This thesis studies the design and implementation of a linear minimum mean-square error (LMMSE) receiver in asynchronous bandlimited direct-sequence code-division multiple-access (DS-CDMA) systems that employ long-code pseudo-noise (PN) sequences and operate in multipath environments. The receiver is shown to be capable of multiple-access interference (MAI) suppression and multipath diversity combining without the knowledge of other users' signature sequences. It outperforms any other linear receiver by maximizing output signal-to-noise ratio (SNR) with the aid of a new chip filter which exploits the cyclostationarity of the received signal and combines all paths of the desired user that fall within its supported time span. This work is motivated by the shortcomings of existing LMMSE receivers which are either incompatible with long-code CDMA or constrained by limitations in the system model. The design methodology is based on the concept of linear/conjugate linear (LCL) filtering and satisfying the orthogonality conditions to achieve the LMMSE filter response. Moreover, the proposed LMMSE receiver addresses two drawbacks of the coherent Rake receiver, the industry's current solution for multipath reception. First, unlike the Rake receiver which uses the chip-matched filter (CMF) and treats interference as additive white Gaussian noise (AWGN), the LMMSE receiver suppresses interference by replacing the CMF with a new chip pulse filter. Second, in contrast to the Rake receiver which only processes a subset of strongest paths of the desired user, the LMMSE receiver harnesses the energy of all paths of the desired user that fall within its time support, at no additional complexity. The performance of the proposed LMMSE receiver is analyzed and compared with that of the coherent Rake receiver with probability of bit error, <i>Pe</i>, as the figure of merit. The analysis is based on the accurate improved Gaussian approximation (IGA) technique. Closed form conditional <i>Pe</i> expressions for both the LMMSE and Rake receivers are derived. Furthermore, it is shown that if quadriphase random spreading, moderate to large spreading factors, and pulses with small excess bandwidth are used, the widely-used standard Gaussian Approximation (SGA) technique becomes accurate even for low regions of <i>Pe</i>. Under the examined scenarios tailored towards current narrowband system settings, the LMMSE receiver achieves 60% gain in capacity (1. 8 dB in output SNR) over the selective Rake receiver. A third of the gain is due to interference suppression capability of the receiver while the rest is credited to its ability to collect the energy of the desired user diversified to many paths. Future wideband systems will yield an ever larger gain. Adaptive implementations of the LMMSE receiver are proposed to rid the receiver from dependence on the knowledge of multipath parameters. The adaptive receiver is based on a fractionally-spaced equalizer (FSE) whose taps are updated by an adaptive algorithm. Training-based, pilot-channel-aided (PCA), and blind algorithms are developed to make the receiver applicable to both forward and reverse links, with or without the presence of pilot signals. The blind algorithms are modified versions of the constant modulus algorithm (CMA) which has not been previously studied for long-code CDMA systems. Extensive simulation results are presented to illustrate the convergence behavior of the proposed algorithms and quantify their performance loss under various levels of MAI. Computational complexities of the algorithms are also discussed. These three criteria (performance loss, convergence rate, and computational complexity) determine the proper choice of an adaptive algorithm with respect to the requirements of the specific application in mind.

This thesis studies the design and implementation of a linear minimum mean-square error (LMMSE) receiver in asynchronous bandlimited direct-sequence code-division multiple-access (DS-CDMA) systems that employ long-code pseudo-noise (PN) sequences and operate in multipath environments. The receiver is shown to be capable of multiple-access interference (MAI) suppression and multipath diversity combining without the knowledge of other users' signature sequences. It outperforms any other linear receiver by maximizing output signal-to-noise ratio (SNR) with the aid of a new chip filter which exploits the cyclostationarity of the received signal and combines all paths of the desired user that fall within its supported time span. This work is motivated by the shortcomings of existing LMMSE receivers which are either incompatible with long-code CDMA or constrained by limitations in the system model. The design methodology is based on the concept of linear/conjugate linear (LCL) filtering and satisfying the orthogonality conditions to achieve the LMMSE filter response. Moreover, the proposed LMMSE receiver addresses two drawbacks of the coherent Rake receiver, the industry's current solution for multipath reception. First, unlike the Rake receiver which uses the chip-matched filter (CMF) and treats interference as additive white Gaussian noise (AWGN), the LMMSE receiver suppresses interference by replacing the CMF with a new chip pulse filter. Second, in contrast to the Rake receiver which only processes a subset of strongest paths of the desired user, the LMMSE receiver harnesses the energy of all paths of the desired user that fall within its time support, at no additional complexity. The performance of the proposed LMMSE receiver is analyzed and compared with that of the coherent Rake receiver with probability of bit error, <i>Pe</i>, as the figure of merit. The analysis is based on the accurate improved Gaussian approximation (IGA) technique. Closed form conditional <i>Pe</i> expressions for both the LMMSE and Rake receivers are derived. Furthermore, it is shown that if quadriphase random spreading, moderate to large spreading factors, and pulses with small excess bandwidth are used, the widely-used standard Gaussian Approximation (SGA) technique becomes accurate even for low regions of <i>Pe</i>. Under the examined scenarios tailored towards current narrowband system settings, the LMMSE receiver achieves 60% gain in capacity (1. 8 dB in output SNR) over the selective Rake receiver. A third of the gain is due to interference suppression capability of the receiver while the rest is credited to its ability to collect the energy of the desired user diversified to many paths. Future wideband systems will yield an ever larger gain. Adaptive implementations of the LMMSE receiver are proposed to rid the receiver from dependence on the knowledge of multipath parameters. The adaptive receiver is based on a fractionally-spaced equalizer (FSE) whose taps are updated by an adaptive algorithm. Training-based, pilot-channel-aided (PCA), and blind algorithms are developed to make the receiver applicable to both forward and reverse links, with or without the presence of pilot signals. The blind algorithms are modified versions of the constant modulus algorithm (CMA) which has not been previously studied for long-code CDMA systems. Extensive simulation results are presented to illustrate the convergence behavior of the proposed algorithms and quantify their performance loss under various levels of MAI. Computational complexities of the algorithms are also discussed. These three criteria (performance loss, convergence rate, and computational complexity) determine the proper choice of an adaptive algorithm with respect to the requirements of the specific application in mind.

