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Information is often stored and transmitted through electrical signals. This information may need refinement, which may be done by processing and altering the electrical signals, in which it is transmitted. When refining a signal, a frequency selective filter is often used. It can be implemented through digital signal processing (DSP). DSP is a concept where signals are refined using a digital compute system. Digital systems are designed to replace their analog counterpart, mitigating their flaws in scalability, complexity and cost. A DSP system is typically implemented using software on a small computer, while analog systems are implemented through various electronic components. The objective of this project is to design a DSP system that filters analog input data using automatically synthesised filters from user-defined input specifications. The DSP system is implemented using a microcontroller. The system designed the filters and found the filter coefficients. It then uses analog to digital converter (ADC) to sample an input signal and applies the filter. Lastly, it uses the digital to analog converter (DAC) to reconstruct a filtered, analog result. A user interface is not designed for the system, and only a limited number of filters are implemented. However, the system is successful in designing filters and finding their coefficients.

Filter devices are central elements of radio units utilized for modern-day telecommunication. Leakages of radio waves and passive intermodulation are evolving more prominent as the frequency bands become crowded. The study in this thesis investigates the screw joints in the filter assembly to determine the influential parameters on the connection between the filter lid and its chassis. The clamp and release torque are vital factors in the resulting pre-load force. Besides the geometrical characteristics of the chassis andthe self-threading screw, the friction and surface roughness on the filter lid are significant contributors. Extensive testing revealed that increasing the countersink in the lead holeand applying the lubricant Molykote 1000 Paste increase the consequent clamp torque. After numerous cycles of thermal shock testing, the release torque does not show significant dissimilarities between the tested parameter combinations. A surface roughness test shows that applying a silver coating to the filter lid increases its surface roughness and thereby the friction of the screw head on the surface. In conclusion, the pre-load force will increase by changing the parameters as mentioned above, while it requires further testing to make a suggestion regarding the effect of the silver coating on the filter lid. / Filteranordningar är centrala delar i radioenheter vilka används vid nutida telekommunikation. Läckage av radiovågor och passiv intermodulation blir mer framträdande när frekvensbanden blir trånga. Studien i denna avhandling undersöker skruvförbanden i filtermonteringen för att bestämma de parametrar som har störst inverkan på anslutningen mellan filterlocket och dess chassi. Klämvrid- och lossdragningsmomentet är avgörande faktorer i den resulterande förspänningskraften. Förutom de geometriska egenskaperna hos chassit och den självgängande skruven, är friktionen och ytfinheten på filterlocket starktbidragande orsaker. Omfattande tester visade att en ökning av försänkningen i pilothålet och applicering av smörjmedlet Molykote 1000 Paste ökar det efterföljande klämvridmomentet. Efter talrika cykler av termisk chocktestning visar lossdragningsmomentet inga signifikanta skillnader mellan de testade parameterkombinationerna. Ett ytjämnhetstestvisar att applicering av en silverbeläggning på filterlocket ökar ytans strävhet och därmedfriktionen av skruvhuvudet på ytan. Sammanfattningsvis kommer förspänningskraften att öka genom att parametrarna ändras som nämnts ovan, samtidigt som det krävs ytterligare tester för att komma med ett förslag angående effekten av silverbeläggningen på filterlocket.

