941 resultados para quantization noise
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The signal-to-noise (S/N) ratio in the reconstructed image from a binary hologram has been quantitatively related to the amplitude and phase quantization levels. The S/N ratio increases monotonically with increasing number of quantization levels. This observation is further supported by experimental results.
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Conselho Nacional de Desenvolvimento Científico e Tecnológico (CNPq)
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Many applications, including communications, test and measurement, and radar, require the generation of signals with a high degree of spectral purity. One method for producing tunable, low-noise source signals is to combine the outputs of multiple direct digital synthesizers (DDSs) arranged in a parallel configuration. In such an approach, if all noise is uncorrelated across channels, the noise will decrease relative to the combined signal power, resulting in a reduction of sideband noise and an increase in SNR. However, in any real array, the broadband noise and spurious components will be correlated to some degree, limiting the gains achieved by parallelization. This thesis examines the potential performance benefits that may arise from using an array of DDSs, with a focus on several types of common DDS errors, including phase noise, phase truncation spurs, quantization noise spurs, and quantizer nonlinearity spurs. Measurements to determine the level of correlation among DDS channels were made on a custom 14-channel DDS testbed. The investigation of the phase noise of a DDS array indicates that the contribution to the phase noise from the DACs can be decreased to a desired level by using a large enough number of channels. In such a system, the phase noise qualities of the source clock and the system cost and complexity will be the main limitations on the phase noise of the DDS array. The study of phase truncation spurs suggests that, at least in our system, the phase truncation spurs are uncorrelated, contrary to the theoretical prediction. We believe this decorrelation is due to the existence of an unidentified mechanism in our DDS array that is unaccounted for in our current operational DDS model. This mechanism, likely due to some timing element in the FPGA, causes some randomness in the relative phases of the truncation spurs from channel to channel each time the DDS array is powered up. This randomness decorrelates the phase truncation spurs, opening the potential for SFDR gain from using a DDS array. The analysis of the correlation of quantization noise spurs in an array of DDSs shows that the total quantization noise power of each DDS channel is uncorrelated for nearly all values of DAC output bits. This suggests that a near N gain in SQNR is possible for an N-channel array of DDSs. This gain will be most apparent for low-bit DACs in which quantization noise is notably higher than the thermal noise contribution. Lastly, the measurements of the correlation of quantizer nonlinearity spurs demonstrate that the second and third harmonics are highly correlated across channels for all frequencies tested. This means that there is no benefit to using an array of DDSs for the problems of in-band quantizer nonlinearities. As a result, alternate methods of harmonic spur management must be employed.
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The resolution of the digital signal path has a crucial impact on the design, performance and the power dissipation of the radio receiver data path, downstream from the ADC. The ADC quantization noise has been traditionally included with the Front End receiver noise in calculating the SNR as well as BER for the receiver. Using the IEEE 802.15.4 as an example, we show that this approach leads to an over-design for the ADC and the digital signal path, resulting in larger power. More accurate specifications for the front-end design can be obtained by making SNRreg a function of signal resolutions. We show that lower resolution signals provide adequate performance and quantization noise alone does not produce any bit-error. We find that a tight bandpass filter preceding the ADC can relax the resolution requirement and a 1-bit ADC degrades SNR by only 1.35 dB compared to 8-bit ADC. Signal resolution has a larger impact on the synchronization and a 1-bit ADC costs about 5 dB in SNR to maintain the same level of performance as a 8-bit ADC.
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Laser interferometer gravitational wave observatory (LIGO) consists of two complex large-scale laser interferometers designed for direct detection of gravitational waves from distant astrophysical sources in the frequency range 10Hz - 5kHz. Direct detection of space-time ripples will support Einstein's general theory of relativity and provide invaluable information and new insight into physics of the Universe.
Initial phase of LIGO started in 2002, and since then data was collected during six science runs. Instrument sensitivity was improving from run to run due to the effort of commissioning team. Initial LIGO has reached designed sensitivity during the last science run, which ended in October 2010.
In parallel with commissioning and data analysis with the initial detector, LIGO group worked on research and development of the next generation detectors. Major instrument upgrade from initial to advanced LIGO started in 2010 and lasted till 2014.
This thesis describes results of commissioning work done at LIGO Livingston site from 2013 until 2015 in parallel with and after the installation of the instrument. This thesis also discusses new techniques and tools developed at the 40m prototype including adaptive filtering, estimation of quantization noise in digital filters and design of isolation kits for ground seismometers.
The first part of this thesis is devoted to the description of methods for bringing interferometer to the linear regime when collection of data becomes possible. States of longitudinal and angular controls of interferometer degrees of freedom during lock acquisition process and in low noise configuration are discussed in details.
