Design of doubly-complementary IIR digital filters using a single complex allpass filter, with multirate applications

It is shown that a large class of real-coefficient doubly-complementary IIR transfer function pairs can be implemented by means of a single complex allpass filter. For a real input sequence, the real part of the output sequence corresponds to the output of one of the transfer functions G(z) (for example, lowpass), whereas the imaginary part of the output sequence corresponds to its "complementary" filter H(z)(for example, highpass). The resulting implementation is structurally lossless, and hence the implementations of G(z) and H(z) have very low passband sensitivity. Numerical design examples are included, and a typical numerical example shows that the new implementation with 4 bits per multiplier is considerably better than a direct form implementation with 9 bits per multiplier. Multirate filter bank applications (quadrature mirror filtering) are outlined

Digital Filters

The new technology advances provide that a great number of system signals can be easily measured with a low cost. The main problem is that usually only a fraction of the signal is useful for different purposes, for example maintenance, DVD-recorders, computers, electric/electronic circuits, econometric, optimization, etc. Digital filters are the most versatile, practical and effective methods for extracting the information necessary from the signal. They can be dynamic, so they can be automatically or manually adjusted to the external and internal conditions. Presented in this book are the most advanced digital filters including different case studies and the most relevant literature

Generalized linear-in-parameter models : theory and audio signal processing applications

This thesis presents a mathematically oriented perspective to some basic concepts of digital signal processing. A general framework for the development of alternative signal and system representations is attained by defining a generalized linear-in-parameter model (GLM) configuration. The GLM provides a direct view into the origins of many familiar methods in signal processing, implying a variety of generalizations, and it serves as a natural introduction to rational orthonormal model structures. In particular, the conventional division between finite impulse response (FIR) and infinite impulse response (IIR) filtering methods is reconsidered. The latter part of the thesis consists of audio oriented case studies, including loudspeaker equalization, musical instrument body modeling, and room response modeling. The proposed collection of IIR filter design techniques is submitted to challenging modeling tasks. The most important practical contribution of this thesis is the introduction of a procedure for the optimization of rational orthonormal filter structures, called the BU-method. More generally, the BU-method and its variants, including the (complex) warped extension, the (C)WBU-method, can be consider as entirely new IIR filter design strategies.reviewe

DSP compensation for distortion in RF filters

There is a growing demand for the high quality TV programs such as High Definition TV (HDTV). The CATV network is often a suitable solution to address this demand using a CATV modem delivering high data rate digital signals in a cost effective manner, thereby, utilizing a complex digital modulation scheme is inevitable. Exploiting complex modulation schemes, entails a more sophisticated modulator and distribution system with much tighter tolerances. However, there are always distortions introduced to the modulated signal in the modulator degrading signal quality. In this research, the effect of distortions introduced by the RF band pass filter in the modulator will be considered which cause degradations on the quality of the output Quadrature Amplitude Modulated (QAM) signal. Since the RF filter's amplitude/group delay distortions are not symmetrical in the frequency domain, once translated into the base band they have a complex effect on the QAM signal. Using Matlab, the degradation effects of these distortions on the QAM signal such as Bit Error Rate (BER) is investigated. In order to compensate for the effects of the RF filter distortions, two different methods are proposed. In the first method, a complex base band compensation filter is placed after the pulse shaping filter (SRRC). The coefficients of this complex filter are determined using an optimization algorithm developed during this research. The second approach, uses a pre-equalizer in the form of a Feed Forward FIR structure placed before the pulse shaping filter (SRRC). The coefficients of this pre-equalizer are determined using the equalization algorithm employed in a test receiver, with its tap weights generating the inverse response of the RF filter. The compensation of RF filter distortions in base band, in turn, improves the QAM signal parameters such as Modulation Error Ratio (MER). Finally, the MER of the modulated QAM signal before and after the base band compensation is compared between the two methods, showing a significant enhancement in the RF modulator performance

Timed array antenna system : application to wideband and ultra-wideband beamforming receivers

