This thesis describes the theoretical development, design, and implementation of a novel measurement system, called a Sampling Swept Time Delay Short Pulse (SSTDSP) wireless channel sounder, capable of real time in field performance characterization of high speed fixed wireless links. The SSTDSP sounder has been designed to provide vital performance metrics for fixed point high data rate applications in the 28 GHz LMDS band at a fraction of the cost and complexity of existing wideband channel sounders. The SSTDSP sounder monitors the behavior of the LMDS channel by sampling the impulse response of the channel in real time. This digitized impulse response is used to assemble a power delay profile and render real-time channel performance metrics such as the mean excess delay, RMS delay spread, maximum excess delay for a given multipath threshold, and coherence bandwidth. The SSTDSP sounder is capable of recording these metrics through three modes of operation - continuous channel monitoring, single instant channel snapshot, or data logging. Swept time delay time dilation processing is combined with precise sample and hold gating to reduce the analog to digital converter sampling rate required to digitize the nanosecond short pulses from 2 Gsps to 1 Msps, while retaining the required effective Nyquist sampling rate of 2 Gsps. This dramatically reduces the memory, digital signal processing, and data logging storage requirements as well as the overall cost of the sounder system. The thesis presents the theory behind channel sounding and discusses whether there is a "bounce path" available to LMDS. Several existing channel sounding methods are compared for this application. A number of specific design and performance criteria from each of these methods are synthesized to produce the Sampling Swept Time Delay Short Pulse Sounder architecture. The design and implementation process used to realize the SSTDSP sounder is presented, including a system overview, module details, and algorithm development details. A calibration and measurement test procedure is outlined and system verification results are presented. Current work in progress on the test platform and future improvements to the modular system are outlined, as well as conclusions and future implications of the system. / Master of Science

International Telemetering Conference Proceedings / October 25-28, 1999 / Riviera Hotel and Convention Center, Las Vegas, Nevada / L-3 Communications Telemetry & Instrumentation (L-3 T&I) has developed an advanced IBM PC-AT Telemetry, Tracking, and Commanding (TT&C) SATCOM system based on the utilization of Field Programmable Gate Array / Digital Signal Processing (FPGA/DSP) microprocessors. This system includes up-link, down-link, and range processing sections. Physically, the system consists of one IF Transceiver and two or more FPGA/DSP microprocessor boards called Advanced Processing Microprocessors (APMs). The form factor of these PWBs is compliant with full length, full height IBM PC PCI bus cards. This paper describes the features and functionality of an advanced Telemetry, Tracking, and Commanding Processing System (TT&CPS) based on the implementation of FPGA and DSP microprocessors. The high-level functional attributes of the TT&CPS are depicted in Figure 1. There are four main functional blocks: the IF Transceiver, the Down-Link Processing Section, the Up-Link Processing Section, and the Range Processor. The analog/IF circuitry in the IF Transceiver card interfaces between the 68–72 MHz (70 MHz, nominal) IF I/O signals and the Up-Link and Down-Link Processing Section's DSP equipment. The down-link portion of the IF Transceiver card has two user-selected input ports. From the selected input, the signal is processed through selectable bandwidth limiting, gain control, Doppler correction (optional), quadrature down-conversion to zero hertz (baseband), selectable baseband filtering, and precision Analog-to-Digital (A/D) conversion. The up-link portion of the IF Transceiver card takes I/Q digital data from the APM performing the up-link processing functions. This baseband I/Q digital data is Digital-to-Analog (D/A) converted, filtered, quadrature up-converted to 68–72 MHz, up-link Doppler corrected (optional), output level detected and level controlled, and sent to a two-position output selector switch. The down-link portion of the TT&CPS provides main carrier linear PM or BPSK or QPSK demodulation and can also, in composite linear PM demodulation mode, receive and demodulate FSK and/or BPSK subcarriers and ranging signals. The demodulators use symbol timing loops and bit decision circuits (matched filters) to perform the bit synchronization function. Several decoding algorithms, including differential, de-interleaving, Viterbi, and Reed-Solomon, are available for the down-link telemetry. Command format checking and CRC status is also available on FSK-demodulated data. Direct carrier BPSK/QPSK demodulation has decoding and frame synchronization capabilities. Because of the modular construction of the firmware and the use of FPGAs and DSPs, the system can be loaded with only the functions in use, lowering initial setup time while increasing overall system capability. To support a particular function, the card is downloaded with an “image,” which programs the FPGAs and DSPs at initialization. The user can change configurations by simply downloading a new set of instructions to the FPGA/DSP on the fly to keep the ground station running with minimal downtime. The flexibility of the design minimizes spare board costs, while achieving greater programmability at the end-user location.

