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1
Shahein, Ahmed [Verfasser], and Yiannos [Akademischer Betreuer] Manoli. "Power optimization methodologies for digital FIR decimation filters = Leistungsoptimierungsmethoden für digitale FIR Dezimationsfiltern." Freiburg : Universität, 2014. http://d-nb.info/1123480664/34.
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Chit, Nassim N. "Weighted Chebyshev complex-valued approximation for FIR digital filters." Thesis, Swansea University, 1987. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.278340.
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Sadeghifar, Mohammad Reza. "On High-Speed Digital-to-Analog Converters and Semi-Digital FIR Filters." Licentiate thesis, Linköpings universitet, Elektroniska Kretsar och System, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-114274.
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High-speed and high-resolution digital-to-analog converters (DACs) are vital components in all telecommunication systems. Radio-frequency digital-to-analog converter (RFDAC) provides high-speed and high-resolution conversion from digital domain to an analog signal. RFDACs can be employed in direct-conversion radio transmitter architectures. The idea of RFDAC is to utilize an oscillatory pulse-amplitude modulation instead of the conventional zero-order hold pulse amplitude modulation, which results in DAC output spectrum to have high energy high-frequency lobe, other than the Nyquist main lobe. The frequency of the oscillatory pulse can be chosen, with respect to the sample frequency, such that the aliasing images of the signal at integer multiples of the sample frequency are landed in the high-energy high-frequency lobes of the DAC frequency response. Therefore the high-frequency images of the signal can be used as the output of the DAC, i.e., no need to the mixing stage for frequency up-conversion after the DAC in the radio transmitter. The mixing stage however is not eliminated but it is rather moved into the DAC elements and therefore the local oscillator (LO) signal with high frequency should be delivered to each individual DAC element. In direct-conversion architecture of IQ modulators which utilize the RFDAC technique, however, there is a problem of finite image rejection. The origin of this problem is the different polarity of the spectral response of the oscillatory pulse-amplitude modulation in I and Q branches. The conditions where this problem can be alleviated in IQ modulator employing RFDACs is also discussed in this work. ΣΔ modulators are used preceding the DAC in the transmitter chain to reduce the digital signal’s number of bits, still maintain the same resolution. By utilizing the ΣΔ modulator now the total number of DAC elements has decreased and therefore the delivery of the high-frequency LO signal to each DAC element is practical. One of the costs of employing ΣΔ modulator, however, is a higher quantization noise power at the output of the DAC. The quantization noise is ideally spectrally shaped to out-of-band frequencies by the ΣΔ modulator. The shaped noise which usually has comparatively high power must be filtered out to fulfill the radio transmission spectral mask requirement. Semi-digital FIR filter can be used in the context of digital-to-analog conversion, cascaded with ΣΔ modulator to filter the out-of-band noise by the modulator. In the same time it converts the signal from digital domain to an analog quantity. In general case, we can have a multi-bit, semi-digital FIR filter where each tap of the filter is realized with a sub-DAC of M bits. The delay elements are also realized with M-bit shift registers. If the output of the modulator is given by a single bit, the semi-digital FIR filter taps are simply controlled by a single switch assuming a current-steering architecture DAC. One of the major advantages is that the static linearity of the DAC is optimum. Since there are only two output levels available in the DAC, the static transfer function, regardless of the mismatch errors, is always given by a straight line. In this work, the design of SDFIR filter is done through an optimization procedure where the ΣΔ noise transfer function is also taken into account. Different constraints are defined for different applications in formulation of the SDFIR optimization problem. For a given radio transmitter application the objective function can be defined as, e.g., the hardware cost for SDFIR implementation while the constraint can be set to fulfill the radio transmitter spectral emission mask.
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Karam, Lina J. "Design of complex digital FIR filters in the chebyshev sense." Diss., Georgia Institute of Technology, 1995. http://hdl.handle.net/1853/22219.
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Zhang, Yuhong. "Design and realization of FIR and bireciprocal wave digital filters." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0024/MQ51826.pdf.
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Rosler, Lucas Owen. "Design and Analysis of an FPGA Based Low Tap Band-stop FIR Filter." Youngstown State University / OhioLINK, 2021. http://rave.ohiolink.edu/etdc/view?acc_num=ysu1619798270047225.
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Chabbi, Charef. "VLSI NMOS hardware design of a linear phase FIR low pass digital filer." Ohio : Ohio University, 1985. http://www.ohiolink.edu/etd/view.cgi?ohiou1183749814.
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Piskin, Hatice. "Design And Implementation Of Fir Digital Filters With Variable Frequency Characteristics." Master's thesis, METU, 2005. http://etd.lib.metu.edu.tr/upload/2/12606853/index.pdf.
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Variable digital filters (VDF) find many application areas in communication, audio, speech and image processing. This thesis analyzes design and implementation of FIR digital filters with variable frequency characteristics and introduces two design methods. The design and implementation of the proposed methods are realized on Matlab software program. Various filter design examples and comparisons are also outlilned. One of the major application areas of VDFs is software defined radio (SDR). The interpolation problem on sample rate converter (SRC) unit of the SDR is solved by using these filters. Realizations of VDFs on SRC are outlined and described. Simulations on Simulink and a specific hardware are examined.
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Kwan, Man-Wai. "Minimal transmit redundancy FIR precoder-equalizer systems design /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20KWAN.
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Nayebi, Kambiz. "A time domain framework for the analysis and design of FIR multirate filter bank systems." Diss., Georgia Institute of Technology, 1990. http://hdl.handle.net/1853/13867.
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Huang, Walter. "Implementation of adaptive digital FIR and reprogrammable mixed-signal filters using distributed arithmetic." Diss., Atlanta, Ga. : Georgia Institute of Technology, 2009. http://hdl.handle.net/1853/31653.
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Thesis (Ph.D)--Electrical and Computer Engineering, Georgia Institute of Technology, 2010.<br>Committee Chair: Anderson, David V.; Committee Member: Ferri, Bonnie H.; Committee Member: Hasler, Paul E.; Committee Member: Kang, Sung Ha; Committee Member: McClellan, James H.; Committee Member: Wolf, Wayne H. Part of the SMARTech Electronic Thesis and Dissertation Collection.
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Fanian, Kaveh Electrical Engineering & Telecommunications Faculty of Engineering UNSW. "Diamond-shaped 2-dimentional digital FIR filters with high performance and low complexity." Publisher:University of New South Wales. Electrical Engineering & Telecommunications, 2008. http://handle.unsw.edu.au/1959.4/42898.