Once interferometer is locked and transitioned to low noise regime, instrument produces astrophysics data that should be calibrated to units of meters or strain. The second part of this thesis describes online calibration technique set up in both observatories to monitor the quality of the collected data in real time. Sensitivity analysis was done to understand and eliminate noise sources of the instrument.
Coupling of noise sources to gravitational wave channel can be reduced if robust feedforward and optimal feedback control loops are implemented. The last part of this thesis describes static and adaptive feedforward noise cancellation techniques applied to Advanced LIGO interferometers and tested at the 40m prototype. Applications of optimal time domain feedback control techniques and estimators to aLIGO control loops are also discussed.
Commissioning work is still ongoing at the sites. First science run of advanced LIGO is planned for September 2015 and will last for 3-4 months. This run will be followed by a set of small instrument upgrades that will be installed on a time scale of few months. Second science run will start in spring 2016 and last for about 6 months. Since current sensitivity of advanced LIGO is already more than factor of 3 higher compared to initial detectors and keeps improving on a monthly basis, upcoming science runs have a good chance for the first direct detection of gravitational waves.
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Automated assembly of mechanical devices is studies by researching methods of operating assembly equipment in a variable manner; that is, systems which may be configured to perform many different assembly operations are studied. The general parts assembly operation involves the removal of alignment errors within some tolerance and without damaging the parts. Two methods for eliminating alignment errors are discussed: a priori suppression and measurement and removal. Both methods are studied with the more novel measurement and removal technique being studied in greater detail. During the study of this technique, a fast and accurate six degree-of-freedom position sensor based on a light-stripe vision technique was developed. Specifications for the sensor were derived from an assembly-system error analysis. Studies on extracting accurate information from the sensor by optimally reducing redundant information, filtering quantization noise, and careful calibration procedures were performed. Prototype assembly systems for both error elimination techniques were implemented and used to assemble several products. The assembly system based on the a priori suppression technique uses a number of mechanical assembly tools and software systems which extend the capabilities of industrial robots. The need for the tools was determined through an assembly task analysis of several consumer and automotive products. The assembly system based on the measurement and removal technique used the six degree-of-freedom position sensor to measure part misalignments. Robot commands for aligning the parts were automatically calculated based on the sensor data and executed.
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Two classes of techniques have been developed to whiten the quantization noise in digital delta-sigma modulators (DDSMs): deterministic and stochastic. In this two-part paper, a design methodology for reduced-complexity DDSMs is presented. The design methodology is based on error masking. Rules for selecting the word lengths of the stages in multistage architectures are presented. We show that the hardware requirement can be reduced by up to 20% compared with a conventional design, without sacrificing performance. Simulation and experimental results confirm theoretical predictions. Part I addresses MultistAge noise SHaping (MASH) DDSMs; Part II focuses on single-quantizer DDSMs..
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Innerhalb digitaler Nebenstellenanlagen können kurzfristig die verbesserten Kommunikationsmöglichkeiten eines „Integrated Services Digital Network" unabhängig von der Existenz eines nur langfristig realisierbaren, durchgehend digitalen öffentlichen Netzes angeboten werden. Beim Anschluß an die bestehenden öffentlichen Netze ergeben sich jedoch einige Probleme: Z. B. müssen die Bedingungen des bestehenden Dämpfungsplans eingehalten werden, auch im ungünstigsten Fall sind ausreichend hohe Signal-Quantisierungsgeräusch-Abstände sicherzustellen, für die Text- und Datenkommunikation mit den derzeit bestehenden öffentlichen Netzen sind u. U. zusätzliche Einrichtungen in der Nebenstellenanlage erforderlich. Diese Einführungsprobleme werden diskutiert und Lösungsmöglichkeiten hierzu vorgeschlagen.