International Telemetering Conference Proceedings / October 23-26, 2000 / Town & Country Hotel and Conference Center, San Diego, California / Range to an orbiting satellite from a ground reference point (ground station) can be determined by measuring the round trip time for a waveform transmitted to the satellite and returned to the ground station (Turnaround Ranging) and more recently by using the Global Positioning System (GPS). This paper first summarizes and compares the two approaches. The paper then describes and analyzes a new turn-around ranging system which uses a flexible ranging waveform that provides spectral compatibility with existing Military, NASA, and Commercial satellite uplink/downlink signals.

Speeding up addition is the key to faster digital signal processing (DSP). This can be achieved by exploiting the properties of redundant number systems. Their expanded symbol (digit) alphabet gives them multiple representations for most values. Utilising redundant representations at the output of an adder permits addition to be performed without carry-propagation, yielding fast, constant time performance irrespective of the word length. A resource efficient implementation of this fast adder structure is developed that re-purposes the fast carry logic of low-cost field programmable gate arrays (FPGAs). Experiments confirm constant time addition and show that it outperforms binary ripple carry addition at word lengths of greater than 44 bits in a Xilinx Spartan 3 FPGA and 24 bits in an Altera Cyclone III FPGA. Redundancy also provides other properties that can be exploited for performance gain. Some redundant representations will have more zero-symbols than others. These maximise the opportunities to exploit the multiplicative absorbing and additive identity properties of zero that when exercised reduce superfluous calculations. A serial recoding algorithm is developed that generates a redundant representation for a specified value with as few nonzero symbols as possible. Unlike previously published methods, it accepts a wide specification of number systems including those with irregularly spaced symbol alphabets. A Markov analysis and analysis of the elementary cycles in the formulated state machine provides average and worst case measures for the tested number system. Typically, the average number of non-zero symbols is less than a third and the worst case is less than a half. Further to the increase in zero-symbols, zero-dominance is proposed as a new property of redundant number representations. It promotes a set of representations that have uniquely positioned zero-symbols, in a Pareto-optimal sense. This set covers all representations of a value and is used to select representations to optimise the calculation of a dot-product. The dot-product or vector-multiply is a fundamental operation in DSP, since it is employed in filtering, correlation and convolution. The nonzero partial products can be packed together, substantially reducing the calculation time. The application of redundant number systems provides a two-fold benefit. Firstly, the number of nonzero partial products is reduced. Secondly, a novel opportunity is identified to use the representations in the zero-dominant set to optimise the packing further, gaining an extra 18% improvement. An implementation of the proposed dot-product with partial product packing is developed for a Cyclone II FPGA. It outperforms a quad-multiplier binary implementation in throughput by 50% . Redundant number systems excel at increasing performance in particular DSP subsystems, those that are numerically intensive and consist of considerable accumulation. The conversion back to a binary result is the performance bottleneck in the DSP algorithm, taking a time proportional to a binary adder. Therefore, redundant number systems are best utilised when this conversion cost can be amortised over many fast redundant additions, which is typical in many DSP and communications applications.