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2-D digital filters have found numerous applications in signal processing and their design is subject to intensive research. In general, these filters are divided into two types: Separable filters as a product of two 1-D filters, and Non-Separable filters. Separable filters are easier to design but work mostly for rectangular-shaped frequency spectrums. On the other hand, non-separable filters are preferred for designing other shapes of spectrum divisions such as diamond-shaped, circular-shaped, fan-shaped, etc. but their design is much more complicated. When designing 2-D filters, there are several important issues which should be considered. These issues are: the accuracy of the pass-band and the transition-band, the pass-band ripple, the stop-band attenuation and the complexity of the filter for its digital implementation. One important class of 2-D filters is the class of 2-D digital filters with Four-Fold symmetry. Despite the fact that these filters can be designed by using some kind of McClellan transform and yet admit fast digital implementation, the shape of their pass-band cannot be easily controlled. The accuracy of the description of the pass-band shape requires a high-order polynomial transformation, but such a transformation leads to the explosive growth of the filter order and its implementation complexity. While there exist some known approaches that can control the pass-band shape more efficiently, they all suffer from the fact that their transition-band should be wide enough to avoid possible singularities that may arise due to the interpolation step. In this study, the semi-definite programming as a tool, is adopted to design non-separable four-fold symmetric 2-D digital filters and it will be shown that unlike previously proposed semi-definite programming based approaches, this approach is advantageous due to the facts that all the filter specifications are met while interpolation is avoided, the dimension of the semi-definite programming formulation is kept moderate, and moreover, the designed filters admit fast digital implementation despite the fact that they are non-separable and are not designed based on 1-D filters. The simulation clearly confirms the viability of this approach. Finally, although only diamond-shaped filters are considered in this study, other filters such as circular-shaped, elliptic-shaped or fan-shaped are expected to be designed in a similar fashion.
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Hounsell, Benjamin Iain. "Programmable architectures for the automated design of digital FIR filters using evolvable hardware." Thesis, University of Edinburgh, 2001. http://hdl.handle.net/1842/14109.
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Continuing increases in both the size and complexity of digital signal processing (DSP) systems places a considerable demand on the design engineer to develop hardware architectures capable of fulfilling the growing functional requirements expected of modern DSP devices. Automated circuit design techniques provide the design engineer with a tool to more effectively generate high performance signal processors capable of meeting demanding specifications. Evolvable hardware (EHW) is a relatively new approach to automated circuit design which utilises advances in reconfigurable hardware technology and the power of modern micro processors to generate circuits based on the principles of natural selection and evolution. This thesis investigates the suitability of software-biased and hardware oriented programmable platforms, configured via EHW, and tailored for the automated design of high performance DSP circuits. Performance criteria such as timing, area and circuit robustness are considered. A number of benchmarked DSP circuits were initially considered. It was shown that by using larger functional logic macros as building blocks EHW is more successful at generating circuit solutions than if only gate primitives are used. In addition, the circuits generated are of comparable or better performance than equivalent circuits developed using a standard digital design methodology. Results also indicated that for more complex DSP functions to be generated, EHW platforms must use larger functional blocks, constrained for a specific application. Finite Impulse Response (FIR) filters were identified as the backbone of many DSP applications, and the multiplication unit was targeted as the performance critical component. A novel Programmable Arithmetic Logic Unit (PALU) was therefore developed as a functional building block suitable for automated digital filter design using EHW. The PALU replaces coefficient multiplication with a series of bit-shifts, additions and subtractions. Two distinct arrays of PALU were developed based on conventional FPGA and PLA re-configurable hardware architectures. Results show that a PLA architecture with 2 levels of hierarchical interconnect and column-based fixed tap outputs provides a platform most suited to automated filter design using the EHW technique. The PLA was also shown to be robust to faults covering up to 25% of the array when configured using EHW.
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Fatine, Steven Carleton University Dissertation Engineering Electronics. "Design and VLSI implementation of CMOS decimation and interpolation half-band FIR digital filters." Ottawa, 1996.
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Rodney, David M. "Digital Channelized Wide Band Receiver Implemented with a Systolic Array of Multi-Rate FIR Filters." Wright State University / OhioLINK, 2006. http://rave.ohiolink.edu/etdc/view?acc_num=wright1150923373.
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Walters, Allison L. "A Scaleable FIR Filter Implementation Using 32-bit Floating-Point Complex Arithmetic on a FPGA Base Custom Computing Platform." Thesis, Virginia Tech, 1998. http://hdl.handle.net/10919/35765.
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This thesis presents a linear phase finite impulse response filter implementation developed on a custom computing platform called WILDFORCE. The work has been motivated by ways to off-load intensive computing tasks to hardware for indoor communications channel modeling. The design entails complex convolution filters with customized lengths that can support channel impulse response profiles generated by SIRCIM. The paper details the partitioning for a fully pipelined convolution algorithm onto field programmable gate arrays through VHDL synthesis. Using WILDFORCE, the filter can achieve calculations at 160 MFLOPs/s.<br>Master of Science
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Bae, Cheolyong, and Madhur Gokhale. "Implementation of High-Speed 512-Tap FIR Filters for Chromatic Dispersion Compensation." Thesis, Linköpings universitet, Datorteknik, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-153435.
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A digital filter is a system or a device that modifies a signal. This is an essential feature in digital communication. Using optical fibers in the communication has various advantages like higher bandwidth and distance capability over copper wires. However, at high-rate transmission, chromatic dispersion arises as a problem to be relieved in an optical communication system. Therefore, it is necessary to have a filter that compensates chromatic dispersion. In this thesis, we introduce the implementation of a new architecture of the filter and compare it with a previously proposed architecture.
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Karlsson, Magnus. "Implementation of digit-serial filters." Doctoral thesis, Linköpings universitet, Institutionen för systemteknik, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-3520.
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In this thesis we discuss the design and implementation of Digital Signal Processing (DSP) applications in a standard digital CMOS technology. The aim is to fulfill a throughput requirement with lowest possible power consumption. As a case study a frequency selective filter is implemented using a half-band FIR filter and a bireciprocal Lattice Wave Digital Filter (LWDF) in a 0.35 µm CMOS process. The thesis is presented in a top-down manner, following the steps in the topdown design methodology. This design methodology, which has been used for bit-serial maximally fast implementations of IIR filters in the past, is here extended and applied for digit-serial implementations of recursive and non-recursive algorithms. Transformations such as pipelining and unfolding for increasing the throughput is applied and compared from throughput and power consumption points of view. A measure of the level of the logic pipelining is developed, i.e., the Latency Model (LM), which is used as a tuning variable between throughput and power consumption. The excess speed gained by the transformations can later be traded for low power operation by lowering the supply voltage, i.e., architecture driven voltage scaling. In the FIR filter case, it is shown that for low power operation with a given throughput requirement, that algorithm unfolding without pipelining is preferable. Decreasing the power consumption with 40, and 50 percent compared to pipelining at the logic or algorithm level, respectively. The digit-size should be tuned with the throughput requirement, i.e., using a large digit-size for low throughput requirement and decrease the digit-size with increasing throughput. In the bireciprocal LWDF case, the LM order can be used as a tuning variable for a trade-off between low energy consumption and high throughput. In this case using LM 0, i.e., non-pipelined processing elements yields minimum energy consumption and LM 1, i.e., use of pipelined processing elements, yields maximum throughput. By introducing some pipelined processing elements in the non-pipelined filter design a fractional LM order is obtained. Using three adders between every pipeline register, i.e., LM 1/3, yields a near maximum throughput and a near minimum energy consumption. In all cases should the digit-size be equal to the number of fractional bits in the coefficient. At the arithmetic level, digit-serial adders is designed and implemented in a 0.35 µm CMOS process, showing that for the digit-sizes, , the Ripple-Carry Adders (RCA) are preferable over Carry-Look-Ahead adders (CLA) from a throughput point of view. It is also shown that fixed coefficient digitserial multipliers based on unfolding of serial/parallel multipliers can obtain the same throughput as the corresponding adder in the digit-size range D = 2...4. A complex multiplier based on distributed arithmetic is used as a test case, implemented in a 0.8 µm CMOS process for evaluation of different logic styles from robustness, area, speed, and power consumption points of view. The evaluated logic styles are, non-overlapping pseudo two-phase clocked C2MOS latches with pass-transistor logic, Precharged True Single Phase Clocked logic (PTSPC), and Differential Cascade Voltage Switch logic (DCVS) with Single Transistor Clocked (STC) latches. In addition we propose a non-precharged true single phase clocked differential logic style, which is suitable for implementation of robust, high speed, and low power arithmetic processing elements, denoted Differential NMOS logic (DN-logic). The comparison shows that the two-phase clocked logic style is the best choice from a power consumption point of view, when voltage scaling can not be applied and the throughput requirement is low. However, the DN-logic style is the best choice when the throughput requirements is high or when voltage scaling is used.