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Esta tesis está incluida dentro del campo del campo de Multiband Orthogonal Frequency Division Multiplexing Ultra Wideband (MB-OFDM UWB), el cual ha adquirido una gran importancia en las comunicaciones inalámbricas de alta tasa de datos en la última década. UWB surgió con el objetivo de satisfacer la creciente demanda de conexiones inalámbricas en interiores y de uso doméstico, con bajo coste y alta velocidad. La disponibilidad de un ancho de banda grande, el potencial para alta velocidad de transmisión, baja complejidad y bajo consumo de energía, unido al bajo coste de implementación, representa una oportunidad única para que UWB se convierta en una solución ampliamente utilizada en aplicaciones de Wireless Personal Area Network (WPAN). UWB está definido como cualquier transmisión que ocupa un ancho de banda de más de 20% de su frecuencia central, o más de 500 MHz. En 2002, la Comisión Federal de Comunicaciones (FCC) definió que el rango de frecuencias de transmisión de UWB legal es de 3.1 a 10.6 GHz, con una energía de transmisión de -41.3 dBm/Hz. Bajo las directrices de FCC, el uso de la tecnología UWB puede aportar una enorme capacidad en las comunicaciones de corto alcance. Considerando las ecuaciones de capacidad de Shannon, incrementar la capacidad del canal requiere un incremento lineal en el ancho de banda, mientras que un aumento similar de la capacidad de canal requiere un aumento exponencial en la energía de transmisión. En los últimos años, s diferentes desarrollos del UWB han sido extensamente estudiados en diferentes áreas, entre los cuales, el protocolo de comunicaciones inalámbricas MB-OFDM UWB está considerado como la mejor elección y ha sido adoptado como estándar ISO/IEC para los WPANs. Combinando la modulación OFDM y la transmisión de datos utilizando las técnicas de salto de frecuencia, el sistema MB-OFDM UWB es capaz de soportar tasas de datos con que pueden variar de los 55 a los 480 Mbps, alcanzando una distancia máxima de hasta 10 metros. Se esperara que la tecnología MB-OFDM tenga un consumo energético muy bajo copando un are muy reducida en silicio, proporcionando soluciones de bajo coste que satisfagan las demandas del mercado. Para cumplir con todas estas expectativas, el desarrollo y la investigación del MBOFDM UWB deben enfrentarse a varios retos, como son la sincronización de alta sensibilidad, las restricciones de baja complejidad, las estrictas limitaciones energéticas, la escalabilidad y la flexibilidad. Tales retos requieren un procesamiento digital de la señal de última generación, capaz de desarrollar sistemas que puedan aprovechar por completo las ventajas del espectro UWB y proporcionar futuras aplicaciones inalámbricas en interiores. Esta tesis se centra en la completa optimización de un sistema de transceptor de banda base MB-OFDM UWB digital, cuyo objetivo es investigar y diseñar un subsistema de comunicación inalámbrica para la aplicación de las Redes de Sensores Inalámbricas Visuales. La complejidad inherente de los procesadores FFT/IFFT y el sistema de sincronización así como la alta frecuencia de operación para todos los elementos de procesamiento, se convierten en el cuello de la botella para el diseño y la implementación del sistema de UWB digital en base de banda basado en MB-OFDM de baja energía. El objetivo del transceptor propuesto es conseguir baja energía y baja complejidad bajo la premisa de un alto rendimiento. Las optimizaciones están realizadas tanto a nivel algorítmico como a nivel arquitectural para todos los elementos del sistema. Una arquitectura hardware eficiente en consumo se propone en primer lugar para aquellos módulos correspondientes a núcleos de computación. Para el procesado de la Transformada Rápida de Fourier (FFT/IFFT), se propone un algoritmo mixed-radix, basado en una arquitectura con pipeline y se ha desarrollado un módulo de Decodificador de Viterbi (VD) equilibrado en coste-velocidad con el objetivo de reducir el consumo energético e incrementar la velocidad de procesamiento. También se ha implementado un correlador signo-bit simple basado en la sincronización del tiempo de símbolo es presentado. Este correlador es usado para detectar y sincronizar los paquetes de OFDM de forma robusta y precisa. Para el desarrollo de los subsitemas de procesamiento y realizar la integración del sistema completo se han empleado tecnologías de última generación. El dispositivo utilizado para el sistema propuesto es una FPGA Virtex 5 XC5VLX110T del fabricante Xilinx. La validación el propuesta para el sistema transceptor se ha implementado en dicha placa de FPGA. En este trabajo se presenta un algoritmo, y una arquitectura, diseñado con filosofía de co-diseño hardware/software para el desarrollo de sistemas de FPGA complejos. El objetivo principal de la estrategia propuesta es de encontrar una metodología eficiente para el diseño de un sistema de FPGA configurable optimizado con el empleo del mínimo esfuerzo posible en el sistema de procedimiento de verificación, por tanto acelerar el periodo de desarrollo del sistema. La metodología de co-diseño presentada tiene la ventaja de ser fácil de usar, contiene todos los pasos desde la propuesta del algoritmo hasta la verificación del hardware, y puede ser ampliamente extendida para casi todos los tipos de desarrollos de FPGAs. En este trabajo se ha desarrollado sólo el sistema de transceptor digital de banda base por lo que la comprobación de señales transmitidas a través del canal inalámbrico en los entornos reales de comunicación sigue requiriendo componentes RF y un front-end analógico. No obstante, utilizando la metodología de co-simulación hardware/software citada anteriormente, es posible