última parte se demuestra en varios casos reales, el estudio de la alineación de tomas multi-microfónicas en las cuales se produce en efecto que se intenta detectar y corregir. Aquí se realizan tres estudios de dicho fenómeno. En el primero se emplean señales digitales internas, concretamente ruido blanco, retrasando algunas muestras dichas señales unas de otras, para luego analizarlas con el software desarrollado y comprobar la eficacia del mismo. En el segundo se analizan la señales de audio obtenidas en el estudio de grabación de varios grupos de música moderna, mostrando los resultados del empleo del software en algunas de ellas, tales como las tomas de batería, bajo y guitarra. En el tercero se analizan las señales de audio obtenidas fuera del estudio de grabación, en donde no se dispone de las supuestas condiciones ideales que se tienen en el entorno que rodea a un estudio de grabación (acústicamente hablando). Se utilizan algunas de las técnicas microfónicas explicadas en el último apartado de la parte dedicada a los conceptos teóricos, para la grabación de una orquesta sinfónica, para luego analizar el efecto buscado mediante nuestro software, presentando los resultados obtenidos. De igual manera se realiza en el estudio con una agrupación coral de cuatro voces dentro de una Iglesia. ABSTRACT This project aims to show delays between audio signals obtained from the same source at diferent points spaced apart. To do this we rely on the analysis of the correlation of multi-microphonic audio signals, to determine the delay between these signals. During three diferent parts that make up this project, we will explain where, how and why this effect occurs in this type of signals. At the first part we present some of the theoretical concepts necessary to understand the subsequent development, such as coherence and correlation between signals, phase delays and the importance of micro-time. Also explains several microphone techniques to be used in the third part. During the second, it presents the software developed to determine and correct the delay between the signals that are desired to analyze. For this we have chosen the programming software Matlab , as it has been the most used in the majority of the subjects in the degree and therefore has suficient command of it. Besides presenting the software at the end of this part there is a user manual of it , which explains the handling for future use by other interested people. The last part is shown in several real cases, the study of aligning multi- microphonic sockets in which it is produced in effect trying to detect and correct. This includes three studies of this phenomenon. In the first internal digital signals are used, basically white noise, delaying some samples the signals from each other, then with software developed analyzing and verifying its efectiveness. In the second analyzes the audio signals obtained in the recording studio several contemporary bands, showing the results of using the software in some of them, such as the taking of drums, bass and guitar. In the third analyzes audio signals obtained outside the recording studio, where there are no ideal conditions alleged to have on the environment surrounding a recording studio (acoustically speaking). We use some of the microphone techniques explained in the last paragraph of the section on theoretical concepts, for the recording of a symphony orchestra, and then analyze the effect sought by our software, presenting the results. Similarly, in the study performed with a four-voice choir in a church.

Distributed Web Interface for Real-Time Spatial Audio Reproduction System

SSR es el acrónimo de SoundScape Renderer (tool for real-time spatial audio reproduction providing a variety of rendering algorithms), es un programa escrito en su mayoría en C++. El programa permite al usuario escuchar tanto sonidos grabados con anterioridad como sonidos en directo. El sonido o los sonidos se oirán, desde el punto de vista del oyente, como si el sonido se produjese en el punto que el programa decida, lo interesante de este proyecto es que el sonido podrá cambiar de lugar, moverse, etc. Todo en tiempo real. Esto se consigue sin modificar el sonido al grabarlo pero sí al emitirlo, el programa calcula las variaciones necesarias para que al emitir el sonido al oyente le llegue como si el sonido realmente se generase en un punto del espacio o lo más parecido posible. La sensación de movimiento no deja de ser el punto anterior cambiando de lugar. La idea era crear una aplicación web basada en Canvas de HTML5 que se comunicará con esta interfaz de usuario remota. Así se solucionarían todos los problemas de compatibilidad ya que cualquier dispositivo con posibilidad de visualizar páginas web podría correr una aplicación basada en estándares web, por ejemplo un sistema con Windows o un móvil con navegador. El protocolo debía de ser WebSocket porque es un protocolo HTML5 y ofrece las “garantías” de latencia que una aplicación con necesidades de información en tiempo real requiere. Nos permite una comunicación full-dúplex asíncrona sin mucho payload que es justo lo que se venía a evitar al no usar polling normal de HTML. El problema que surgió fue que la interfaz de usuario de red que tenía el programa no era compatible con WebSocket debido a un handshacking inicial y obligatorio que realiza el protocolo, por lo que se necesitaba otra interfaz de red. Se decidió entonces cambiar a JSON como formato para el intercambio de mensajes. Al final el proyecto comprende no sólo la aplicación web basada en Canvas sino también un servidor funcional y la definición de una nueva interfaz de usuario de red con su protocolo añadido. ABSTRACT. This project aims to become a part of the SSR tool to extend its capabilities in the field of the access. SSR is an acronym for SoundScape Renderer, is a program mostly written in C++ that allows you to hear already recorded or live sound with a variety of sound equipment as if the sound came from a desired place in the space. Like the web-page of the SSR says surely better explained: “The SoundScape Renderer (SSR) is a tool for real-time spatial audio reproduction providing a variety of rendering algorithms.” The application can be used with a graphical interface written in Qt but has also a network interface for external applications to use it. This network interface communicates using XML messages. A good example of it is the Android client. This Android client is already working. In order to use the application should be run it by loading an audio source and the wanted environment so that the renderer knows what to do. In that moment the server binds and anyone can use the network interface. Since the network interface is documented everyone can make an application to interact with this network interface. So the application can have as many user interfaces as wanted. The part that is developed in this project has nothing to do neither with audio rendering nor even with the reproduction of the spatial audio. The part that is developed here is about the interface used in the SSR application. As it can be deduced from the title: “Distributed Web Interface for Real-Time Spatial Audio Reproduction System”, this work aims only to offer the interface via web for the SSR (“Real-Time Spatial Audio Reproduction System”). The idea is not to make a new graphical interface for SSR but to allow more types of interfaces and communication. To accomplish the objective of allowing more graphical interfaces this project is going to use a new network interface. By now the SSR application is using only XML for data interchange but this new network interface support JSON. This project comprehends the server that launch the application, the user interface and the new network interface. It is done with these modules in order to allow creating new user interfaces that can communicate with the server or new servers that can communicate with the user interface by defining a complete network interface for data interchange.