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Mousa, Wail Abdul-Hakim. "Design &implementation of complex-valued FIR digital filters with application to migration of seismic data." Thesis, University of Leeds, 2006. http://etheses.whiterose.ac.uk/4712/.
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One-dimensional (I-D) and two-dimensional (2-D) frequency-space seismic migration FIR digital filter coefficients are of complex values when such filters require special space domain as well as wavenumber domain characteristics. In this thesis, such FIR digital filters are designed using Vector Space Projection Methods (VSPMs), which can satisfy the desired predefined filters' properties, for 2-D and three-dimensional (3-D) seismic data sets, respectively. More precisely, the pure and the relaxed projection algorithms, which are part of the VSPM theory, are derived. Simulation results show that the relaxed version of the pure algorithm can introduce significant savings in terms of the number of iterations required. Also, due to some undesirable background artifacts on migrated sections, a modified version of the pure algorithm was used to eliminate such effects. This modification has also led to a significant reduction in the number of computations when compared to both the pure and relaxed algorithms. We further propose a generalization of the l-D (real/complex-valued) pure algorithm to multi-dimensional (m-D) complex-valued FIR digital filters, where the resulting frequency responses possess an approximate equiripple nature. Superior designs are obtained when compared with other previously reported methods. In addition, we also propose a new scheme for implementing the predesigned 2-D migration FIR filters. This realization is based on Singular Value Decomposition (SVD). Unlike the existing realization methods which are used for this geophysical application, this cheap realization via SVD, compared with the true 2-D convolution, results in satisfactory wavenumber responses. Finally, an application to seismic migration of 2-D and 3-D synthetic sections is shown to confirm our theoretical conclusions. The proposed resulting migration FIR filters are applied also to the challenging SEGIEAGE Salt model data. The migrated section (image) outperformed images obtained using other FIR filters and with other standard migration techniques where difficult structures contained in such a challenging model are imaged clearly.
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Sweeney, Paul. "THE NEXT GENERATION AIRBORNE DATA ACQUISITION SYSTEMS. PART 1 - ANTI-ALIASING FILTERS: CHOICES AND SOME LESSONS LEARNED." International Foundation for Telemetering, 2003. http://hdl.handle.net/10150/605378.
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International Telemetering Conference Proceedings / October 20-23, 2003 / Riviera Hotel and Convention Center, Las Vegas, Nevada<br>The drive towards higher accuracy and sampling rates has raised the bar for modern FTI signal
conditioning. This paper focuses on the issue of anti-alias filtering. Today's 16-bit (and greater
resolution) ADC’s, coupled with the drive for optimum sampling rates, means that filters have to be
more accurate and yet more flexible than ever before. However, in order to take full advantage of
these advances, it is important to understand the trade-offs involved and to correctly specify the
system filtering requirements.
Trade-offs focus on:
• Analog vs. Digital signal conditioning
• FIR vs. IIR Digital Filters
• Signal bandwidth vs. Sampling rate
• Coherency issues such as filter phase distortion vs. delay
This paper will discuss each of these aspects. In particular, it will focus on some of the advantages
of digital filtering various analog filter techniques. This paper will also look at some ideas for
specifying filter cut-off and characteristics.
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Norén, Andreas. "All-Digital Aggregator for Multi-Standard Video Distribution." Thesis, Linköpings universitet, Kommunikationssystem, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-149301.
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In video transmission there is a need to compose a wide-band signal from a numberof narrow-band sub-signals. A flexible solution offers the possibility to place any narrow-band sub-signal anywhere in the wide-band signal, making better use of the frequency space of the wide-band signal. A multi-standard supportive solution will also consider the three standard bandwidths of digital and analog video transmissions, both terrestrial and cable (6; 7 and 8 MHz), in use today. This thesis work will study the efficiency of a flexible aggregation solution, in terms of computational complexity and error vector magnitude (EVM). The solution uses oversampled complex modulated filter banks and inner channelizers, to reduce the total workload on the system. Each sub-signal is channelized through an analysis filter bank and together all channelized sub-signals are aggregated through one synthesis filter bank to form the wide-band composite signal. The EVM between transmitted and received sub-signals are investigated for an increasing number of sub-signals. The solution in this thesis work is performing good for the tested number of up to 100 narrow-band sub-signals. The result indicates that the multi-standard flexible aggregation solution is efficient for an increasing number of transmitted sub-signals.
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Rachamadugu, Arun. "Digital implementation of high speed pulse shaping filters and address based serial peripheral interface design." Thesis, Atlanta, Ga. : Georgia Institute of Technology, 2008. http://hdl.handle.net/1853/26603.
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Thesis (M. S.)--Electrical and Computer Engineering, Georgia Institute of Technology, 2009.<br>Committee Chair: Laskar, Joy; Committee Member: Anderson, David; Committee Member: Cressler, John. Part of the SMARTech Electronic Thesis and Dissertation Collection.
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Akyurek, Sefa. "The Implementation Complexity Of Finite Impulse Response Digital Filters Under Different Coefficient Quantization Schemes And Realization Structures." Master's thesis, METU, 2004. http://etd.lib.metu.edu.tr/upload/2/12605586/index.pdf.
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It has been aimed to investigate the complexity of discrete-coefficient FIR filters when they are implemented in transposed form and the coefficient redundancy is removed by the n-Dimensional Reduced Adder Graph (RAG-n) approach. Filters with coefficients represented by different quantization schemes have been designed or selected from the literture<br>their transposed form implemetations after RAG-n process have been compared in terms of complexity. A Genetic Algorithm (GA) based design algorithm has been implemented and used for the design of integer coefficient filters. Algorithms for the realization of filter coefficients in Canonic Signed Digit (CSD) form and realization of n-Dimensional Reduced Adder Graph (RAG-n) have also been implemented. Filter performance is measured as Normalized Peak Ripple Magnitude and implementation complexity as the number of adders used to implement filter coefficients. Number of adders used to implement filter coefficients is calculated by using two different methods: CSD and RAG-n. RAG-n method has been applied to FIR digital filter design methods that don&rsquo<br>t
consider reduction of implementation complexity via RAG-n with transposed direct form filter structure. For implementation complexity, it is concluded that
&ldquo<br>RAG-n algorithm with transposed direct form filter structure&rdquo<br>provides better results over the &ldquo<br>CSD, SPT coefficient design followed by transposed direct form filter structure&rdquo<br>in terms of number of adders used in the implementation.