In recent years, genetic algorithms (GAs) began to be used in many disciplines such as pattern recognition, robotics, biology, and medicine to name just a few. GAs are based on Darwin's principle of natural selection which happens to be a slow process and, as a result, these algorithms tend to require a large amount of computation. However, they offer certain advantages as well over classical gradient-based optimization algorithms such as steepest-descent and Newton-type algorithms. For example, having located local suboptimal solutions they can discard them in favor of more promising local solutions and, therefore, they are more likely to obtain better solutions in multimodal problems. By contrast, classical optimization algorithms though very efficient, they are not equipped to discard inferior local solutions in favour of more optimal ones. This dissertation is concerned with the design of several types of digital filters by using GAs as detailed bellow. In Chap. 2, two approaches for the design of fractional delay (FD) filters based on a GA are developed. The approaches exploit the advantages of a global search technique to determine the coefficients of FD FIR and allpass-IIR filters based on the so-called Farrow structure. The GA approach was compared with a least-squares approach and was found to lead to improvements in the amplitude response and/or delay characteristic. In Chap. 3, a GA-based approach is developed for the design of delay equalizers. In this approach, the equalizer coefficients are optimized using an objective function based on the passband filter-equalizer group delay. The required equalizer is built by adding new second-order sections until the desired accuracy in terms of the flatness of the group delay with respect to the passband is achieved. With this approach stable delay equalizers satisfying arbitrary prescribed specifications with the desired degree of group-delay flatness can easily be obtained. In Chap. 4, a GA-based approach for the design of multiplierless FIR filters is developed. A recently-introduced GA, called orthogonal GA (OGA) based on the so-called experimental design technique, is exploited to obtain fixed-point implementations of linear-phase FIR filters. In this approach, the effects of finite word length are minimized by considering the filter as a cascade of two sections. The OGA leads to an improved amplitude response relative to that of an equivalent direct-form cascade filter obtained using the Remez exchange algorithm. In Chap. 5, a multiobjective GA for the design of asymmetric FIR filters is proposed. This GA uses a specially tailored elitist nondominated sorting GA (ENSGA) to obtain so-called Pareto-optimal solutions for the problem at hand. Flexibility is introduced in the design by imposing phase-response linearity only in the passband instead of the entire baseband as in conventional designs. Three objective functions based on the amplitude-response error and the flatness of the group-delay characteristic are explored in the design examples considered. When compared with a WLS design method, the ENSGA was found to lead to improvements in the amplitude response and passband group-delay characteristic. In Chap. 6, a hybrid approach for the design of IIR filters using a GA along with a quasi-Newton (QN) algorithm is developed. The hybrid algorithm, referenced to as the genetic quasi-Newton (GQN) algorithm combines the flexibility and reliability inherent in the GA with the fast convergence and precision of the QN algorithm. The GA is used as a global search tool to explore different regions in the parameter space whereas the QN algorithm exploits the efficiency of a gradient-based algorithm in locating local solutions. The GQN algorithm works well with an arbitrary random initialization and filters that would satisfy prescribed amplitude-response specifications can easily be designed

Submitted by Renata Lopes (renatasil82@gmail.com) on 2016-06-07T14:12:32Z No. of bitstreams: 1 hugoviveirosschettino.pdf: 1159424 bytes, checksum: 924405f3d1d2d0ad229acb0e08e88be9 (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2016-07-13T13:15:33Z (GMT) No. of bitstreams: 1 hugoviveirosschettino.pdf: 1159424 bytes, checksum: 924405f3d1d2d0ad229acb0e08e88be9 (MD5) / Made available in DSpace on 2016-07-13T13:15:33Z (GMT). No. of bitstreams: 1 hugoviveirosschettino.pdf: 1159424 bytes, checksum: 924405f3d1d2d0ad229acb0e08e88be9 (MD5) Previous issue date: 2016-02-26 / Este trabalho investiga o uso de filtros digitais de resposta ao impulso finita (finite impulse response - FIR) e de resposta ao impulso infinita (infinite impulse response - IIR) para separar os sinais dos clusters de esquemas de multiplexação por divisão de frequência ortogonal clusterizado (clustered orthogonal frequency division multiplexing - clustered-OFDM), aplicado à sistema de comunicação via rede elétrica (power line communication - PLC). Nesse contexto, introduz-se uma formulação para explicitar a influência do filtro digital no esquema clustered- OFDM. A partir dessa formulação, uma equação que quantifica a vazão em função do uso dos filtros digitais, a qual leva em consideração a relação sinal ruído multicanal, a especificação do filtro e, principalmente, o comprimento dele, é apresentada. Através de simulações computacionais, analisa-se o desempenho do esquema sujeito a um conjunto de filtros digitais previamente selecionados, considerando três cenários: canal corrompido pelo ruído aditivo, gaussiano e branco (additive white Gaussian noise - AWGN), canal PLC indoor corrompido ruído (medido numa residência localizada na cidade de Juiz de Fora, MG) e canal PLC outdoor corrompido pelo ruído (medido numa rede de distribuição de energia elétrica de baixa tensão na cidade de Juiz de Fora, MG). De acordo com os resultados computacionais, dentre os filtros digitais analisados, os filtros digitais Chebychev tipo II, FIR interpolado (interpolated FIR - IFIR) e FIR de fase mínima e equiripple são, de maneira geral, os filtros digitais que proporcionam os melhores benefícios (maior vazão e menor complexidade computacional). Considerando-se a influência da precisão finita nesses três filtros digitais, conclui-se que o filtro digital IIR Chebychev tipo II quantizado proporciona a menor perda de vazão e, portanto, é um filtro bastante adequado para esquemas clustered-OFDM. / This work investigates the use of finite impulse response (FIR) and infinite impulse response (IIR) digital filters to separate the signals belonging distinct clusters in a clustered orthogonal frequency division multiplexing (clustered-OFDM) scheme, when it is applied to power line communication (PLC) system. In this context, we introduce a formulation to take into account the choice of the digital filter for the clustered-OFDM scheme. Based on this formulation, an equation that quantifies the data-rate as a function ofmultichannel signal noise ratio, digital filter design specification and length of the digital filter is presented. Performance analyses based on a previously selected set of digital filters over three scenarios, channel corrupted by additivewhite Gaussian noise (AWGN), indoor PLC channel corrupted by the noise (measured in a residence located in Juiz de Fora city, MG) and outdoor PLC corrupted by the noise (measured in a low voltage distribution network in Juiz de Fora city, MG), are presented. According to numerical results, Chebychev type II, interpolated FIR (IFIR) and equiripple minimum phase FIR digital filters offers the best benefits in terms of less data-rate loss and computer complexity reduction. Considering the influence of the finite precision in these three digital filters, simulation results shows that the quantized Chebychev type II IIR digital filter attains the smallest data-rate loss and, as a consequence, it a very appropriate digital filter for clustered-OFDM schemes.