This thesis reports on new music technology research which investigates the use of cellular automata (CA) for the digital synthesis of dynamic sounds. The research addresses the problem of the sound design limitations of synthesis techniques based on CA. These limitations fundamentally stem from the unpredictable and autonomous nature of these computational models. Therefore, the aim of this thesis is to develop a sound synthesis technique based on CA capable of allowing a sound design process. A critical analysis of previous research in this area will be presented in order to justify that this problem has not been previously solved. Also, it will be discussed why this problem is worthwhile to solve. In order to achieve such aim, a novel approach is proposed which considers the output of CA as digital signals and uses DSP procedures to analyse them. This approach opens a large variety of possibilities for better understanding the self-organization process of CA with a view to identifying not only mapping possibilities for making the synthesis of sounds possible, but also control possibilities which enable a sound design process. As a result of this approach, this thesis presents a technique called Histogram Mapping Synthesis (HMS), which is based on the statistical analysis of CA evolutions by histogram measurements. HMS will be studied with four different automatons, and a considerable number of control mechanisms will be presented. These will show that HMS enables a reasonable sound design process. With these control mechanisms it is possible to design and produce in a predictable and controllable manner a variety of timbres. Some of these timbres are imitations of sounds produced by acoustic means and others are novel. All the sounds obtained present dynamic features and many of them, including some of those that are novel, retain important characteristics of sounds produced by acoustic means.

Master of Science / Department of Computing and Information Sciences / William H. Hsu / Pitch detection and instrument identification can be achieved with relatively high accuracy when considering monophonic signals in music; however, accurately classifying polyphonic signals in music remains an unsolved research problem. Pitch and instrument classification is a subset of Music Information Retrieval (MIR) and automatic music transcription, both having numerous research and real-world applications. Several areas of research are covered in this thesis, including the fast Fourier transform, onset detection, convolution, and filtering. Basic music theory and terms are also presented in order to explain the context and structure of data used. The focus of this thesis is on the representation of musical signals in the frequency domain. Polyphonic signals with many different voices and frequencies can be exceptionally complex. This thesis presents a new model for representing the spectral structure of polyphonic signals: Uniform MAx Gaussian Envelope (UMAGE). The new spectral envelope precisely approximates the distribution of frequency parts in the spectrum while still being resilient to oscillating rapidly (noise) and is able to generalize well without losing the representation of the original spectrum. When subjectively compared to other spectral envelope methods, such as the linear predictive coding envelope method and the cepstrum envelope method, UMAGE is able to model high order polyphonic signals without dropping partials (frequencies present in the signal). In other words, UMAGE is able to model a signal independent of the signal’s periodicity. The performance of UMAGE is evaluated both objectively and subjectively. It is shown that UMAGE is robust at modeling the distribution of frequencies in simple and complex polyphonic signals. Combined classification (combiners), a methodology for learning large concepts, is used to simplify the learning process and boost classification results. The output of each learner is then averaged to get the final result. UMAGE is less accurate when identifying pitches; however, it is able to achieve accuracy in identifying instrument groups on order-10 polyphonic signals (ten voices), which is competitive with the current state of the field.

Dynamic Time Warping (DTW) é uma técnica do tipo pattern matching para reconhecimento de padrões de voz, sendo baseada no alinhamento temporal de um sinal com os diversos modelos de referência. Uma desvantagem da DTW é o seu alto custo computacional. Este trabalho apresenta uma versão da DTW que, utilizando a Transformada Wavelet Discreta (DWT), reduz a sua complexidade. O desempenho obtido com a proposta foi muito promissor, ganhando em termos de velocidade de reconhecimento e recursos de memória consumidos, enquanto a precisão da DTW não é afetada. Os testes foram realizados com alguns fonemas extraídos da base de dados TIMIT do Linguistic Data Consortium (LDC) / Dynamic TimeWarping (DTW) is a pattern matching technique for speech recognition, that is based on a temporal alignment of the input signal with the template models. One drawback of this technique is its high computational cost. This work presents a modified version of the DTW, based on the DiscreteWavelet Transform (DWT), that reduces the complexity of the original algorithm. The performance obtained with the proposed algorithm is very promising, improving the recognition in terms of time and memory allocation, while the precision is not affected. Tests were performed with speech data collected from TIMIT corpus provided by Linguistic Data Consortium (LDC).

