Integrated chaos generators

This paper surveys the different design issues, from mathematical model to silicon, involved on the design of integrated circuits for the generation of chaotic behavior.Comisión Interministerial de Ciencia y Tecnología 1FD97-1611(TIC)European Commission ESPRIT 3110

Analysis and design of ΣΔ Modulators for Radio Frequency Switchmode Power Amplifiers

Power amplifiers are an integral part of every basestation, macrocell, microcell and mobile phone, enabling data to be sent over the distances needed to reach the receiver’s antenna. While linear operation is needed for transmitting WCDMA and OFDM signals, linear operation of a power amplifier is characterized by low power efficiency, and contributes to unwanted power dissipation in a transmitter. Recently, a switchmode power amplifier operation was considered for reducing power losses in a RF transmitter. A linear and efficient operation of a PA can be achieved when the transmitted RF signal is ΣΔ modu- lated, and subsequently amplified by a nonlinear device. Although in theory this approach offers linearity and efficiency reaching 100%, the use of ΣΔ modulation for transmitting wideband signals causes problems in practical implementation: it requires high sampling rate by the digital hardware, which is needed for shaping large contents of a quantization noise induced by the modulator but also, the binary output from the modulator needs an RF power amplifier operating over very wide frequency band. This thesis addresses the problem of noise shaping in a ΣΔ modulator and nonlinear distortion caused by broadband operation in switchmode power amplifier driven by a ΣΔ modulated waveform. The problem of sampling rate increase in a ΣΔ modulator is solved by optimizing structure of the modulator, and subsequent processing of an input signal’s samples in parallel. Independent from the above, a novel technique for reducing quan- tization noise in a bandpass ΣΔ modulator using single bit quantizer is presented. The technique combines error pulse shaping and 3-level quantization for improving signal to noise ratio in a 2-level output. The improvement is achieved without the increase of a digital hardware’s sampling rate, which is advantageous also from the perspective of power consumption. The new method is explored in the course of analysis, and verified by simulated and experimental results. The process of RF signal conversion from the Cartesian to polar form is analyzed, and a signal modulator for a polar transmitter with a ΣΔ-digitized envelope signal is designed and implemented. The new modulator takes an advantage of bandpass digital to analog conversion for simplifying the analog part of the modulator. A deformation of the pulsed RF signal in the experimental modulator is demonstrated to have an effect primarily on amplitude of the RF signal, which is correctable with simple predistortion

SPS pilot signal design and power transponder analysis, volume 2, phase 3

The problem of pilot signal parameter optimization and the related problem of power transponder performance analysis for the Solar Power Satellite reference phase control system are addressed. Signal and interference models were established to enable specifications of the front end filters including both the notch filter and the antenna frequency response. A simulation program package was developed to be included in SOLARSIM to perform tradeoffs of system parameters based on minimizing the phase error for the pilot phase extraction. An analytical model that characterizes the overall power transponder operation was developed. From this model, the effects of different phase noise disturbance sources that contribute to phase variations at the output of the power transponders were studied and quantified. Results indicate that it is feasible to hold the antenna array phase error to less than one degree per power module for the type of disturbances modeled