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

Vertical-cavity semiconductor optical amplifiers (VCSOAs) as optical sensing elements

Semiconductor Optical Amplifiers (SOAs) have mainly found application in optical telecommunication networks for optical signal regeneration, wavelength switching or wavelength conversion. The objective of this paper is to report the use of semiconductor optical amplifiers for optical sensing taking into account their optical bistable properties. As it was previously reported, some semiconductor optical amplifiers, including Fabry-Perot and Distributed-Feedback Semiconductor Optical Amplifiers (FPSOAs and DFBSOAs), may exhibit optical bistability. The characteristics of the attained optical bistability in this kind of devices are strongly dependent on different parameters including wavelength, temperature or applied bias current and small variations lead to a change on their bistable properties. As in previous analyses for Fabry-Perot and DFB SOAs, the variations of these parameters and their possible application for optical sensing are reported in this paper for the case of the Vertical-Cavity Semiconductor Optical Amplifier (VCSOA). When using a VCSOA, the input power needed for the appearance of optical bistability is one order of magnitude lower than that needed in edge-emitting devices. This feature, added to the low manufacturing costs of VCSOAs and the ease to integrate them in 2-D arrays, makes the VCSOA a very promising device for its potential use in optical sensing applications.

Unidad de efectos de audio en Simulink

La tecnología moderna de computación ha permitido cambiar radicalmente la investigación tecnológica en todos los ámbitos. El proceso general utilizado previamente consistía en el desarrollo de prototipos analógicos, creando múltiples versiones del mismo hasta llegar al resultado adecuado. Este es un proceso costoso a nivel económico y de carga de trabajo. Es por ello por lo que el proceso de investigación actual aprovecha las nuevas tecnologías para lograr el objetivo final mediante la simulación. Gracias al desarrollo de software para la simulación de distintas áreas se ha incrementado el ritmo de crecimiento de los avances tecnológicos y reducido el coste de los proyectos en investigación y desarrollo. La simulación, por tanto, permite desarrollar previamente prototipos simulados con un coste mucho menor para así lograr un producto final, el cual será llevado a cabo en su ámbito correspondiente. Este proceso no sólo se aplica en el caso de productos con circuitería, si bien es utilizado también en productos programados. Muchos de los programas actuales trabajan con algoritmos concretos cuyo funcionamiento debe ser comprobado previamente, para después centrarse en la codificación del mismo. Es en este punto donde se encuentra el objetivo de este proyecto, simular algoritmos de procesado digital de la señal antes de la codificación del programa final. Los sistemas de audio están basados en su totalidad en algoritmos de procesado de la señal, tanto analógicos como digitales, siendo estos últimos los que están sustituyendo al mundo analógico mediante los procesadores y los ordenadores. Estos algoritmos son la parte más compleja del sistema, y es la creación de nuevos algoritmos la base para lograr sistemas de audio novedosos y funcionales. Se debe destacar que los grupos de desarrollo de sistemas de audio presentan un amplio número de miembros con cometidos diferentes, separando las funciones de programadores e ingenieros de la señal de audio. Es por ello por lo que la simulación de estos algoritmos es fundamental a la hora de desarrollar nuevos y más potentes sistemas de audio. Matlab es una de las herramientas fundamentales para la simulación por ordenador, la cual presenta utilidades para desarrollar proyectos en distintos ámbitos. Sin embargo, en creciente uso actualmente se encuentra el software Simulink, herramienta especializada en la simulación de alto nivel que simplifica la dificultad de la programación en Matlab y permite desarrollar modelos de forma más rápida. Simulink presenta una completa funcionalidad para el desarrollo de algoritmos de procesado digital de audio. Por ello, el objetivo de este proyecto es el estudio de las capacidades de Simulink para generar sistemas de audio funcionales. A su vez, este proyecto pretende profundizar en los métodos de procesado digital de la señal de audio, logrando al final un paquete de sistemas de audio compatible con los programas de edición de audio actuales. ABSTRACT. Modern computer technology has dramatically changed the technological research in multiple areas. The overall process previously used consisted of the development of analog prototypes, creating multiple versions to reach the proper result. This is an expensive process in terms of an economically level and workload. For this reason actual investigation process take advantage of the new technologies to achieve the final objective