comunicar el sistema de transmisor y el receptor digital utilizando los modelos de canales propuestos por IEEE 802.15.3a, implementados en MATLAB. Por tanto, simplemente ajustando las características de cada modelo de canal, por ejemplo, un incremento del retraso y de la frecuencia central, podemos estimar el comportamiento del sistema propuesto en diferentes escenarios y entornos. Las mayores contribuciones de esta tesis son: • Se ha propuesto un nuevo algoritmo 128-puntos base mixto FFT usando la arquitectura pipeline multi-ruta. Los complejos multiplicadores para cada etapa de procesamiento son diseñados usando la arquitectura modificada shiftadd. Los sistemas word length y twiddle word length son comparados y seleccionados basándose en la señal para cuantización del SQNR y el análisis de energías. • El desempeño del procesador IFFT es analizado bajo diferentes situaciones aritméticas de bloques de punto flotante (BFP) para el control de desbordamiento, por tanto, para encontrar la arquitectura perfecta del algoritmo IFFT basado en el procesador FFT propuesto. • Para el sistema de receptor MB-OFDM UWB se ha empleado una sincronización del tiempo innovadora, de baja complejidad y esquema de compensación, que consiste en funciones de Detector de Paquetes (PD) y Estimación del Offset del tiempo. Simplificando el cross-correlation y maximizar las funciones probables solo a sign-bit, la complejidad computacional se ve reducida significativamente. • Se ha propuesto un sistema de decodificadores Viterbi de 64 estados de decisión-débil usando velocidad base-4 de arquitectura suma-comparaselecciona. El algoritmo Two-pointer Even también es introducido en la unidad de rastreador de origen con el objetivo de conseguir la eficiencia en el hardware. • Se han integrado varias tecnologías de última generación en el completo sistema transceptor basebanda , con el objetivo de implementar un sistema de comunicación UWB altamente optimizado. • Un diseño de flujo mejorado es propuesto para el complejo sistema de implementación, el cual puede ser usado para diseños de Cadena de puertas de campo programable general (FPGA). El diseño mencionado no sólo reduce dramáticamente el tiempo para la verificación funcional, sino también provee un análisis automático como los errores del retraso del output para el sistema de hardware implementado. • Un ambiente de comunicación virtual es establecido para la validación del propuesto sistema de transceptores MB-OFDM. Este método es provisto para facilitar el uso y la conveniencia de analizar el sistema digital de basebanda sin parte frontera analógica bajo diferentes ambientes de comunicación. Esta tesis doctoral está organizada en seis capítulos. En el primer capítulo se encuentra una breve introducción al campo del UWB, tanto relacionado con el proyecto como la motivación del desarrollo del sistema de MB-OFDM. En el capítulo 2, se presenta la información general y los requisitos del protocolo de comunicación inalámbrica MBOFDM UWB. En el capítulo 3 se habla de la arquitectura del sistema de transceptor digital MB-OFDM de banda base . El diseño del algoritmo propuesto y la arquitectura para cada elemento del procesamiento está detallado en este capítulo. Los retos de diseño del sistema que involucra un compromiso de discusión entre la complejidad de diseño, el consumo de energía, el coste de hardware, el desempeño del sistema, y otros aspectos. En el capítulo 4, se ha descrito la co-diseñada metodología de hardware/software. Cada parte del flujo del diseño será detallado con algunos ejemplos que se ha hecho durante el desarrollo del sistema. Aprovechando esta estrategia de diseño, el procedimiento de comunicación virtual es llevado a cabo para probar y analizar la arquitectura del transceptor propuesto. Los resultados experimentales de la co-simulación y el informe sintético de la implementación del sistema FPGA son reflejados en el capítulo 5. Finalmente, en el capítulo 6 se incluye las conclusiones y los futuros proyectos, y también los resultados derivados de este proyecto de doctorado. ABSTRACT In recent years, the Wireless Visual Sensor Network (WVSN) has drawn great interest in wireless communication research area. They enable a wealth of new applications such as building security control, image sensing, and target localization. However, nowadays wireless communication protocols (ZigBee, Wi-Fi, and Bluetooth for example) cannot fully satisfy the demands of high data rate, low power consumption, short range, and high robustness requirements. New communication protocol is highly desired for such kind of applications. The Ultra Wideband (UWB) wireless communication protocol, which has increased in importance for high data rate wireless communication field, are emerging as an important topic for WVSN research. UWB has emerged as a technology that offers great promise to satisfy the growing demand for low-cost, high-speed digital wireless indoor and home networks. The large bandwidth available, the potential for high data rate transmission, and the potential for low complexity and low power consumption, along with low implementation cost, all present a unique opportunity for UWB to become a widely adopted radio solution for future Wireless Personal Area Network (WPAN) applications. UWB is defined as any transmission that occupies a bandwidth of more than 20% of its center frequency, or more than 500 MHz. In 2002, the Federal Communications Commission (FCC) has mandated that UWB radio transmission can legally operate in the range from 3.1 to 10.6 GHz at a transmitter power of -41.3 dBm/Hz. Under the FCC guidelines, the use of UWB technology can provide enormous capacity over short communication ranges. Considering Shannon’s capacity