Digital filter synthesis

"Cornell Aeronautical Laboratory, Inc. has assigned Report no. XA-2177-B-1 to this document."

Análise e implementação de filtros digitais fraccionários

O projeto realizado teve como tema a aplicação das derivadas e integrais fraccionários para a implementação de filtros digitais numa perspetiva de processamento digital de sinais. Numa primeira fase do trabalho, é efetuado uma abordagem teórica sobre os filtros digitais e o cálculo fraccionário. Estes conceitos teóricos são utilizados posteriormente para o desenvolvimento do presente projeto. Numa segunda fase, é desenvolvida uma interface gráfica em ambiente MatLab, utilizando a ferramenta GUIDE. Esta interface gráfica tem como objetivo a implementação de filtros digitais fraccionários. Na terceira fase deste projeto são implementados os filtros desenvolvidos experimentalmente através do ADSP-2181, onde será possível analisar e comparar os resultados experimentais com os resultados obtidos por simulação no MatLab. Como quarta e última fase deste projeto é efetuado uma reflexão sobre todo o desenvolvimento da Tese e o que esta me proporcionou. Com este relatório pretendo apresentar todo o esforço aplicado na realização deste trabalho, bem como alguns dos conhecimentos adquiridos ao longo do curso.

Disseny d'un sistema de filtrat d'àudio

Tant el medi transmissor com els equips d'enregistrament o reproducció de so introdueixen components de soroll d'alta freqüència als senyals. En aquest treball de final de carrera (TFC), s'ha dissenyat i implementat un sistema de filtrat d'àudio encaminat a filtrar aquestes components d'alta freqüència. Donat que l'oïda humana no pot percebre sons de més de 20 kHz, s'ha considerat aquest límit com a freqüència màxima a mantenir en la senyal.S'ha començat estudiant el senyal problema a través del seu espectre de freqüències simulat mitjançant la transformada discreta de Fourier (DFT, en anglès). Una vegada identificades les components d'alta freqüència a atenuar, s'han estudiat les diferents opcions de filtre passabaix.Inicialment, s'ha valorat la possibilitat del disseny de filtres analògics de Butterworth o Chebyshev, o de filtres digitals de tipus IIR (Infinite Impulse Response) basats en els primers. Tanmateix, malgrat assolir les especificacions en magnitud, mitjançant aquest filtres no s'obté una fase lineal en la banda de pas. Per això, s'ha realitzat un disseny de filtre digital tipus FIR (Finite Infinite Response) que compleix estrictament amb les especificacions i presenta una fase lineal en la banda de pas. S'ha simulat el comportament d'aquest filtre amb el senyal problema per tal d'assegurar el seu correcte funcionament.A continuació, s'ha implementat aquest últim disseny en llenguatge C i compilat per un microcontrolador de l'empresa Microchip. S'han realitzat proves de simulació mitjançant Stimulus del programa MPLAB. En definitiva, s'ha dissenyat un filtre passabaix de tipus FIR per acondicionar una senyal d'àudio que posteriorment s'ha implementat en un microcontrolador de Microchip.

Filtros digitais por transformadas de Fourier aplicados em eletroquímica

The electrochemical properties of micro and nano-electrodes are widely investigated due to their low faradaic and capacitive currents, leading to a new generation of smart and implantable devices. However, the current signals obtained in low-dimensional devices are strongly influenced by noise sources. In this paper, we show the evaluation of filters based on Fast Fourier Transform (FFT) and their implementation in a graphical user interface (GUI) in MATLAB®. As a case study, we evaluated an electrochemical reaction process of charge transfer via outer-sphere. Results showed successful removal of most of the noise in signals, thus proving a promising tool for low-scale measurement.