through simulation. Thanks to the software development for simulation in different areas the growth rate of technological progress has been increased and the cost of research and development projects has been decreased. Hence, simulation allows previously the development of simulated protoypes with a much lower cost to obtain a final product, which will be held in its respective field. This process is not only applied in the case of circuitry products, but is also used in programmed products. Many current programs work with specific algorithms whose performance should be tested beforehand, which allows focusing on the codification of the program. This is the main point of this project, to simulate digital signal processing algorithms before the codification of the final program. Audio systems are entirely based on signal processing, both analog and digital systems, being the digital systems which are replacing the analog world thanks to the processors and computers. This algorithms are the most complex part of every system, and the creation of new algorithms is the most important step to achieve innovative and functional new audio systems. It should be noted that development groups of audio systems have a large number of members with different roles, separating them into programmers and audio signal engineers. For this reason, the simulation of this algorithms is essential when developing new and more powerful audio systems. Matlab is one of the most important tools for computer simulation, which has utilities to develop projects in different areas. However, the use of the Simulink software is constantly growing. It is a simulation tool specialized in high-level simulations which simplifies the difficulty of programming in Matlab and allows the developing of models faster. Simulink presents a full functionality for the development of algorithms for digital audio processing. Therefore, the objective of this project is to study the posibilities of Simulink to generate funcional audio systems. In turn, this projects aims to get deeper into the methods of digital audio signal processing, making at the end a software package of audio systems compatible with the current audio editing software.

Design of a Two-Phase buck converter with fourth-order output filter for envelope amplifiers of limited bandwidth

The use of techniques such as envelope tracking (ET) and envelope elimination and restoration (EER) can improve the efficiency of radio frequency power amplifiers (RFPA). In both cases, high-bandwidth DC/DC converters called envelope amplifiers (EA) are used to modulate the supply voltage of the RFPA. This paper addresses the analysis and design of a modified two-phase Buck converter optimized to operate as EA. The effects of multiphase operation on the tracking capabilities are analyzed. The use of a fourth-order output filter is proposed to increase the attenuation of the harmonics generated by the PWM operation, thus allowing a reduction of the ratio between the switching frequency and the converter bandwidth. The design of the output filter is addressed considering envelope tracking accuracy and distortion caused by the side bands arising from the nonlinear modulation process. Finally, the proposed analysis and design methods are supported by simulation results, as well as demonstrated by experiments obtained using two 100-W, 10-MHz, two-phase Buck EAs capable of accurately tracking a 1.5-MHz bandwidth OFDM signal.

Desarrollo de una biblioteca de efectos digitales de audio para ser usada en un laboratorio docente

Desarrollo de una librería de efectos de audio en lenguaje nativo de Matlab con procesamiento a tiempo no real. Incluye una interfaz de usuario sencilla y auto explicativa, y ofrece un control libre de los parámetros del efecto, elección y visualización del audio de entrada, reproducción y visualización del audio de salida, y representación característica del procesamiento que se está realizando. El objetivo principal de la librería es que sea usada por alumnos en un laboratorio docente, permitiendo la experimentación con diversos parámetros y entradas de audio facilitando, de esta forma, la comprensión de los diferentes procesamientos que se están realizando. El proyecto incluye una extensa documentación y una plantilla con el objetivo de que se puedan añadir en un futuro más programas de efectos, puesto que la intención del proyecto es ofrecer una librería a largo plazo y facilitar el mantenimiento y las modificaciones futuras.

Analysis of the performance of tapered semiconductor optical amplifiers: role of the taper angle

The beam properties of tapered semiconductor optical amplifiers emitting at 1.57 μm are analyzed by means of simulations with a self-consistent steady state electro-optical and thermal simulator. The results indicate that the self-focusing caused by carrier lensing is delayed to higher currents for devices with taper angle slightly higher than the free diffraction angle.