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Johansson, Kenny. "Low Power and Low complexity Constant Multiplication using Serial Arithmetic." Licentiate thesis, Linköping : Department of Electrical Engineering, Linköpings universitet, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-7965.
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Johansson, Kenny. "Low Power and Low Complexity Shift-and-Add Based Computations." Doctoral thesis, Linköping : Department of Electrical Engineering, Linköping University, 2008. http://www.bibl.liu.se/liupubl/disp/disp2008/tek1201s.pdf.
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Filip, Silviu-Ioan. "Robust tools for weighted Chebyshev approximation and applications to digital filter design." Thesis, Lyon, 2016. http://www.theses.fr/2016LYSEN063/document.
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De nombreuses méthodes de traitement du signal reposent sur des résultats puissants d'approximation numérique. Un exemple significatif en est l'utilisation de l'approximation de type Chebyshev pour l'élaboration de filtres numériques.En pratique, le caractère fini des formats numériques utilisés en machine entraîne des difficultés supplémentaires pour la conception de filtres numériques (le traitement audio et le traitement d'images sont deux domaines qui utilisent beaucoup le filtrage). La majorité des outils actuels de conception de filtres ne sont pas optimisés et ne certifient pas non plus la correction de leurs résultats. Notre travail se veut un premier pas vers un changement de cette situation.La première partie de la thèse traite de l'étude et du développement de méthodes relevant de la famille Remez/Parks-McClellan pour la résolution de problèmes d'approximation polynomiale de type Chebyshev, en utilisant l'arithmétique virgule-flottante.Ces approches sont très robustes, tant du point de vue du passage à l'échelle que de la qualité numérique, pour l'élaboration de filtres à réponse impulsionnelle finie (RIF).Cela dit, dans le cas des systèmes embarqués par exemple, le format des coefficients du filtre qu'on utilise en pratique est beaucoup plus petit que les formats virgule flottante standard et d'autres approches deviennent nécessaires.Nous proposons une méthode (quasi-)optimale pour traîter ce cas. Elle s'appuie sur l'algorithme LLL et permet de traiter des problèmes de taille bien supérieure à ceux que peuvent traiter les approches exactes. Le résultat est ensuite utilisé dans une couche logicielle qui permet la synthèse de filtres RIF pour des circuits de type FPGA.Les résultats que nous obtenons en sortie sont efficaces en termes de consommation d'énergie et précis. Nous terminons en présentant une étude en cours sur les algorithmes de type Remez pour l'approximation rationnelle. Ce type d'approches peut être utilisé pour construire des filtres à réponse impulsionnelle infinie (RII) par exemple. Nous examinons les difficultés qui limitent leur utilisation en pratique<br>The field of signal processing methods and applications frequentlyrelies on powerful results from numerical approximation. One suchexample, at the core of this thesis, is the use of Chebyshev approximationmethods for designing digital filters.In practice, the finite nature of numerical representations adds an extralayer of difficulty to the design problems we wish to address using digitalfilters (audio and image processing being two domains which rely heavilyon filtering operations). Most of the current mainstream tools for thisjob are neither optimized, nor do they provide certificates of correctness.We wish to change this, with some of the groundwork being laid by thepresent work.The first part of the thesis deals with the study and development ofRemez/Parks-McClellan-type methods for solving weighted polynomialapproximation problems in floating-point arithmetic. They are veryscalable and numerically accurate in addressing finite impulse response(FIR) design problems. However, in embedded and power hungry settings,the format of the filter coefficients uses a small number of bits andother methods are needed. We propose a (quasi-)optimal approach basedon the LLL algorithm which is more tractable than exact approaches.We then proceed to integrate these aforementioned tools in a softwarestack for FIR filter synthesis on FPGA targets. The results obtainedare both resource consumption efficient and possess guaranteed accuracyproperties. In the end, we present an ongoing study on Remez-type algorithmsfor rational approximation problems (which can be used for infinite impulseresponse (IIR) filter design) and the difficulties hindering their robustness
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Gardos, Thomas R. "Analysis and design of multidimensional FIR filter banks." Diss., Georgia Institute of Technology, 1993. http://hdl.handle.net/1853/15621.
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Dallmeyer, Matthew John. "Reducing Fir Filter Costs: A Review of Approaches as Applied to Massive Fir Filter Arrays." University of Dayton / OhioLINK, 2014. http://rave.ohiolink.edu/etdc/view?acc_num=dayton1417544448.
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Wong, Ok Yuen Lily 1974. "A hybrid digital FIR lattice filter for PRML magnetic read channel." Thesis, Massachusetts Institute of Technology, 1998. http://hdl.handle.net/1721.1/49657.
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Thesis (S.B. and M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1998.<br>Includes bibliographical references (p. 117-121).<br>by Lily Wong Ok Yuen.<br>S.B.and M.Eng.
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Burich, Lawrence D. "Digital Wideband Spectral Sensing Receiver." Wright State University / OhioLINK, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=wright1345689526.
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Chabbi, Charef. "VLSI NMOS hardware design of a linear phase FIR low pass digital filter." Ohio University / OhioLINK, 1985. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1183749814.
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Schoerner, Sven-Markus, and Erik Zakrisson. "Audioeffects with digital soundprocessing." Thesis, Linköping University, Department of Electrical Engineering, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-3777.
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<p>To effectively demonstrate the strength of using digital signal processing when producing sound effects, a sound effects demo is used at the lectures of the course TSRT78, Digital signal processing, which is given at the university in Linköping.</p><p>The amount of effects, that in an instructive way can be used for an educational purpose, are many and the existing version of the sound effects demo is somewhat limited in its range of effects.</p><p>This reports main focus lies in the presentation of what kind of effects which can be interesting in this kind of demo. All of the effects are presented with their background theory and examples on how they can be implemented in software, mainly with the focus on MATLABTM. Investigations on how well the effects can be run in realtime, in the toolbox SimulinkTM, has been made.</p><p>In the report there is also a presentation of a new version of the sound effect demo that has been produced with user friendlieness and further updates in mind. In the new demo all of the effects are implemented, according to their presentations. The report finishes with suggestions for further work on the sound effects demo.</p>
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Grossman, Hy, and Steve Pellarin. "A Time Correlated Approach to Adaptable Digital Filtering." International Foundation for Telemetering, 2006. http://hdl.handle.net/10150/604156.
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ITC/USA 2006 Conference Proceedings / The Forty-Second Annual International Telemetering Conference and Technical Exhibition / October 23-26, 2006 / Town and Country Resort & Convention Center, San Diego, California<br>Signal conditioning is a critical element in all data telemetry systems. Data from all sensors
must be band limited prior to digitization and transmission to prevent the potentially
disastrous effects of aliasing. While the 6th order analog low-pass Butterworth filter has long
been the de facto standard for data channel filtering, advances in digital signal processing
techniques now provide a potentially better alternative.
This paper describes the challenges in developing a flexible approach to adaptable data
channel filtering using DSP techniques. Factors such as anti-alias filter requirements, time
correlated sampling, decimation and filter delays will be discussed. Also discussed will be
the implementation and relative merits and drawbacks of various symmetrical FIR and IIR
filters. The discussion will be presented from an intuitive and practical perspective as much
as possible.
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Rosa, Vagner Santos da. "Uma ferramenta para geração de filtros FIR paralelos otimizados com coeficientes constantes." reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2005. http://hdl.handle.net/10183/5661.