Antenna array systems have a broad range of applications in radio frequency (RF) and ultra-wideband (UWB) communications to receive/transmit electromagnetic waves from/to the sky. They can enhance the amplitude of the input signals, steer beams electronically, and reject interferences thanks to beamforming technique. In an antenna array beamforming system, delay cells with the tunable capability of delay amount compensate the relative delay of signals received by antennas. In fact, each antenna almost acts individually depending upon time delaying effects on the input signals. As a result, the delay cells are the basic elements of the beamforming systems. For this purpose, novel active true time delay (TTD) cells suitable for RF antenna arrays have been presented in this thesis. These active delay cells are based on 1st- and 2nd-order all-pass filters (APFs) and achieve quite a flat gain and delay within up to 10-GHz frequency range. Various techniques such as phase linearity and delay tunability have been accomplished to improve the design and performance. The 1st-order APF has been designed for a frequency range of 5 GHz, showing desirable frequency responses and linearity which is comparable with the state-of-the-art. This 1st-order APF is able to convert into a 2nd-order APF via adding a grounded capacitor. A compact 2nd-order APF using an active inductor has been also designed and simulated for frequencies up to 10 GHz. The active inductor has been utilized to tune the amount of delay and to reduce the on-chip size of the filter. In order to validate the performance of the delay cells, two UWB four-channel timed array beamforming receivers realized by the active TTD cells have been proposed. Each antenna channel exploits digitally controllable gain and delay on the input signal and demonstrates desirable gain and delay resolutions. The beamforming receivers have been designed for different UWB applications depending on their operating frequency ranges (that is, 3-5 and 3.1-10.6 GHz), and thus they have different system requirements and specifications. All the circuits and topologies presented in this dissertation have been designed in standard 180-nm CMOS technologies, featuring a unity gain frequency ( ft) up to 60 GHz.Els sistemes matricials d’antenes tenen una àmplia gamma d’aplicacions en radiofreqüència (RF) i comunicacions de banda ultraampla (UWB) per rebre i transmetre ones electromagnètics. Poden millorar l’amplitud dels senyals d’entrada rebuts, dirigir els feixos electrònicament i rebutjar les interferències gràcies a la tècnica de formació de feixos (beamforming). En un sistema beamforming de matriu d’antenes, les cèl·lules de retard amb capacitat ajustable del retard, compensen aquest retard relatiu dels senyals rebuts per les diferents antenes. De fet, cada antena gairebé actua individualment depenent dels efectes de retard de temps sobre el senyals d’entrada. Com a resultat, les cel·les de retard són els elements bàsics en el disseny dels actuals sistemes beamforming. Amb aquest propòsit, en aquesta tesi es presenten noves cèl·lules actives de retard en temps real (TTD, true time delay) adequades per a matrius d’antenes de RF. Aquestes cèl·lules de retard actives es basen en cèl·lules de primer i segon ordre passa-tot (APF), i aconsegueixen un guany i un retard força plans, en el rang de freqüència de fins a 10 GHz. Diverses tècniques com ara la linealitat de fase i la sintonització del retard s’han aconseguit per millorar el disseny i el rendiment. La cèl·lula APF de primer ordre s’ha dissenyat per a un rang de freqüències de fins a 5 GHz, mostrant unes respostes freqüencials i linealitat que són comparables amb l’estat de l’art actual. Aquestes cèl·lules APF de primer ordre es poden convertir en un APF de segon ordre afegint un condensador més connectat a massa. També s’ha dissenyat un APF compacte de segon ordre que utilitza una emulació d’inductor actiu per a freqüències de treball de fins a 10 GHz. S’ha utilitzat l'inductor actiu per ajustar la quantitat de retard introduït i reduir les dimensions del filtre al xip. Per validar les prestacions de les cel·les de retard propostes, s’han proposat dos receptors beamforming basats en matrius d’antenes de 4 canals, realitzats por cèl·lules TTD actives. Cada canal d’antena aprofita el guany i el retard controlables digitalment aplicats al senyal d’entrada, i demostra resolucions de guany i retard desitjables. Els receptors beamforming s’han dissenyat per a diferents aplicacions UWB segons els seus rangs de freqüències de funcionament (en aquest cas, 3-5 i 3,1-10,6 GHz) i, per tant, tenen diferents requisits i especificacions de disseny del sistema. Tots els circuits i topologies presentats en aquesta tesi s’han dissenyat en tecnologies CMOS estàndards de 180 nm, amb una freqüència de guany unitari (ft) de fins a 60 GHz.Postprint (published version