Controle Adaptativo Robusto para Filtros Ativos de Potência Paralelo

Conventional control strategies used in shunt active power filters (SAPF) employs real-time instantaneous harmonic detection schemes which is usually implements with digital filters. This increase the number of current sensors on the filter structure which results in high costs. Furthermore, these detection schemes introduce time delays which can deteriorate the harmonic compensation performance. Differently from the conventional control schemes, this paper proposes a non-standard control strategy which indirectly regulates the phase currents of the power mains. The reference currents of system are generated by the dc-link voltage controller and is based on the active power balance of SAPF system. The reference currents are aligned to the phase angle of the power mains voltage vector which is obtained by using a dq phase locked loop (PLL) system. The current control strategy is implemented by an adaptive pole placement control strategy integrated to a variable structure control scheme (VS-APPC). In the VS-APPC, the internal model principle (IMP) of reference currents is used for achieving the zero steady state tracking error of the power system currents. This forces the phase current of the system mains to be sinusoidal with low harmonics content. Moreover, the current controllers are implemented on the stationary reference frame to avoid transformations to the mains voltage vector reference coordinates. This proposed current control strategy enhance the performance of SAPF with fast transient response and robustness to parametric uncertainties. Experimental results are showing for determining the effectiveness of SAPF proposed control system

Estratégia de controle robusto para filtro ativo paralelo sem detecção de harmônicos de correntes

Conventional control strategies used in shunt active power filters (SAPF) employs real-time instantaneous harmonic detection schemes which is usually implements with digital filters. This increase the number of current sensors on the filter structure which results in high costs. Furthermore, these detection schemes introduce time delays which can deteriorate the harmonic compensation performance. Differently from the conventional control schemes, this paper proposes a non-standard control strategy which indirectly regulates the phase currents of the power mains. The reference currents of system are generated by the dc-link voltage controller and is based on the active power balance of SAPF system. The reference currents are aligned to the phase angle of the power mains voltage vector which is obtained by using a dq phase locked loop (PLL) system. The current control strategy is implemented by an adaptive pole placement control strategy integrated to a variable structure control scheme (VS¡APPC). In the VS¡APPC, the internal model principle (IMP) of reference currents is used for achieving the zero steady state tracking error of the power system currents. This forces the phase current of the system mains to be sinusoidal with low harmonics content. Moreover, the current controllers are implemented on the stationary reference frame to avoid transformations to the mains voltage vector reference coordinates. This proposed current control strategy enhance the performance of SAPF with fast transient response and robustness to parametric uncertainties. Experimental results are showing for determining the effectiveness of SAPF proposed control system

Another prefiltering algorithm for adaptive IIR filtering

An algorithm for adaptive IIR filtering that uses prefiltering structure in direct form is presented. This structure has an estimation error that is a linear function of the coefficients. This property greatly simplifies the derivation of gradient-based algorithms. Computer simulations show that the proposed structure improves convergence speed.

Discrete wavelet transform signal analyzer

This paper addresses the problem of processing biological data, such as cardiac beats in the audio and ultrasonic range, and on calculating wavelet coefficients in real time, with the processor clock running at a frequency of present application-specified integrated circuits and field programmable gate array. The parallel filter architecture for discrete wavelet transform (DWT) has been improved, calculating the wavelet coefficients in real time with hardware reduced up to 60%. The new architecture, which also processes inverse DWT, is implemented with the Radix-2 or the Booth-Wallace constant multipliers. One integrated circuit signal analyzer in the ultrasonic range, including series memory register banks, is presented. © 2007 IEEE.

Monitoramento da queima da peça no processo de retificação plana baseado no estudo espectral do sinal de emissão acústica com implementação por hardware

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)

Reconhecimento de voz para aplicações em automação implementado em FPGA

In many movies of scientific fiction, machines were capable of speaking with humans. However mankind is still far away of getting those types of machines, like the famous character C3PO of Star Wars. During the last six decades the automatic speech recognition systems have been the target of many studies. Throughout these years many technics were developed to be used in applications of both software and hardware. There are many types of automatic speech recognition system, among which the one used in this work were the isolated word and independent of the speaker system, using Hidden Markov Models as the recognition system. The goals of this work is to project and synthesize the first two steps of the speech recognition system, the steps are: the speech signal acquisition and the pre-processing of the signal. Both steps were developed in a reprogrammable component named FPGA, using the VHDL hardware description language, owing to the high performance of this component and the flexibility of the language. In this work it is presented all the theory of digital signal processing, as Fast Fourier Transforms and digital filters and also all the theory of speech recognition using Hidden Markov Models and LPC processor. It is also presented all the results obtained for each one of the blocks synthesized e verified in hardware