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Esta dissertação trata da elaboração de uma ferrramenta para a geração de filtros FIR otimizados paralelos com coeficientes constantes. A ferramenta desenvolvida é capaz de gerar uma descrição VHDL de um filtro FIR paralelo com coeficientes constantes a partir das especificações do filtro. São exploradas técnicas de otimização de coeficientes e de otimização arquitetural. As técnicas empregadas são baseadas no uso de representações ternárias e redução do número de digitos não-zero dos coeficientes, uso de fatores de escala e eliminação de sub-expressões comuns. No texto, uma breve introdução sobre os filtros digitais é apresentada seguida por uma série de trabalhos encontrados na literatura relacionados às técnicas mencionadas e que são apresentados como base para o desenvolvimento da ferramenta implementada nesta dissertação. O funcionamento da ferramenta é detalhado tanto nos seus aspectos de algoritmo quanto em nível de implementação. São apresentados resultados de síntese em alguns de filtros hipotéticos projetados utilizando a ferramenta desenvolvida. Uma análise detalhada dos resultados obtidos é realizada. Os apêndices deste trabalho apresentam o código fonte da ferramenta de síntese de filtros desenvolvida.
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Vykydal, Martin. "Zpracování signálu z digitálního mikrofonu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-218904.
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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.
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Liu, Qian. "Wideband Digital Filter-and-Sum Beamforming with Simultaneous Correction of Dispersive Cable and Antenna Effects." Diss., Virginia Tech, 2012. http://hdl.handle.net/10919/27738.
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Optimum filter-and-sum beamforming is useful for array systems that suffer from spatially correlated noise and interference over large
bandwidth. The set of finite impulse response (FIR) filter coefficients used to implement the optimum filter-and-sum beamformer are selected to optimize signal-to-noise ratio (SNR) and reduce
interference from the certain directions. However, these array systems may also be vulnerable to dispersion caused by physical components such as antennas and cables, especially when the dispersion is unequal between sensors. The unequal responses can be equalized by using FIR filters. Although the problems of optimum-SNR beamforming, interference mitigation, and per-sensor dispersion have previously been individually investigated, their combined effects
and strategies for mitigating their combined effects do not seem to have been considered.
In this dissertation, combination strategies for optimum filter-and-sum beamforming and sensor dispersion correction are investigated. Our objective is to simultaneously implement optimum
filter-and-sum beamforming and per-sensor dispersion correction using a single FIR filter per sensor. A contribution is to reduce overall filter length, possibly also resulting in a significant reduction in implementation complexity, power consumption, and cost.
Expressions for optimum filter-and-sum beamforming weights and per-sensor dedispersion filter coefficients are derived. One solution is found via minimax optimization. To assess feasibility, the cost is analyzed in terms of filter length. These designs are considered in the context of LWA1, the first ``station'' of the Long Wavelength Array (LWA) radio telescope, consisting of 512 bowtie-type antennas and operating at frequencies between 10 MHz and 88 MHz. However, this work is applicable to a variety of systems which suffer from non-white spatial noise and directional interference and are vulnerable to sensor dispersion; e.g., sonar
arrays, HF/VHF-band riometers, radar arrays, and other radio telescopes.<br>Ph. D.
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Pristach, Marián. "Návrh optimalizovaných architektur digitálních filtrů pro nízkopříkonové integrované obvody." Doctoral thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-234534.
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The doctoral thesis deals with development and design of novel architectures of digital filters for low-power integrated circuits. The main goal was to achieve optimum parameters of digital filters with respect to the chip area, power consumption and operating frequency. The target group of the proposed architectures are application specific integrated circuits designed for signal processing from sensors using delta-sigma modulators. Three novel architectures of digital filters optimized for low-power integrated circuits are presented in the thesis. The thesis provides analysis and comparison of parameters of the new filter architectures with the parameters of architectures generated by Matlab tool. A software tool has been designed and developed for the practical application of the proposed architectures of digital filters. The developed software tool allows generating hardware description of the filters with respect to defined parameters.
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Toman, Petr. "Návrh digitálního decimačního filtru v technologii CMOS." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-219227.
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This Master’s thesis deals with digital decimation filter design for undersampling and filtering of sigma-delta ADC signal. Filter cascade is designed in Matlab according to given requirements and is then described in VHDL language aiming for minimum area. Implemented filter functionality is compared to Matlab-generated reference filters in created verification environment. Finally the design is synthesized in specified technology and verified on gate level.
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Kovalev, Anton. "Implementation and Evaluation of Two 512-Tap Complex FIR Filter Architectures for Compensation of Chromatic Dispersion in Optical Networks." Thesis, Linköpings universitet, Institutionen för systemteknik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-143965.
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Filtering is an important part of digital processing, since the applications often require a change of features of a digital or analog signal. A digital filter is a device or a system that removes or alters certain parts of a signal. Optical fibers are used to transmit information over longer distances and at higher bandwidths than traditional copper cables. In order to enable high-rate transmission in optical communication systems, it is necessary to have a filter that compensates for chromatic dispersion in optic links, since the dispersion alters the signal in an unwanted way. This thesis presents the implementation and evaluation of two filter architectures, used in fiber-optic communication. The clock frequency of the implemented designs reaches 475 MHz, which results in a processing speed of 60 GS/s.
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Lindahl, Erik. "Design and implementation of a decimation filter using a multi-precision multiply and accumulate unit for an audio range delta sigma analog to digital converter." Thesis, Linköping University, Department of Electrical Engineering, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-11261.
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<p>This work presents the design and implementation of a decimation filter for a three bits sigma delta analog to digital converter. The input is audio with a oversampling ratio of 32. Filter optimization and tradeoffs concerning the design is described. The filter is a multistage filter consisting of two cascaded FIR filters. The arithmetic unit is a multi-precision unit that can handle three or 24 bits MAC operations. The designed decimation filter is synthesized on standard cells of a 0.13 μm CMOS library.</p>
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Brito, Luís Filipe Ferrão. "Compensação digital da dispersão cromática." Master's thesis, Universidade de Aveiro, 2014. http://hdl.handle.net/10773/14852.