Data transmission through channels pertubed by impulsive noise

Imperial Users onl

TECHNOLOGIES AND METRICS FOR METROLOGICAL CHARACTERISTICS IMPROVEMENT OF LOW-COST DEVICES FOR SMART GRIDS MEASUREMENTS

In recent years, rapid changes have occurred in electric grids. This includes the significant integration of renewable energy sources into the grid, alongside the liberalization of the electricity market. Additionally, there has been a rise in the number of "prosumers" – users who transit from passive consumers to active producers, generating their own energy, primarily through domestic photovoltaic systems. Although mini and micro wind generators are less common, they also contribute to this trend. These transformations have not only altered the traditional concept of the grid, which previously involved a unidirectional power flow from generation sites to users, but have also transformed it from a passive into an active and intelligent network – known as a smart grid.In smart grids, the measurement of voltage and current parameters, as well as the amount of energy exchanged between users and producers, takes on a fundamental role. For example, the load flow problem, that needs to know the active and reactive power in the different nodes of the network, for a correct application of dispatching, or the frequency and voltage regulation, key features to provide an efficient energy service. The increased complexity of the electricity grid, combined with the new needs related to remote control, communication and interconnection, has made it necessary to use distributed measurement systems, which require a greater number of instruments capable of carrying out different activities. For this reason, the possibility of using low-cost devices for measurements, and in particular for power quality measurements, has become increasingly important. The use of programmable devices also allows great flexibility in adapting to the required measures. One of the main aspects taken into account in the management of the network is “power quality”, a term that includes the main characteristics of voltage and current, and that requires continuous and different measurement techniques. Among the various aspects related to power quality, the evaluation of the harmonic content of the voltage is one of those that attracts the greatest interest both from users, who are increasingly informed and attentive to the management and economic aspects of domestic utilities, that by energy producers and managers, in order to maximize profits. The Ph.D. project involved the participation and financial support of the University of Palermo, the National Research Council (CNR) and the company STMicroelectronics. The goal of this thesis is to investigate the feasibility of the development of a measuring instrument, based on a low-cost microcontroller and AFE device, able to perform a harmonic analysis of the power system voltage respecting the requirements of the standard (to reach the highest accuracy class).Specifically, the project started with standard review to establish the requirements from current legislation, for a device within the highest accuracy class. Different technological solutions and metrics have been developed and compared in order to ensure compliance with the standard while using low-level hardware, optimizing the software component. For the validation, different tests have been conducted using an existing microcontroller board, with typical specifications of a low-cost device.This thesis is the conclusion of this project, in which this possibility was evaluated defining which specifications this device should have.The thesis consists of five chapters: in the first one, an in-depth analysis of power quality measurements for smart grids is reported, with a focus on current standards (in particular the IEC 61000-4-7 and 61000-4-30). In the second chapter the studies concerning the techniques of harmonic analysis and the aspects related to the possibility of realizing a harmonics measurement instrument using a low-cost device, with all the consequent difficulties, are reported. The third chapter includes simulation tests to assess the feasibility of implementing the necessary metrics, in order to narrow the field to workable solutions and to be able to define some minimum specifications. The simulated metrics were then implemented on a real device, which was appropriately characterized, and tested under different conditions, described in chapter four. The last chapter contains the conclusions

The development of speech coding and the first standard coder for public mobile telephony

This thesis describes in its core chapter (Chapter 4) the original algorithmic and design features of the ??rst coder for public mobile telephony, the GSM full-rate speech coder, as standardized in 1988. It has never been described in so much detail as presented here. The coder is put in a historical perspective by two preceding chapters on the history of speech production models and the development of speech coding techniques until the mid 1980s, respectively. In the epilogue a brief review is given of later developments in speech coding. The introductory Chapter 1 starts with some preliminaries. It is de- ??ned what speech coding is and the reader is introduced to speech coding standards and the standardization institutes which set them. Then, the attributes of a speech coder playing a role in standardization are explained. Subsequently, several applications of speech coders - including mobile telephony - will be discussed and the state of the art in speech coding will be illustrated on the basis of some worldwide recognized standards. Chapter 2 starts with a summary of the features of speech signals and their source, the human speech organ. Then, historical models of speech production which form the basis of di??erent kinds of modern speech coders are discussed. Starting with a review of ancient mechanical models, we will arrive at the electrical source-??lter model of the 1930s. Subsequently, the acoustic-tube models as they arose in the 1950s and 1960s are discussed. Finally the 1970s are reviewed which brought the discrete-time ??lter model on the basis of linear prediction. In a unique way the logical sequencing of these models is exposed, and the links are discussed. Whereas the historical models are discussed in a narrative style, the acoustic tube models and the linear prediction tech nique as applied to speech, are subject to more mathematical analysis in order to create a sound basis for the treatise of Chapter 4. This trend continues in Chapter 3, whenever instrumental in completing that basis. In Chapter 3 the reader is taken by the hand on a guided tour through time during which successive speech coding methods pass in review. In an original way special attention is paid to the evolutionary aspect. Speci??cally, for each newly proposed method it is discussed what it added to the known techniques of the time. After presenting the relevant predecessors starting with Pulse Code Modulation (PCM) and the early vocoders of the 1930s, we will arrive at Residual-Excited Linear Predictive (RELP) coders, Analysis-by-Synthesis systems and Regular- Pulse Excitation in 1984. The latter forms the basis of the GSM full-rate coder. In Chapter 4, which constitutes the core of this thesis, explicit forms of Multi-Pulse Excited (MPE) and Regular-Pulse Excited (RPE) analysis-by-synthesis coding systems are developed. Starting from current pulse-amplitude computation methods in 1984, which included solving sets of equations (typically of order 10-16) two hundred times a second, several explicit-form designs are considered by which solving sets of equations in real time is avoided. Then, the design of a speci??c explicitform RPE coder and an associated eÆcient architecture are described. The explicit forms and the resulting architectural features have never been published in so much detail as presented here. Implementation of such a codec enabled real-time operation on a state-of-the-art singlechip digital signal processor of the time. This coder, at a bit rate of 13 kbit/s, has been selected as the Full-Rate GSM standard in 1988. Its performance is recapitulated. Chapter 5 is an epilogue brie y reviewing the major developments in speech coding technology after 1988. Many speech coding standards have been set, for mobile telephony as well as for other applications, since then. The chapter is concluded by an outlook