Three main analog circuit building blocks that are important for a mixed-signal system are investigated in this work. New building blocks with emphasis on power efficiency and compatibility with deep-submicron technology are proposed and experimental results from prototype integrated circuits are presented. Firstly, a 1.1GHz, 5th order, active-LC, Butterworth wideband equalizer that controls inter-symbol interference and provides anti-alias filtering for the subsequent analog to digital converter is presented. The equalizer design is based on a new series LC resonator biquad whose power efficiency is analytically shown to be better than a conventional Gm-C biquad. A prototype equalizer is fabricated in a standard 0.18μm CMOS technology. It is experimentally verified to achieve an equalization gain programmable over a 0-23dB range, 47dB SNR and -48dB IM3 while consuming 72mW of power. This corresponds to more than 7 times improvement in power efficiency over conventional Gm-C equalizers. Secondly, a load capacitance aware compensation for 3-stage amplifiers is presented. A class-AB 16W headphone driver designed using this scheme in 130nm technology is experimentally shown to handle 1pF to 22nF capacitive load while consuming as low as 1.2mW of quiescent power. It can deliver a maximum RMS power of 20mW to the load with -84.8dB THD and 92dB peak SNR, and it occupies a small area of 0.1mm2. The power consumption is reduced by about 10 times compared to drivers that can support such a wide range of capacitive loads. Thirdly, a novel approach to design of ADC in deep-submicron technology is described. The presented technique enables the usage of time-to-digital converter (TDC) in a delta-sigma modulator in a manner that takes advantage of its high timing precision while noise-shaping the error due to its limited time resolution. A prototype ADC designed based on this deep-submicron technology friendly architecture was fabricated in a 65nm digital CMOS technology. The ADC is experimentally shown to achieve 68dB dynamic range in 20MHz signal bandwidth while consuming 10.5mW of power. It is projected to reduce power and improve speed with technology scaling.

Η χρήση επαναληπτικών τεχνικών προσαρμοστικής ισοστάθμισης διαύλου αποτελεί μια σχετικά πρόσφατη και πολλά υποσχόμενη μέθοδο αντιμετώπισης του φαινομένου της διασυμβολικής παρεμβολής που εισάγεται από το κανάλι λόγω του φαινομένου της πολυδιόδευσης. Ο αλγόριθμος που έχει επικρατήσει στις περισσότερες προσαρμοστικές εφαρμογές είναι ο ελαχίστων μέσων τετραγώνων (LMS). Διακρίνεται για την απλότητά του, έχει όμως φτωχές ιδιότητες σύγκλισης. Η μέθοδος των αναδρομικών ελαχίστων τετραγώνων (RLS) είναι επίσης αρκετά διαδεδομένη και κατέχει υπερέχουσες ιδιότητες σύγκλισης. Ωστόσο παρουσιάζει μεγάλη υπολογιστική πολυπλοκότητα και αυξημένες απαιτήσεις σε μνήμη. Στα πλαίσια της εργασίας αυτής εγίνε μια προσπάθεια ανάλυσης των τεχνικών που βασίζονται στη μέθοδο των συζυγών παραγώγων (Conjugate Gradient), χρησιμοποιούνται σε προβλήματα προσαρμοστικού φιλτραρίσματος και πιο ειδικά στο πρόβλημα της προσαρμοστικής ισοστάθμισης διαύλου. Οι τεχνικές αυτές επεξεργάζονται τα δεδομένα και ανά μπλοκ. Είναι ικανές να παρέχουν ιδιότητες σύγκλισης συγκρίσιμες με αυτές της (RLS) μεθόδου, εισάγοντας υπολογιστική πολυπλοκότητα ενδιάμεσων απαιτήσεων μεταξύ των μεθόδων LMS και RLS χωρίς να παρουσιάζουν προβλήματα αριθμητικής ευστάθειας. / The use of iteration methods for adaptive equalization has received considerable attention during the past several decades. The Least Mean Squares (LMS) method, which has found widespread use owing to its simplicity, has poor convergence properties. The Recursive Least Squares (RLS) method possess superior convergence properties, but it is computationally intensive and has high storage requirements for matrix manipulations. In this MSc thesis the technique of conjugate gradients is applied for the adaptive filtering problem. Conjugate gradient algorithms for adaptive filtering applications suitable for efficient implementation has been developed and has been applied for the design of an adaptive transversal equalizer. Low cost block algorithms using the preconditioned conjugate gradient method are also discussed. The algorithms are capable of providing convergence comparable to RLS schemes at a computational complexity between the LMS and the RLS methods and does not suffer from any known instability problems.