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Mestrado em Engenharia Eletrónica e Telecomunicações<br>O mundo globalizado atual tem originado um aumento ininterrupto do tráfego de Internet pressionando os limites de capacidade da rede instalada. Esta utiliza ainda em grande parte dos casos sistemas de modulação na intensidade/deteção direta capazes de codificar informação apenas na intensidade do sinal ótico o que implica densidades espetrais reduzidas. Os sistemas coerentes representam uma mais valia em termos de aumento da densidade espetral uma vez que utilizam dois graus de liberdade (amplitude e fase) permitindo assim uma melhor utilização dos recursos disponíveis e uma maior capacidade de transportar informação. Este sistemas permitem o uso de técnicas de processamento digital de sinal tais como: recuperação da portadora, estimação de fase e desmultiplexagem da polarização. Adicionalmente, estas técnicas tornam ainda possível a pós-compensação digital dos efeitos dispersivos que afetam o sinal, evitando assim a utilização de métodos de compensação óticos que são ineficientes para as exigências atuais e requerem esquemas de amplificação complexos e dispendiosos. Este trabalho tem por objetivo central estudar os efeitos da dispersão cromática em sinais coerentes e implementar um algoritmo, que recorrendo a um filtro de resposta impulsionai finita, seja capaz de efetuar a sua compensação no domínio do tempo. Outro aspeto relevante prende-se com a otimização da performance do filtro através da escolha justificada e ponderada dos vários parâmetros do projeto. Entre os principais referem-se pela sua importância: o estudo do número de coeficientes necessários, a distribuição do número de bits entre parte inteira e fracionária ao longo dos vários blocos funcionais e a ocupação de recursos na FPGA. Utilizaram-se as plataformas de edição e simulação da Xilinx (ISE e ISim) bem como a aplicação Matlab®. Por último procedeu-se à implementação do algoritmo em tempo real, utilizando uma placa Virtex-6 FPGA ML605.<br>The current globalized world has led to a continuous increase of the Internet traffic pushing the capacity limits of the installed network. In most cases the network still uses the intensity modulation/direct detection systems, that are able to encode information only in the intensity of the optical signal, which implies reduced spectral densities. Optical Coherent systems represent a huge advantage in terms of increased spectral density, since they use two degrees of freedom (amplitude and phase) thus allowing better use of the available resources and a greater capability to carry information. These systems allow the use of digital signal processing techniques, such as: carrier recovering, phase estimation and polarization multiplexing. Additionally, these techniques make possible the compensation of chromatic dispersion, which effects degrades the signal, thus avoiding the use of optical compensation methods which are inefficient to the current demands and requiring complex and expensive amplification schemes. This work has as main objective to study the effects of chromatic dispersion in coherent signals and implement an algorithm that, by using a finite impulse response filter, will be able to make the signal compensation in time domain. Another relevant topic is the optimization of the filter performance due to a justified reasonable choice of the main project parameter. The most important of these parameters are: the study of the required number of coefficients, the appropriate distribution of number of bits between the whole and the fractional parts, through several functional blocks and the resources occupation in FPGA. Were used the editing and simulation platforms of Xilinx (ISE and ISIM) and the application Matlab ®. Finally the implementation of the algorithm proceeded in real-time using a Virtex-6 FPGA ML605 board.
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Vásquez, Pérez Franck Paul. "Interpolación del Patrón del Haz Acústico Omnidireccional utilizando Filtros Digitales FIR invariables en el tiempo para Sonares Pasivos con Arreglo Cilíndrico de Hidrófonos." Bachelor's thesis, Universidad Ricardo Palma. Programa Cybertesis PER, 2008. http://cybertesis.urp.edu.pe/urp/2008/vasquez_fp/html/index-frames.html.
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MELLO, JÚNIOR Harold Dias de. "Caracterização de canais multipercurso utilizando filtros digitais parametrizados com técnicas de traçado de raios." Universidade Federal do Pará, 2006. http://repositorio.ufpa.br/jspui/handle/2011/7185.
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Previous issue date: 2006-06-23<br>CNPq - Conselho Nacional de Desenvolvimento Científico e Tecnológico<br>CAPES - Coordenação de Aperfeiçoamento de Pessoal de Nível Superior<br>Neste trabalho, o canal rádio-móvel com efeito de multipercurso é caracterizado por um filtro de resposta finita ao impulso (FIR) cujos parâmetros são obtidos a partir de técnicas de traçado de raios. Como exemplo de aplicação deste tipo de abordagem, simulou-se o comportamento de um canal em um cenário exterior (outdoor) definido, derivando-se a resposta impulsiva a partir da resposta em freqüência, sendo esse resultado representado por um filtro FIR. Aspectos relacionados á dispersividade temporal e ao desenvolvimento também são discutidos.<br>In this work, multipath radio propagation channels are characterized by FIR filters. The filters parameters are obtained from ray-tracing techniques, jointly with electromagnetic theory needed to is application. As study of case, modeling of an outdoor wideband channel impulse response is undertaken and time dispersion issues are discussed.
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MUCOLLARI, Irena. "K-edge filter subtraction technique used for mapping elemental distribution on paintings." Doctoral thesis, Università degli studi di Ferrara, 2014. http://hdl.handle.net/11392/2388940.
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Abstract
Several techniques exist for mapping the element pigments on painting using the synchrotron light or quasi monochromatic sources. K-edge subtraction technique is a well known technique and based on discontinuity in the attenuation coefficient due to photoelectric K-edge of the absorbing materials. A couple of filters with slightly different K -edge energies, if are with balanced thicknesses, can isolate spectra in a narrow energy band. Quasi monoenergetic X-rays with a spectral width Ek that is the difference in K-edge energies will be taken by subtracting images or spectral data.
Thus by using filter foils with appropriate thicknesses with the K-edge absorption just below and above the K-edge energy of an element allows us evaluating the distribution of that element in whole painting.
The aim of this thesis has been application of K-edge subtraction technique to determine pigment composition in paintings by using common X-ray sources with balanced filters.
The K-edge digital subtraction technique actually is investigated for estimation of Cadmium element. By choosing the Silver Cadmium and Indium as filter materials with properly thickness and with K-edge energies very close to the Cadmium element, determination of cadmium content has been presented as qualitative and quantitative result. The technique is tested by using a RadEye200 CMOS sensor. Theoretical simulations considering the experimental setup of technique has been presented as preliminary investigation. Complementary analyses are preformed with micro X-ray fluorescence spectrometry (μXRF), in order to validate the technique for cadmium element with the same samples.
Theoretical aspects of K-edge imaging, the algorithms used for elaboration of images, detailed information for cadmium pigments and samples tests, are described in first chapters. An evaluation of X-ray performance of RadEye200 CMOS sensor has been performed in terms of MTF, NPS and DQE.
Preliminary theoretical simulation is performed to determine the range where KES imaging technique with balanced filters response linear with the element content in paintings.
In chapter 5 after description of spectral characteristics of the tungsten X-ray tube with additional K-edge filtering, imaging and steps of elaborating images has been described. Determination of Cadmium element in its pigments it is performed by KES and Lehmann algorithm applied on digital images. Results obtained on different samples are presented by a couple of images, graphically, tabulated data, and discussed.
In chapter 6 introduction in theoretical aspects of XRF spectrometry is presented. Measurements for sampling area of μXRF ArtaxBruker200 and for Cadmium content in same samples used for imaging are presented.
In chapter 7 the results carry out from KES technique and XRF for cadmium element in different samples are compared. The comparison between KES and XRF technique performed with the same sampling area of two detection systems (CMOS sensor with SDD solid drift detector) on the same area of the test sample shows a very good correlation among them.
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Joakim, Holmlund. "IP block signalbehandling." Thesis, Uppsala universitet, Institutionen för elektroteknik, 2021. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-447850.
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The thesis aims to implement different digital filters such as finite impulse response (FIR), infinite impulse response (IIR) and cascade integrator comb (CIC) on the field-programmable gate array (FPGA) development board using hardware description language (VHDL). To this purpose, Intel’s systems integration tool Platform designer is used to convert the implementation to an IP core. The implemented FIR and IIR filters include different filter types such as lowpass, highpass, bandpass and bandstop. All the filters have a pipeline architecture as well as adjustable parameters such as filter order, frequency specifications and resolution. The coefficients of the filters are calculated according to the user's specifications. The calculated coefficients are verified using simulation. Furthermore the IP has been validated on hardware by the FPGA board MAX DE-10 lite. The IP is also analyzed regarding timing and power consumtion with good results. FIR filters of different types have been implemented and tested up to 501 taps with a coefficient width of 24 bits, which covered just below 50% of the available logic gates on the MAX 10-DE lite board with 50000 gates in total. The FIR filters have an option to be used with a Kaiser window with a maximum tap level of 51. Different IIR filters have also been implemented and tested on the hardware. However, the results have shown that the IIR filters do not perform so well, especially those of order higher than 6. One of the main reasons for this is the overflow caused by instability of the IIR.