Design and Implementation of a Re-Configurable Arbitrary Signal Generator and Radio Frequency Spectrum Analyser

This research is focused on the design, simulation and implementation of a reconfigurable arbitrary signal generator and the design, simulation and implementation of a radio frequency spectrum analyser based on digital signal processing. Until recently, Application Specific Integrated Circuits (ASICs) were used to produce high performance re-configurable function and arbitrary waveform generators with comprehensive modulation capabilities. However, that situation is now changing with the availability of advanced but low cost Field Programmable Gate Arrays (FPGAs), which could be used as an alternative to ASICs in these applications. The availability of high performance FPGA families opens up the opportunity to compete with ASIC solutions at a fraction of the development cost of an ASIC solution. A fast digital signal processing algorithm for digital waveform generation, using primarily but not limited to Direct Digital Synthesis (DDS) technologies, developed and implemented in a field-configurable logic, with control provided by an embedded microprocessor replacing a high cost ASIC design appeared to be a very attractive concept. This research demonstrates that such a concept is feasible in its entirety. A fully functional, low-complexity, low cost, pulse, Gaussian white noise and DDS based function and arbitrary waveform generator, capable of being amplitude, frequency and phase modulated by an internally generated or external modulating signal was implemented in a low-cost FPGA. The FPGA also included the capabilities to perform pulse width modulation and pulse delay modulation on pulse waveforms. Algorithms to up-convert the sampling rate of the external modulating signal using Cascaded Integrator Comb (CIC) filters and using interpolation method were analysed. Both solutions were implemented to compare their hardware complexities. Analysis of generating noise with user-defined distribution is presented. The ability of triggering the generator by an internally generated or an external event to generate a burst of waveforms where the time between the trigger signal and waveform output is fixed was also implemented in the FPGA. Finally, design of interface to a microprocessor to provide control of the versatile waveform generator was also included in the FPGA. This thesis summarises the literature, design considerations, simulation and implementation of the generator design. The second part of the research is focused on radio frequency spectrum analysis based on digital signal processing. Most existing spectrum analysers are analogue in nature and their complexity increases with frequency. Therefore, the possibility of using digital techniques for spectrum analysis was considered. The aim was to come up with digital system architecture for spectrum analysis and to develop and implement the new approach on a suitable digital platform. This thesis analyses the current literature on shifting algorithms to remove spurious responses and highlights its drawbacks. This thesis also analyses existing literature on quadrature receivers and presents novel adaptation of the existing architectures for application in spectrum analysis. A wide band spectrum analyser receiver with compensation for gain and phase imbalances in the Radio Frequency (RF) input range, as well as compensation for gain and phase imbalances within the Intermediate Frequency (IF) pass band complete with Resolution Band Width (RBW) filtering, Video Band Width (VBW) filtering and amplitude detection was implemented in a low cost FPGA. The ability to extract the modulating signal from a frequency or amplitude modulated RF signal was also implemented. The same family of FPGA used in the generator design was chosen to be the digital platform for this design. This research makes arguments for the new architecture and then summarises the literature, design considerations, simulation and implementation of the new digital algorithm for the radio frequency spectrum analyser

Error modeling, self-calibration and design of pipelined analog to digital converters

Typescript (photocopy).As the field of signal processing accelerates toward the use of high performance digital techniques, there is a growing need for increasingly fast and accurate analog to digital converters. Three highly visible examples of this trend originated in the last decade. The advent of the compact disc revolutionized the way high-fidelity audio is stored, reproduced, recorded and processed. Digital communication links, fiber optic cables and in the near future ISDN networks (Integrated Services Digital Network) are steadily replacing major portions of telephone systems. Finally, video-conferencing, multi-media computing and currently emerging high definition television (HDTV) systems rely more and more on real-time digital data compression and image enhancing techniques. All these applications rely on analog to digital conversion. In the field of digital audio, the required conversion accuracy is high, but the conversion speed limited (16 bits, 2 x 20 kHz signal bandwidth). In the field of image processing, the required accuracy is less, but the data conversion speed high (8-10 bits, 5-20MHz bandwidth). New applications keep pushing for increasing conversion rates and simultaneously higher accuracies. This dissertation discusses new analog to digital converter architectures that could accomplish this. As a consequence of the trend towards digital processing, prominent analog designers throughout the world have engaged in very active research on the topic of data conversion. Unfortunately, literature has not always kept up. At the time of this writing, it seemed rather difficult to find detailed fundamental publications about analog to digital converter design. This dissertation represents a modest attempt to remedy this situation. It is hoped that anyone with a back-ground in analog design could go through this work and pick up the fundamentals of converter operation, as well as a number of more advanced design techniques