The aim of this work is to implement digital filters into programmable gate array. The work also includes a description of the MEMS technology, including comparisons with the technology of MEMS microphones from various manufacturers. Another part is devoted to the Sigma-delta modulation. The main section is the design and implementation of digital CIC and FIR filters for signal processing of digital microphone, including simulation and verification of properties of the proposed filter in Matlab.

Dissertation thesis is focused on real digital signal processing by quadrature mirror digital filter banks. In the first part a dual channel quadrature mirror digital filter bank is described briefly. Mainly transfer functions for distorted subband signals are described. In the next part generalized sigma-delta modulator and its linear model are described. Subsequently the generalized sigma-delta modulator is used in decomposition part of quadrature mirror digital filter bank. Designed structure is analyzed. Two design method of transfer functions are designed for the structure on the basis of analysis results. The first method is suitable for hand-made design by intuitive distribution of zeros and poles. The second method is more suitable for computer design. It is iterative method based on correlation. Transfer functions for quadrature mirror digital filter bank with sigma-delta modulators design examples are also part of thesis. Application of designed structure of quadrature mirror digital filter bank can lead to bigger compression ration in lossy data compression.

The diploma thesis is focused on design and optimalization of digital octave and fraction-octave band filters. This thesis describe the behavior of filters in systems with fixed point arithmetics and investigate the impact of quantization coefficients for frequency response of filter. Filters, whitch has been designed, are implemented into simple software in C. Designed filters are in accordance with standard IEC 61260.

In neural implants and biohybrid research systems, the integration of electrode recording and stimulation front-ends with pre-processing circuitry promises a drastic increase in real-time capabilities [1,6]. In our proposed neural recording system, constant sampling with a bandwidth of 9.8kHz yields 6.73μV input-referred noise (IRN) at a power-per-channel of 0.34μW for the time-continuous ΔΣ−modulator, and 0.52μW for the digital filters and spike detectors. We introduce dynamic current/bandwidth selection at the ΔΣ and digital filter to reduce recording bandwidth at the absence of spikes (i.e. local field potentials). This is controlled by a two-level spike detection and adjusted by adaptive threshold estimation (ATE). Dynamic bandwidth selection reduces power by 53.7%, increasing the available channel count at a low heat dissipation. Adaptive back-gate voltage tuning (ABGVT) compensates for PVT variation in subthreshold circuits. This allows 1.8V input/output (IO) devices to operate at 0.4V supply voltage robustly. The proposed 64-channel neural recording system moreover includes a 16-channel adaptive compression engine (ACE) and an 8-channel on-chip current stimulator at 3.3V. The stimulator supports field-shaping approaches, promising increased selectivity in future research.