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Ryšavý, Marek. "Zlomkooktává analýza akustických signálů." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2016. http://www.nusl.cz/ntk/nusl-241138.
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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.
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Zadeh, Ramin Agha. "Performance control of distributed generation using digital estimation of signal parameters." Thesis, Queensland University of Technology, 2010. https://eprints.qut.edu.au/47011/1/Ramin_Agha_Zadeh_Thesis.pdf.
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The Queensland University of Technology (QUT) allows the presentation of a thesis for the Degree of Doctor of Philosophy in the format of published or submitted papers, where such papers have been published, accepted or submitted during the period of candidature. This thesis is composed of seven published/submitted papers, of which one has been published, three accepted for publication and the other three are under review. This project is financially supported by an Australian Research Council (ARC) Discovery Grant with the aim of proposing strategies for the performance control of Distributed Generation (DG) system with digital estimation of power system signal parameters.
Distributed Generation (DG) has been recently introduced as a new concept for the generation of power and the enhancement of conventionally produced electricity. Global warming issue calls for renewable energy resources in electricity production. Distributed generation based on solar energy (photovoltaic and solar thermal), wind, biomass, mini-hydro along with use of fuel cell and micro turbine will gain substantial momentum in the near future. Technically, DG can be a viable solution for the issue of the integration of renewable or non-conventional energy resources. Basically, DG sources can be connected to local power system through power electronic devices, i.e. inverters or ac-ac converters. The interconnection of DG systems to power system as a compensator or a power source with high quality performance is the main aim of this study. Source and load unbalance, load non-linearity, interharmonic distortion, supply voltage distortion, distortion at the point of common coupling in weak source cases, source current power factor, and synchronism of generated currents or voltages are the issues of concern.
The interconnection of DG sources shall be carried out by using power electronics switching devices that inject high frequency components rather than the desired current. Also, noise and harmonic distortions can impact the performance of the control strategies. To be able to mitigate the negative effect of high frequency and harmonic as well as noise distortion to achieve satisfactory performance of DG systems, new methods of signal parameter estimation have been proposed in this thesis. These methods are based on processing the digital samples of power system signals. Thus, proposing advanced techniques for the digital estimation of signal parameters and methods for the generation of DG reference currents using the estimates provided is the targeted scope of this thesis. An introduction to this research – including a description of the research problem, the literature review and an account of the research progress linking the research papers – is presented in Chapter 1.
One of the main parameters of a power system signal is its frequency. Phasor Measurement (PM) technique is one of the renowned and advanced techniques used for the estimation of power system frequency. Chapter 2 focuses on an in-depth analysis conducted on the PM technique to reveal its strengths and drawbacks. The analysis will be followed by a new technique proposed to enhance the speed of the PM technique while the input signal is free of even-order harmonics. The other techniques proposed in this thesis as the novel ones will be compared with the PM technique comprehensively studied in Chapter 2.
An algorithm based on the concept of Kalman filtering is proposed in Chapter 3. The algorithm is intended to estimate signal parameters like amplitude, frequency and phase angle in the online mode. The Kalman filter is modified to operate on the output signal of a Finite Impulse Response (FIR) filter designed by a plain summation. The frequency estimation unit is independent from the Kalman filter and uses the samples refined by the FIR filter. The frequency estimated is given to the Kalman filter to be used in building the transition matrices. The initial settings for the modified Kalman filter are obtained through a trial and error exercise.
Another algorithm again based on the concept of Kalman filtering is proposed in Chapter 4 for the estimation of signal parameters. The Kalman filter is also modified to operate on the output signal of the same FIR filter explained above. Nevertheless, the frequency estimation unit, unlike the one proposed in Chapter 3, is not segregated and it interacts with the Kalman filter. The frequency estimated is given to the Kalman filter and other parameters such as the amplitudes and phase angles estimated by the Kalman filter is taken to the frequency estimation unit.
Chapter 5 proposes another algorithm based on the concept of Kalman filtering. This time, the state parameters are obtained through matrix arrangements where the noise level is reduced on the sample vector. The purified state vector is used to obtain a new measurement vector for a basic Kalman filter applied. The Kalman filter used has similar structure to a basic Kalman filter except the initial settings are computed through an extensive math-work with regards to the matrix arrangement utilized.
Chapter 6 proposes another algorithm based on the concept of Kalman filtering similar to that of Chapter 3. However, this time the initial settings required for the better performance of the modified Kalman filter are calculated instead of being guessed by trial and error exercises. The simulations results for the parameters of signal estimated are enhanced due to the correct settings applied. Moreover, an enhanced Least Error Square (LES) technique is proposed to take on the estimation when a critical transient is detected in the input signal. In fact, some large, sudden changes in the parameters of the signal at these critical transients are not very well tracked by Kalman filtering. However, the proposed LES technique is found to be much faster in tracking these changes. Therefore, an appropriate combination of the LES and modified Kalman filtering is proposed in Chapter 6. Also, this time the ability of the proposed algorithm is verified on the real data obtained from a prototype test object.
Chapter 7 proposes the other algorithm based on the concept of Kalman filtering similar to those of Chapter 3 and 6. However, this time an optimal digital filter is designed instead of the simple summation FIR filter. New initial settings for the modified Kalman filter are calculated based on the coefficients of the digital filter applied. Also, the ability of the proposed algorithm is verified on the real data obtained from a prototype test object.
Chapter 8 uses the estimation algorithm proposed in Chapter 7 for the interconnection scheme of a DG to power network. Robust estimates of the signal amplitudes and phase angles obtained by the estimation approach are used in the reference generation of the compensation scheme. Several simulation tests provided in this chapter show that the proposed scheme can very well handle the source and load unbalance, load non-linearity, interharmonic distortion, supply voltage distortion, and synchronism of generated currents or voltages. The purposed compensation scheme also prevents distortion in voltage at the point of common coupling in weak source cases, balances the source currents, and makes the supply side power factor a desired value.
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Son, Han-Woong. "A Fully Integrated Fractional-N Frequency Synthesizer for Wireless Communications." Diss., Georgia Institute of Technology, 2004. http://hdl.handle.net/1853/5254.
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A fully integrated, fast-locking fractional-N frequency synthesizer is proposed and demonstrated in this work. In this design, to eliminate the need for large, inaccurate capacitors and resistors in a loop filter, an analog continuous-time loop filter whose performance is sensitive to process and temperature variations and aging has been replaced with a programmable digital Finite Impulse Response (FIR) filter. In addition, using the adaptive loop gain control proportional to the frequency difference, the frequency-locking time has been reduced. Also, the phase noise and spurs have been reduced by a Multi-stAge noise SHaping (MASH) controlled Fractional Frequency Detector (FFD) that generates a digital output corresponding directly to the frequency difference. The proposed frequency synthesizer provides many benefits in terms of high integration ability, technological robustness, fast locking time, low noise level, and multimode flexibility.