equations, increasing the channel capacity requires linear increasing in bandwidth, whereas similar channel capacity increases would require exponential increases in transmission power. In recent years, several different UWB developments has been widely studied in different area, among which, the MB-OFDM UWB wireless communication protocol is considered to be the leading choice and has recently been adopted in the ISO/IEC standard for WPANs. By combing the OFDM modulation and data transmission using frequency hopping techniques, the MB-OFDM UWB system is able to support various data rates, ranging from 55 to 480 Mbps, over distances up to 10 meters. The MB-OFDM technology is expected to consume very little power and silicon area, as well as provide low-cost solutions that can satisfy consumer market demands. To fulfill these expectations, MB-OFDM UWB research and development have to cope with several challenges, which consist of high-sensitivity synchronization, low- complexity constraints, strict power limitations, scalability, and flexibility. Such challenges require state-of-the-art digital signal processing expertise to develop systems that could fully take advantages of the UWB spectrum and support future indoor wireless applications. This thesis focuses on fully optimization for the MB-OFDM UWB digital baseband transceiver system, aiming at researching and designing a wireless communication subsystem for the Wireless Visual Sensor Networks (WVSNs) application. The inherent high complexity of the FFT/IFFT processor and synchronization system, and high operation frequency for all processing elements, becomes the bottleneck for low power MB-OFDM based UWB digital baseband system hardware design and implementation. The proposed transceiver system targets low power and low complexity under the premise of high performance. Optimizations are made at both algorithm and architecture level for each element of the transceiver system. The low-power hardwareefficient structures are firstly proposed for those core computation modules, i.e., the mixed-radix algorithm based pipelined architecture is proposed for the Fast Fourier Transform (FFT/IFFT) processor, and the cost-speed balanced Viterbi Decoder (VD) module is developed, in the aim of lowering the power consumption and increasing the processing speed. In addition, a low complexity sign-bit correlation based symbol timing synchronization scheme is presented so as to detect and synchronize the OFDM packets robustly and accurately. Moreover, several state-of-the-art technologies are used for developing other processing subsystems and an entire MB-OFDM digital baseband transceiver system is integrated. The target device for the proposed transceiver system is Xilinx Virtex 5 XC5VLX110T FPGA board. In order to validate the proposed transceiver system in the FPGA board, a unified algorithm-architecture-circuit hardware/software co-design environment for complex FPGA system development is presented in this work. The main objective of the proposed strategy is to find an efficient methodology for designing a configurable optimized FPGA system by using as few efforts as possible in system verification procedure, so as to speed up the system development period. The presented co-design methodology has the advantages of easy to use, covering all steps from algorithm proposal to hardware verification, and widely spread for almost all kinds of FPGA developments. Because only the digital baseband transceiver system is developed in this thesis, the validation of transmitting signals through wireless channel in real communication environments still requires the analog front-end and RF components. However, by using the aforementioned hardware/software co-simulation methodology, the transmitter and receiver digital baseband systems get the opportunity to communicate with each other through the channel models, which are proposed from the IEEE 802.15.3a research group, established in MATLAB. Thus, by simply adjust the characteristics of each channel model, e.g. mean excess delay and center frequency, we can estimate the transmission performance of the proposed transceiver system through different communication situations. The main contributions of this thesis are: • A novel mixed radix 128-point FFT algorithm by using multipath pipelined architecture is proposed. The complex multipliers for each processing stage are designed by using modified shift-add architectures. The system wordlength and twiddle word-length are compared and selected based on Signal to Quantization Noise Ratio (SQNR) and power analysis. • IFFT processor performance is analyzed under different Block Floating Point (BFP) arithmetic situations for overflow control, so as to find out the perfect architecture of IFFT algorithm based on the proposed FFT processor. • An innovative low complex timing synchronization and compensation scheme, which consists of Packet Detector (PD) and Timing Offset Estimation (TOE) functions, for MB-OFDM UWB receiver system is employed. By simplifying the cross-correlation and maximum likelihood functions to signbit only, the computational complexity is significantly reduced. • A 64 state soft-decision Viterbi Decoder system by using high speed radix-4 Add-Compare-Select architecture is proposed. Two-pointer Even algorithm is also introduced into the Trace Back unit in the aim of hardware-efficiency. • Several state-of-the-art technologies are integrated into the complete baseband transceiver system, in the aim of implementing a highly-optimized UWB communication system. • An improved design flow is proposed for complex system implementation which can be used for general Field-Programmable Gate Array (FPGA) designs. The design