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

A new broadband beamformer composed of nested arrays (NAs), multi-dimensional (MD) filters, and multirate techniques is proposed for both linear and planar arrays. It is shown that this combination results in frequency-invariant response. For a given number of sensors, the advantage of using NAs is that the effective aperture for low temporal frequencies is larger than in the case of using uniform arrays. This leads to high spatial selectivity for low frequencies. For a given aperture size, the proposed beamformer can be implemented with significantly fewer sensors and less computation than uniform arrays with a slight deterioration in performance. Taking advantage of the Noble identity and polyphase structures, the proposed method can be efficiently implemented. Simulation results demonstrate the good performance of the proposed beamformer in terms of frequency-invariant response and computational requirements. The broadband beamformer requires a filter bank with a non-compatible set of sampling rates which is challenging to be designed. To address this issue, a filter bank design approach is presented. The approach is based on formulating the design problem as an optimization problem with a performance index which consists of a term depending on perfect reconstruction (PR) and a term depending on the magnitude specifications of the analysis filters. The design objectives are to achieve almost perfect reconstruction (PR) and have the analysis filters satisfying some prescribed frequency specifications. Several design examples are considered to show the satisfactory performance of the proposed method. A new blind multi-stage space-time equalizer (STE) is proposed which can separate narrowband sources from a mixed signal. Neither the direction of arrival (DOA) nor a training sequence is assumed to be available for the receiver. The beamformer and equalizer are jointly updated to combat both co-channel interference (CCI) and inter-symbol interference (ISI) effectively. Using subarray beamformers, the DOA, possibly time-varying, of the captured signal is estimated and tracked. The estimated DOA is used by the beamformer to provide strong CCI cancellation. In order to alleviate inter-stage error propagation significantly, a mean-square-error sorting algorithm is used which assigns detected sources to different stages according to the reconstruction error at different stages. Further, to speed up the convergence, a simple-yet-efficient DOA estimation algorithm is proposed which can provide good initial DOAs for the multi-stage STE. Simulation results illustrate the good performance of the proposed STE and show that it can effectively deal with changing DOAs and time variant channels. / Graduate / 0544 / imanmoaz@uvic.ca

This diploma thesis deals with design and implementation of complex software system for post-production of popular music. The system was implemented as a plug-in module in C++ language using JUCE application framework. The emphasis was on creating a well arranged and intuitive graphic user interface. The plug-in provides a set of audio effects and processors that can be connected into the desired graph structure. For less experienced users, there is a database of preset configurations usable for a variety of input signals.

This thesis describes a design of a simulation of transmission of digital information via communication system and equalization of communication function. The layout of communication channel with multiway transmission is described in following part. Next part is about hardware modulator which generate modulated signal which is transmitted via communication channel and after is sampled by A/D convertion card to computer, where is equalizated and demodulated in Simulink. In the last part of this thesis, there is proposal of the laboratory task and its sample solution.

This master´s thesis deals with a design and simulation of an audio power phono amplifier using vakuum tubes. The frequency response of suggested solution should comply with the norm specified by RIAA. It also describes a vacuum tube basic description and also its advantages for audio applications and principle of usage a magnetodynamic pick-up. Last part contains a design of microprocessor system and power supply.

This work deals with the design of a 10-zones equalizer with optimized frequency characteristics with a spectrum audio analyzer. In this work the problem of processing audio signals using equalization for filtering the interference frequencies, correction of frequency cover signal boost or suppression of the required zones of the audible spectrum are also analyzed. The influence of subjective perception of sound intensity of the audio signals reproduction and its use in working with equalizer is discussed too. The work describes the principles and usage of the audio-analyzer as a tool for the optimization of the audio equalization setting when ensuring the appropriate listening conditions of music reproduction, spoken word, sounds, etc. It also focuses on the signal source for testing audio-chains, their generation and measurement using the audio analyzer. The equalizer equipment, audio-analyzer generator of reference signals equipment and power supply are designed as a single unit.