Digital Filter Design Using Improved Artificial Bee Colony Algorithms

Digital filters are often used in digital signal processing applications. The design objective of a digital filter is to find the optimal set of filter coefficients, which satisfies the desired specifications of magnitude and group delay responses. Evolutionary algorithms are population-based meta-heuristic algorithms inspired by the biological behaviors of species. Compared to gradient-based optimization algorithms such as steepest descent and Newton’s like methods, these bio-inspired algorithms have the advantages of not getting stuck at local optima and being independent of the starting point in the solution space. The limitations of evolutionary algorithms include the presence of control parameters, problem specific tuning procedure, premature convergence and slower convergence rate. The artificial bee colony (ABC) algorithm is a swarm-based search meta-heuristic algorithm inspired by the foraging behaviors of honey bee colonies, with the benefit of a relatively fewer control parameters. In its original form, the ABC algorithm has certain limitations such as low convergence rate, and insufficient balance between exploration and exploitation in the search equations. In this dissertation, an ABC-AMR algorithm is proposed by incorporating an adaptive modification rate (AMR) into the original ABC algorithm to increase convergence rate by adjusting the balance between exploration and exploitation in the search equations through an adaptive determination of the number of parameters to be updated in every iteration. A constrained ABC-AMR algorithm is also developed for solving constrained optimization problems.There are many real-world problems requiring simultaneous optimizations of more than one conflicting objectives. Multiobjective (MO) optimization produces a set of feasible solutions called the Pareto front instead of a single optimum solution. For multiobjective optimization, if a decision maker’s preferences can be incorporated during the optimization process, the search process can be confined to the region of interest instead of searching the entire region. In this dissertation, two algorithms are developed for such incorporation. The first one is a reference-point-based MOABC algorithm in which a decision maker’s preferences are included in the optimization process as the reference point. The second one is a physical-programming-based MOABC algorithm in which physical programming is used for setting the region of interest of a decision maker. In this dissertation, the four developed algorithms are applied to solve digital filter design problems. The ABC-AMR algorithm is used to design Types 3 and 4 linear phase FIR differentiators, and the results are compared to those obtained by the original ABC algorithm, three improved ABC algorithms, and the Parks-McClellan algorithm. The constrained ABC-AMR algorithm is applied to the design of sparse Type 1 linear phase FIR filters of filter orders 60, 70 and 80, and the results are compared to three state-of-the-art design methods. The reference-point-based multiobjective ABC algorithm is used to design of asymmetric lowpass, highpass, bandpass and bandstop FIR filters, and the results are compared to those obtained by the preference-based multiobjective differential evolution algorithm. The physical-programming-based multiobjective ABC algorithm is used to design IIR lowpass, highpass and bandpass filters, and the results are compared to three state-of-the-art design methods. Based on the obtained design results, the four design algorithms are shown to be competitive as compared to the state-of-the-art design methods

A System Approach to the Design of Multirate Filter Banks.

This dissertation studies the design of multirate filter banks by adopting a so-called system approach. The design issue of Johnston\u27s method is first investigated in which an explicit expression of the reconstruction error is derived using Lyapunov stability theory, and new convergent iterative algorithms are proposed through non-linear optimization. The results are extended to the two-dimensional filter banks. The design issue of more general multirate filter banks is also investigated through model matching method. Using standard results from modern control theory, new design algorithms are developed which minimize the reconstruction error while completely eliminating the aliasing error. State-space realizations, inner-outer factorizations, and optimal Hankel norm approximation are used to reduce the complexity of computation and improve the accuracy of the proposed design algorithms

Reduced Receivers for Faster-than-Nyquist Signaling and General Linear Channels

Fast and reliable data transmission together with high bandwidth efficiency are important design aspects in a modern digital communication system. Many different approaches exist but in this thesis bandwidth efficiency is obtained by increasing the data transmission rate with the faster-than-Nyquist (FTN) framework while keeping a fixed power spectral density (PSD). In FTN consecutive information carrying symbols can overlap in time and in that way introduce a controlled amount of intentional intersymbol interference (ISI). This technique was introduced already in 1975 by Mazo and has since then been extended in many directions. Since the ISI stemming from practical FTN signaling can be of significant duration, optimum detection with traditional methods is often prohibitively complex, and alternative equalization methods with acceptable complexity-performance tradeoffs are needed. The key objective of this thesis is therefore to design reduced-complexity receivers for FTN and general linear channels that achieve optimal or near-optimal performance. Although the performance of a detector can be measured by several means, this thesis is restricted to bit error rate (BER) and mutual information results. FTN signaling is applied in two ways: As a separate uncoded narrowband communication system or in a coded scenario consisting of a convolutional encoder, interleaver and the inner ISI mechanism in serial concatenation. Turbo equalization where soft information in the form of log likelihood ratios (LLRs) is exchanged between the equalizer and the decoder is a commonly used decoding technique for coded FTN signals. The first part of the thesis considers receivers and arising stability problems when working within the white noise constraint. New M-BCJR algorithms for turbo equalization are proposed and compared to reduced-trellis VA and BCJR benchmarks based on an offset label idea. By adding a third low-complexity M-BCJR recursion, LLR quality is improved for practical values of M. M here measures the reduced number of BCJR computations for each data symbol. An improvement of the minimum phase conversion that sharpens the focus of the ISI model energy is proposed. When combined with a delayed and slightly mismatched receiver, the decoding allows a smaller M without significant loss in BER. The second part analyzes the effect of the internal metric calculations on the performance of Forney- and Ungerboeck-based reduced-complexity equalizers of the M-algorithm type for both ISI and multiple-input multiple-output (MIMO) channels. Even though the final output of a full-complexity equalizer is identical for both models, the internal metric calculations are in general different. Hence, suboptimum methods need not produce the same final output. Additionally, new models working in between the two extremes are proposed and evaluated. Note that the choice of observation model does not impact the detection complexity as the underlying algorithm is unaltered. The last part of the thesis is devoted to a different complexity reducing approach. Optimal channel shortening detectors for linear channels are optimized from an information theoretical perspective. The achievable information rates of the shortened models as well as closed form expressions for all components of the optimal detector of the class are derived. The framework used in this thesis is more general than what has been previously used within the area