method not only dramatically reduces the time for functional verification, but also provides automatic analysis such as errors and output delays for the implemented hardware systems. • A virtual communication environment is established for validating the proposed MB-OFDM transceiver system. This methodology is proved to be easy for usage and convenient for analyzing the digital baseband system without analog frontend under different communication environments. This PhD thesis is organized in six chapters. In the chapter 1 a brief introduction to the UWB field, as well as the related work, is done, along with the motivation of MBOFDM system development. In the chapter 2, the general information and requirement of MB-OFDM UWB wireless communication protocol is presented. In the chapter 3, the architecture of the MB-OFDM digital baseband transceiver system is presented. The design of the proposed algorithm and architecture for each processing element is detailed in this chapter. Design challenges of such system involve trade-off discussions among design complexity, power consumption, hardware cost, system performance, and some other aspects. All these factors are analyzed and discussed. In the chapter 4, the hardware/software co-design methodology is proposed. Each step of this design flow will be detailed by taking some examples that we met during system development. Then, taking advantages of this design strategy, the Virtual Communication procedure is carried out so as to test and analyze the proposed transceiver architecture. Experimental results from the co-simulation and synthesis report of the implemented FPGA system are given in the chapter 5. The chapter 6 includes conclusions and future work, as well as the results derived from this PhD work.
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El uso de aritmética de punto fijo es una opción de diseño muy extendida en sistemas con fuertes restricciones de área, consumo o rendimiento. Para producir implementaciones donde los costes se minimicen sin impactar negativamente en la precisión de los resultados debemos llevar a cabo una asignación cuidadosa de anchuras de palabra. Encontrar la combinación óptima de anchuras de palabra en coma fija para un sistema dado es un problema combinatorio NP-hard al que los diseñadores dedican entre el 25 y el 50 % del ciclo de diseño. Las plataformas hardware reconfigurables, como son las FPGAs, también se benefician de las ventajas que ofrece la aritmética de coma fija, ya que éstas compensan las frecuencias de reloj más bajas y el uso más ineficiente del hardware que hacen estas plataformas respecto a los ASICs. A medida que las FPGAs se popularizan para su uso en computación científica los diseños aumentan de tamaño y complejidad hasta llegar al punto en que no pueden ser manejados eficientemente por las técnicas actuales de modelado de señal y ruido de cuantificación y de optimización de anchura de palabra. En esta Tesis Doctoral exploramos distintos aspectos del problema de la cuantificación y presentamos nuevas metodologías para cada uno de ellos: Las técnicas basadas en extensiones de intervalos han permitido obtener modelos de propagación de señal y ruido de cuantificación muy precisos en sistemas con operaciones no lineales. Nosotros llevamos esta aproximación un paso más allá introduciendo elementos de Multi-Element Generalized Polynomial Chaos (ME-gPC) y combinándolos con una técnica moderna basada en Modified Affine Arithmetic (MAA) estadístico para así modelar sistemas que contienen estructuras de control de flujo. Nuestra metodología genera los distintos caminos de ejecución automáticamente, determina las regiones del dominio de entrada que ejercitarán cada uno de ellos y extrae los momentos estadísticos del sistema a partir de dichas soluciones parciales. Utilizamos esta técnica para estimar tanto el rango dinámico como el ruido de redondeo en sistemas con las ya mencionadas estructuras de control de flujo y mostramos la precisión de nuestra aproximación, que en determinados casos de uso con operadores no lineales llega a tener tan solo una desviación del 0.04% con respecto a los valores de referencia obtenidos mediante simulación. Un inconveniente conocido de las técnicas basadas en extensiones de intervalos es la explosión combinacional de términos a medida que el tamaño de los sistemas a estudiar crece, lo cual conlleva problemas de escalabilidad. Para afrontar este problema presen tamos una técnica de inyección de ruidos agrupados que hace grupos con las señales del sistema, introduce las fuentes de ruido para cada uno de los grupos por separado y finalmente combina los resultados de cada uno de ellos. De esta forma, el número de fuentes de ruido queda controlado en cada momento y, debido a ello, la explosión combinatoria se minimiza. También presentamos un algoritmo de particionado multi-vía destinado a minimizar la desviación de los resultados a causa de la pérdida de correlación entre términos de ruido con el objetivo de mantener los resultados tan precisos como sea posible. La presente Tesis Doctoral también aborda el desarrollo de metodologías de optimización de anchura de palabra basadas en simulaciones de Monte-Cario que se ejecuten en tiempos razonables. Para ello presentamos dos nuevas técnicas que exploran la reducción del tiempo de ejecución desde distintos ángulos: En primer lugar, el método interpolativo aplica un interpolador sencillo pero preciso para estimar la sensibilidad de cada señal, y que es usado después durante la etapa de optimización. En segundo lugar, el método incremental gira en torno al hecho de que, aunque es estrictamente necesario mantener un intervalo de confianza dado para los resultados finales de nuestra