Neste trabalho foi feita uma investigação sobre a realização de processamento de áudio digital em tempo real utilizando três dispositivos com características computacionais fundamentalmente distintas porém bastante acessíveis em termos de custo e disponibilidade de tecnologia: Arduino, GPU e Android. Arduino é um dispositivo com licenças de hardware e software abertas, baseado em um microcontrolador com baixo poder de processamento, muito utilizado como plataforma educativa e artística para computações de controle e interface com outros dispositivos. GPU é uma arquitetura de placas de vídeo com foco no processamento paralelo, que tem motivado o estudo de modelos de programação específicos para sua utilização como dispositivo de processamento de propósito geral. Android é um sistema operacional para dispositivos móveis baseado no kernel do Linux, que permite o desenvolvimento de aplicativos utilizando linguagem de alto nível e possibilita o uso da infraestrutura de sensores, conectividade e mobilidade disponível nos aparelhos. Buscamos sistematizar as limitações e possibilidades de cada plataforma através da implementação de técnicas de processamento de áudio digital em tempo real e da análise da intensidade computacional em cada ambiente. / This dissertation describes an investigation about real time audio signal processing using three platforms with fundamentally distinct computational characteristics, but which are highly available in terms of cost and technology: Arduino, GPU boards and Android devices. Arduino is a device with open hardware and software licences, based on a microcontroller with low processing power, largely used as educational and artistic platform for control computations and interfacing with other devices. GPU is a video card architecture focusing on parallel processing, which has motivated the study of specific programming models for its use as a general purpose processing device. Android is an operating system for mobile devices based on the Linux kernel, which allows the development of applications using high level language and allows the use of sensors, connectivity and mobile infrastructures available on devices. We search to systematize the limitations and possibilities of each platform through the implementation of real time digital audio processing techinques and the analysis of computational intensity in each environment.

As elevadas perdas existentes no setor elétrico tem causado preocupação nas empresas de distribuição, aliadas ainda a necessidade crescente de um desenvolvimento econômico sustentável. Neste cenário a calibração periódica dos instrumentos destinados a medição (entre eles os transformadores de instrumentos) tornam-se essenciais e tais procedimentos encontram-se previstos no novo modelo de operação do setor elétrico. Porém, as dificuldades logísticas e operacionais de transporte a laboratórios metrológicos credenciados dificultam a execução de tais serviços. As técnicas e métodos desenvolvidos nesse trabalho visam a implementação de uma unidade eletrônica capaz de aquisitar e processar dados provenientes de transformadores de instrumentos, de natureza indutiva (denominado de convencional) e ópticos, bem como seus subsistemas de apoio, como ferramentas de medição e calibração portátil, móvel, para execução dos serviços metrológicos in situ nos ambientes das subestações e cabines primárias. Estes serviços, apesar de estarem em estágio incipiente, são de extremo interesse para empresas de energia elétrica. Este projeto está baseado no estado da arte de componentes da eletrônica analógica e digital, onde destacam-se conversores analógico/digital (A/D), microprocessadores, osciladores, FPGA e técnicas computacionais para processamento digital de sinais. São apresentadas as formas de implementação tanto em hardware como em software para esta unidade eletrônica de forma a atender aos requisitos funcionais especificados e às normas do INMETRO e normas internacionais equivalentes para aplicações metrológicas. A validação é baseada em testes comparativos dos fasores na frequência fundamental dos sinais obtidos, analisando os valores de amplitude (para cálculo de erro de relação) e de fase ( para cálculo de erro de fase) entre transformadores ópticos e convencionais, sendo que estes últimos podem ser de referência ou não. / The high losses in the electricity sector have caused concern in distribution companies, together with the growing need for sustainable economic development. In this scenario the periodic calibration of instruments intended for measurement (including instrument transformers) become essential and such procedures are provided for in the new model of operation of the electric sector. However, the logistical and operational difficulties of transportation to accredited metrological laboratories make it difficult to perform such services. The techniques and methods developed in this work are aimed at the implementation of an electronic unit capable of acquiring and processing data from instrument transformers of an inductive (conventional) and optical nature, as well as its supporting subsystems, such as portable and mobile measuring and calibration tools for the execution of on-site metrological services in the substations and primary cabins. These services, although in an incipient stage, are of extreme interest to electric energy companies. This project is based on the state-of-the-art components of analog and digital electronics, including analog/digital (A/D) converters, microprocessors, oscillators, FPGA and computational techniques for digital signal processing. The forms of implementation in both hardware and software for this electronic unit are presented in order to meet the functional requirements specified and the standards of the Instituto Nacional de Metrologia (INMETRO) and equivalent international standards for metrological applications. The validation is based on comparative tests of the phasors at the fundamental frequency of the obtained signals, analyzing the amplitude (for ratio error calculation) and phase (for phase error calculation) between optical and conventional transformers, the last one can be reference or not.