To prove performance of the proposed frequency synthesizer, the frequency synthesizers analysis, design, and simulation have been carried out at both the system and the circuit levels. Then, the performance was also verified after fabrication and packaging.
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Dommaraju, Sunny Raj. "Design and Implementation of a 16-Bit Flexible ROM-less Direct Digital Synthesizer in FPGA and CMOS 90nm Technology." Wright State University / OhioLINK, 2013. http://rave.ohiolink.edu/etdc/view?acc_num=wright1374351629.
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Colonna, Juan Gabriel, and (92) 98416-0589. "Uma abordagem para monitoramento de anuros baseada em processamento digital de sinais bioacústicos." Universidade Federal do Amazonas, 2017. http://tede.ufam.edu.br/handle/tede/6014.
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Previous issue date: 2017-09-15<br>CNPq - Conselho Nacional de Desenvolvimento Científico e Tecnológico<br>Wildlife monitoring is often used by biologists and ecologists to acquire information
about animals and their natural habitats. In survey programs, specialists collect environmental
information to infer about animal population status and their variations
over time. The main goal of such programs is to identify environmental problems in
early stages. However, acquiring the necessary data for this purpose is a manual work
and must be carried out by groups of experts in areas of di cult access during long
periods of time. In this context, Wireless Sensor Networks (WSNs) are useful alternatives
to alleviate the manual work. Such networks are made up of small sensors with
transmission, storage, and local processing capabilities. These networks enable bioacoustic
methods for automatic species recognition to be embedded in the sensor nodes
in order to automate and simplify the monitoring task. Since animal sounds usually
provide a species ngerprint, it can be used to recognize the presence or absence of a
target species in a site. Accordingly, in this thesis, we present an approach that combines
machine learning methods, WSNs and bioacoustic signal processing techniques for
wildlife monitoring based on animal calls. As a proof-of-concept, we choose anurans
as the target animals. The reason is that anurans are already used by biologists as an
early indicator of ecological stress, since they provide relevant information about terrestrian
and aquatic ecosystems. Our solution integrates four fundamental steps: noise
ltering and bioacoustic signal enhancement, automatic signal segmentation, acoustic
features extraction, and classi cation. We also consider the WSNs limitations, trying
to reduce the communication and processing load to extend the sensors' lifetime. To
accomplish with the restriction imposed by the hardware, we represent the acoustic
signals by a set of low-level acoustic descriptors (LLDs or features). This representation
allows us to identify speci c signal patterns of each species, reducing the amount
of information necessary to classify it. The adverse environmental conditions of the
rainforest pose additional challenges, such as noise ltering. We developed a ltering
method based on Singular Spectrum Analysis (SSA). This choice was based on several
comparisons with other ltering methods. The SSA method has additional advantages:
it is non-parametric, it adapts to the di erent input signals, and it has an equivalent<br>O monitoramento de animais silvestres em seu habitat natural é objeto de estudo de
biólogos e ecólogos que coletam informações ambientais para inferir o estado das popula
ções animais e suas variações ao longo do tempo. Um objetivo especí co desses
estudos é identi car problemas ecológicos em estágios iniciais. No entanto, a coleta
das informações é um trabalho manual que deve ser realizado por um grupo de especialistas
em áreas de difícil acesso durante períodos de tempo prolongados. Neste
contexto, as Redes de Sensores Sem Fio (RSSF) são uma alternativa viável ao monitoramento
manual. Estas redes são constituídas por pequenos sensores com capacidade
de transmissão, armazenamento e processamento local. Isto possibilita que métodos
bioacústicos para reconhecimento automático de espécies sejam embarcados nos nós
sensores para automatizar e simpli car a tarefa de monitoramento. Como os sons produzidos
pelos animais oferecem uma impressão digital bioacústica, esta pode ser usada
para identi car a presença ou ausência de uma espécie particular em uma região. Neste
trabalho, apresentamos uma abordagem que utiliza aprendizagem de máquina, RSSF e
processamento digital de sinais bioacústicos para reconhecer espécies animais com base
em suas vocalizações. Como prova-de-conceito, aplicamos nossa solução para identi-
car de forma automática diferentes espécies de anuros. Escolhemos anuros uma vez
que são utilizados como indicadores precoces de estresse ecológico, pelo fato de serem
sensíveis às mudanças do habitat e oferecerem informações sobre os ecossistemas terrestre
e aquático. Nossa abordagem integra quatro operações fundamentais: filtragem
de ruídos e aprimoramento dos sinais acústicos, segmentação automática desses sinais,
extração de descritores acústicos e classificação. Além disso, nossa solução considera
as limitações de RSSF, buscando reduzir a carga de processamento e comunicação
para prolongar o tempo de vida dos sensores. Portanto, representamos os sinais por
um conjunto de descritores acústicos de baixo nível (Low-Level Acoustic Descriptors
- LLDs) conhecidos como Mel Frequency Cepstral Coe cients (MFCCs). A técnica
escolhida para filtrar os ruídos ambientais foi o Singular Spectrum Analysis (SSA),
esta escolha foi baseada nas diversas comparações que zemos com outros métodos
de filtragem. Além disso, o SSA é não paramétrico, se adapta ao coaxar de cada espécie e possui um esquema equivalente na teoria de ltros FIR, o que possibilita ter
uma implementação com complexidade computacional constante. Ainda no método
de ltragem, desenvolvemos uma versão robusta do SSA. Esta nova versão é mais tolerante
aos diferentes ruídos ambientais, sejam estes Gaussianos ou não. A robustez
também permitiu identi car os componentes acústicos causados pelos ruídos ambientais
associados com as baixas frequências. No que diz respeito à segmentação, primeiro
realizamos uma comparação entre diferentes LLDs baseados na teoria da informação.
Nesta etapa, desenvolvemos um método não supervisionado capaz de se adaptar às diferentes
condições de ruídos ambientais, sejam estes branco ou coloridos. Na segunda
etapa, adaptamos dois dos LLDs comparados para funcionamento incremental. Assim,
foi possível de nir uma metodologia para segmentar os sinais acústicos em tempo real
com custo de memória constante, ideal para ser embarcado em um nó sensor de baixo
custo e obter as porções dos áudios que possuem as informações relevantes para o reconhecimento
das espécies. Finalmente, avaliamos diferentes estratégias de classi cação
e propusemos uma nova forma de validação cruzada para avaliar a capacidade de generaliza
ção do método. Portanto, a validação cruzada tradicional de sílaba-por-sílaba foi
substituída por uma validação cruzada que separa diferentes indivíduos nos conjuntos
de teste e treinamento. Isto viabilizou uma avaliação mais justa e permitiu estimar o
comportamento nal que o método de classi cação embarcado no nó sensor teria em
uma situação real. Dentre os métodos de classi cação planos comparados descobrimos
que SVM e kNN são os mais promissores. Todavia, propomos e desenvolvemos uma
estratégia de classi cação hierárquica multirótulo para decompor e simpli car o espaço
de decisões do classi cador e simultaneamente reconhecer a família, o gênero e a espécie
de cada amostra. Isto nos permite concluir que nossa abordagem é exível o su ciente
para se adaptar aos diferentes cenários monitorados, sem deixar de otimizar a relação
custo-benefício da solução de monitoramento proposta.