búsqueda, podemos emplear niveles de confianza más relajados, lo cual deriva en un menor número de pruebas por simulación, en las etapas iniciales de la búsqueda, cuando todavía estamos lejos de las soluciones optimizadas. Mediante estas dos aproximaciones demostramos que podemos acelerar el tiempo de ejecución de los algoritmos clásicos de búsqueda voraz en factores de hasta x240 para problemas de tamaño pequeño/mediano. Finalmente, este libro presenta HOPLITE, una infraestructura de cuantificación automatizada, flexible y modular que incluye la implementación de las técnicas anteriores y se proporciona de forma pública. Su objetivo es ofrecer a desabolladores e investigadores un entorno común para prototipar y verificar nuevas metodologías de cuantificación de forma sencilla. Describimos el flujo de trabajo, justificamos las decisiones de diseño tomadas, explicamos su API pública y hacemos una demostración paso a paso de su funcionamiento. Además mostramos, a través de un ejemplo sencillo, la forma en que conectar nuevas extensiones a la herramienta con las interfaces ya existentes para poder así expandir y mejorar las capacidades de HOPLITE. ABSTRACT Using fixed-point arithmetic is one of the most common design choices for systems where area, power or throughput are heavily constrained. In order to produce implementations where the cost is minimized without negatively impacting the accuracy of the results, a careful assignment of word-lengths is required. The problem of finding the optimal combination of fixed-point word-lengths for a given system is a combinatorial NP-hard problem to which developers devote between 25 and 50% of the design-cycle time. Reconfigurable hardware platforms such as FPGAs also benefit of the advantages of fixed-point arithmetic, as it compensates for the slower clock frequencies and less efficient area utilization of the hardware platform with respect to ASICs. As FPGAs become commonly used for scientific computation, designs constantly grow larger and more complex, up to the point where they cannot be handled efficiently by current signal and quantization noise modelling and word-length optimization methodologies. In this Ph.D. Thesis we explore different aspects of the quantization problem and we present new methodologies for each of them: The techniques based on extensions of intervals have allowed to obtain accurate models of the signal and quantization noise propagation in systems with non-linear operations. We take this approach a step further by introducing elements of MultiElement Generalized Polynomial Chaos (ME-gPC) and combining them with an stateof- the-art Statistical Modified Affine Arithmetic (MAA) based methodology in order to model systems that contain control-flow structures. Our methodology produces the different execution paths automatically, determines the regions of the input domain that will exercise them, and extracts the system statistical moments from the partial results. We use this technique to estimate both the dynamic range and the round-off noise in systems with the aforementioned control-flow structures. We show the good accuracy of our approach, which in some case studies with non-linear operators shows a 0.04 % deviation respect to the simulation-based reference values. A known drawback of the techniques based on extensions of intervals is the combinatorial explosion of terms as the size of the targeted systems grows, which leads to scalability problems. To address this issue we present a clustered noise injection technique that groups the signals in the system, introduces the noise terms in each group independently and then combines the results at the end. In this way, the number of noise sources in the system at a given time is controlled and, because of this, the combinato rial explosion is minimized. We also present a multi-way partitioning algorithm aimed at minimizing the deviation of the results due to the loss of correlation between noise terms, in order to keep the results as accurate as possible. This Ph.D. Thesis also covers the development of methodologies for word-length optimization based on Monte-Carlo simulations in reasonable times. We do so by presenting two novel techniques that explore the reduction of the execution times approaching the problem in two different ways: First, the interpolative method applies a simple but precise interpolator to estimate the sensitivity of each signal, which is later used to guide the optimization effort. Second, the incremental method revolves on the fact that, although we strictly need to guarantee a certain confidence level in the simulations for the final results of the optimization process, we can do it with more relaxed levels, which in turn implies using a considerably smaller amount of samples, in the initial stages of the process, when we are still far from the optimized solution. Through these two approaches we demonstrate that the execution time of classical greedy techniques can be accelerated by factors of up to ×240 for small/medium sized problems. Finally, this book introduces HOPLITE, an automated, flexible and modular framework for quantization that includes the implementation of the previous techniques and is provided for public access. The aim is to offer a common ground for developers and researches for prototyping and verifying new techniques for system modelling and word-length optimization easily. We describe its work flow, justifying the taken design decisions, explain its public API and we do a step-by-step demonstration of its execution. We also show, through an example, the way new extensions to the flow should be connected to the existing interfaces in order to expand and improve the capabilities of HOPLITE.