Sabe-se que a relaxação de líquidos em meios porosos envolve três mecanismos principais: relaxação bulk, relaxação de superfície e difusão. Muitas vezes, os processos de relaxação de líquidos confinados em meios porosos são dominados pelo processo de relaxação de superfície e difusão do fluído. No chamado regime de difusão rápida, a relaxação de um único poro é comandada por uma função mono exponencial que depende, principalmente, da relação superfície-volume do poro, de modo que em um material poroso, isto é, contendo uma distribuição ampla de tamanho de poros, o sinal de decaimento de magnetização obtido por meio da ressonância magnética nuclear é formado pela soma de exponenciais com diferentes tempos de relaxação. O problema-chave abordado neste trabalho consiste, portanto, em obter por meio desse sinal de magnetização a distribuição dos tempos de relaxação que controlam o decaimento das funções mono-exponenciais. Matematicamente, esse sinal de decaimento de magnetização pode ser descrito na forma geral de uma equação integral de Fredholm do primeiro tipo, cuja solução é um reconhecido problema inverso mal-posto. As abordagens utilizadas na tentativa de solucionar o problema são oriundas de uma área conhecida como processamento digital de sinais, e os seguintes métodos são analisados e comparados neste trabalho: algoritmo dos mínimos quadrados médios com restrição de não negatividade (LMS-NN), algoritmo dos mínimos quadrados médios com restrição de não negatividade e regularizado (LMS-RNN), redes recorrentes de Hopfield e o já bem conhecido na solução de problemas inversos mal-postos, o algoritmo dos mínimos quadrados regularizado (LS-R). Os resultados obtidos no trabalho são bastante positivos, demonstrando que, além do LS-R, existem outras alternativas na solução do problema, que principalmente, permitem atestar as soluções obtidas por qualquer um dos algoritmos. / It is known that the relaxation of liquids in porous media involves three principal mechanisms: bulk relaxation, surface relaxation, and diffusion. Relaxation processes of confined fluids in porous media are often controlled by surface relaxation process and diffusion. In the so-called fast diffusion regime, the relaxation of a single pore is governed by a mono-exponential function that depends primarily on the relation surface-volume of the pore, so that in a porous medium, i.e, in a medium which contains a wide distribution of pore sizes, the signal of magnetization decay obtained by nuclear magnetic resonance is composed by a sum of exponentials controlled by different relaxation times. The main issue discussed in this work consists in obtaining the distribution of relaxation times that controls the decay of the mono-exponential functions that comprise the magnetization signal. Mathematically this signal of magnetization decay can be generally described as a Fredholm integral equation of the first kind, whose solution is a recognized ill-posed inverse problem. The approaches adopted to try to solve the problem come from an area known as digital signal processing, and the following methods analyzed and compared are: non-negative least mean square algorithm (NN-LMS), regularized and nonnegative nleast mean square algorithm (RNN-LMS), recurrent Hopfield networks and regularized least square algorithm (R-LS), acknowledged in the solution of ill-posed inverse problems. The results obtained are very positive, and show that in addition to R-LS there are other alternatives in the solution of the problem, which mainly allow to attest the results achieved through any of the algorithms.