A low-power silicon-on-insulator PWM discriminator for biomedical applications

A CMOS/SOI circuit to decode PWM signals is presented as part of a body-implanted neurostimulator for visual prosthesis. Since encoded data is the sole input to the circuit, the decoding technique is based on a double-integration concept and does not require dc filtering. Nonoverlapping control phases are internally derived from the incoming pulses and a fast-settling comparator ensures good discrimination accuracy in the megahertz range. The circuit was integrated on a 2 mu m single-metal SOI fabrication process and has an effective area of 2mm(2) Typically, the measured resolution of encoding parameter a was better than 10% at 6MHz and V-DD=3.3V. Stand-by consumption is around 340 mu W. Pulses with frequencies up to 15MHz and alpha = 10% can be discriminated for V-DD spanning from 2.3V to 3.3V. Such an excellent immunity to V-DD deviations meets a design specification with respect to inherent coupling losses on transmitting data and power by means of a transcutaneous link.

Impact of biannual rossby waves on the Indian Ocean Dipole

Topex/Poseidon sea surface height anomalies during 1993-2002 are decomposed using 2-D finite impulse response filters which showed biannual Rossby waves (BRWs) in the equatorial Indian Ocean (peak at 1.5 degrees S) and in the southern tropical Indian Ocean (peak at 10.5 degrees S) during Indian Ocean Dipole (IOD) years. Anomalous downwelling BRWs in the equatorial Indian Ocean triggered by the wind stress curl-induced Ekman pumping near the eastern boundary started propagating westward from the eastern boundary in July/August 1993 and 1996, i.e., more than one year prior to the formation of the IOD events of 1994 and 1997 respectively. These strong downwelling signals reach the western equatorial Indian Ocean during the peak dipole time.

Height variability from the MIROC - IPCC model for the 20th century compared to that of the TOPEX/POSEIDON altimeter

Planetary waves are key to large-scale dynamical adjustment in the global ocean as they transfer energy from the east to the west side of oceanic basins; they connect the forcing in the ocean interior with the variability at its boundaries: and they change the local heat content, thus coupling oceanic, atmospheric, and biological processes. Planetary waves, mostly of the first baroclinic mode, are observed as distinctive patterns in global time series of sea surface height anomaly (SSHA) and heat storage. The goal of this study is to compare and validate large-scale SSHA signals from coupled ocean-atmosphere general circulation Model for Interdisciplinary Research on Climate (MIROC) with TOPEX/POSEIDON satellite altimeter observations. The last decade of the models` time series is selected for comparison with the altimeter data. The wave patterns are separated from the meso- and large-scale SSHA signals by digital filters calibrated to select the same spectral bands in both model and altimeter data. The band-wise comparison allows for an assessment of the model skill to simulate the dynamical components of the observed wave field. Comparisons regarding both the seasonal cycle and the Rossby wave Held differ significantly among basins. When carried within the same basin, differences can occur between equal latitudes in opposite hemispheres. Furthermore, at some latitudes the MIROC reproduces biannual, annual and semiannual planetary waves with phase speeds and average amplitudes similar to those observed by the altimeter, but with significant differences in phase. (C) 2008 Elsevier Ltd. All rights reserved.

Parallel recusive algorithms for inverse discrete Legendre transform and inverse discrete Laguerre transform

This paper presents parallel recursive algorithms for the computation of the inverse discrete Legendre transform (DPT) and the inverse discrete Laguerre transform (IDLT). These recursive algorithms are derived using Clenshaw's recurrence formula, and they are implemented with a set of parallel digital filters with time-varying coefficients.

Parallel-recusive filter structures for the computation of discrete transforms

A general approach is presented for implementing discrete transforms as a set of first-order or second-order recursive digital filters. Clenshaw's recurrence formulae are used to formulate the second-order filters. The resulting structure is suitable for efficient implementation of discrete transforms in VLSI or FPGA circuits. The general approach is applied to the discrete Legendre transform as an illustration.