Resumo:
The need for low bit-rate speech coding is the result of growing demand on the available radio bandwidth for mobile communications both for military purposes and for the public sector. To meet this growing demand it is required that the available bandwidth be utilized in the most economic way to accommodate more services. Two low bit-rate speech coders have been built and tested in this project. The two coders combine predictive coding with delta modulation, a property which enables them to achieve simultaneously the low bit-rate and good speech quality requirements. To enhance their efficiency, the predictor coefficients and the quantizer step size are updated periodically in each coder. This enables the coders to keep up with changes in the characteristics of the speech signal with time and with changes in the dynamic range of the speech waveform. However, the two coders differ in the method of updating their predictor coefficients. One updates the coefficients once every one hundred sampling periods and extracts the coefficients from input speech samples. This is known in this project as the Forward Adaptive Coder. Since the coefficients are extracted from input speech samples, these must be transmitted to the receiver to reconstruct the transmitted speech sample, thus adding to the transmission bit rate. The other updates its coefficients every sampling period, based on information of output data. This coder is known as the Backward Adaptive Coder. Results of subjective tests showed both coders to be reasonably robust to quantization noise. Both were graded quite good, with the Forward Adaptive performing slightly better, but with a slightly higher transmission bit rate for the same speech quality, than its Backward counterpart. The coders yielded acceptable speech quality of 9.6kbps for the Forward Adaptive and 8kbps for the Backward Adaptive.
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Possible integration of Single Electron Transistor (SET) with CMOS technology is making the study of semiconductor SET more important than the metallic SET and consequently, the study of energy quantization effects on semiconductor SET devices and circuits is gaining significance. In this paper, for the first time, the effects of energy quantization on SET inverter performance are examined through analytical modeling and Monte Carlo simulations. It is observed that the primary effect of energy quantization is to change the Coulomb Blockade region and drain current of SET devices and as a result affects the noise margin, power dissipation, and the propagation delay of SET inverter. A new model for the noise margin of SET inverter is proposed which includes the energy quantization effects. Using the noise margin as a metric, the robustness of SET inverter is studied against the effects of energy quantization. It is shown that SET inverter designed with CT : CG = 1/3 (where CT and CG are tunnel junction and gate capacitances respectively) offers maximum robustness against energy quantization.
Resumo:
In this paper, for the first time, the effects of energy quantization on single electron transistor (SET) inverter performance are analyzed through analytical modeling and Monte Carlo simulations. It is shown that energy quantization mainly changes the Coulomb blockade region and drain current of SET devices and thus affects the noise margin, power dissipation, and the propagation delay of SET inverter. A new analytical model for the noise margin of SET inverter is proposed which includes the energy quantization effects. Using the noise margin as a metric, the robustness of SET inverter is studied against the effects of energy quantization. A compact expression is developed for a novel parameter quantization threshold which is introduced for the first time in this paper. Quantization threshold explicitly defines the maximum energy quantization that an SET inverter logic circuit can withstand before its noise margin falls below a specified tolerance level. It is found that SET inverter designed with CT:CG=1/3 (where CT and CG are tunnel junction and gate capacitances, respectively) offers maximum robustness against energy quantization.
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It is shown that the use of a coarsely quantized binary digital hologram as a matched filter on an optical computer does not degrade signal-to-noise ratio (SNR) appreciably.
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In this paper, the effects of energy quantization on different single-electron transistor (SET) circuits (logic inverter, current-biased circuits, and hybrid MOS-SET circuits) are analyzed through analytical modeling and Monte Carlo simulations. It is shown that energy quantizationmainly increases the Coulomb blockade area and Coulomb blockade oscillation periodicity, and thus, affects the SET circuit performance. A new model for the noise margin of the SET inverter is proposed, which includes the energy quantization effects. Using the noise margin as a metric, the robustness of the SET inverter is studied against the effects of energy quantization. An analytical expression is developed, which explicitly defines the maximum energy quantization (termed as ``quantization threshold'') that an SET inverter can withstand before its noise margin falls below a specified tolerance level. The effects of energy quantization are further studiedfor the current-biased negative differential resistance (NDR) circuitand hybrid SETMOS circuit. A new model for the conductance of NDR characteristics is also formulated that explains the energy quantization effects.