Multidimensional Filter Banks and Multiscale Geometric Representations by Minh N
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Brislawn-2006-Polyphase.Pdf
LA-UR-04-2090 Approved for public release; distribution is unlimited. Title: The polyphase-with-advance representation and linear phase lifting factorizations Author(s): Christopher M. Brislawn Brendt Wohlberg Submitted to: to appear in IEEE Transactions on Signal Processing Los Alamos National Laboratory, an affirmative action/equal opportunity employer, is operated by the University of California for the U.S. Department of Energy under contract W-7405-ENG-36. By acceptance of this article, the publisher recognizes that the U.S. Government retains a nonexclusive, royalty-free license to publish or reproduce the published form of this contribution, or to allow others to do so, for U.S. Government purposes. Los Alamos National Laboratory requests that the publisher identify this article as work performed under the auspices of the U.S. Department of Energy. Los Alamos National Laboratory strongly supports academic freedom and a researcher’s right to publish; as an institution, however, the Laboratory does not endorse the viewpoint of a publication or guarantee its technical correctness. Form 836 (8/00) 1 The Polyphase-with-Advance Representation and Linear Phase Lifting Factorizations Christopher M. Brislawn*, Member, IEEE, and Brendt Wohlberg Los Alamos National Laboratory, Los Alamos, NM 87545 USA (505) 665–1165 (Brislawn); (505) 667–6886 (Wohlberg); (505) 665–5220 (FAX) [email protected]; [email protected] Abstract A matrix theory is developed for the noncausal polyphase representation that underlies the theory of lifted filter banks and wavelet transforms. The theory presented here develops an extensive matrix algebra framework for analyzing and implementing linear phase two- channel filter banks via lifting cascade schemes. -
CHAPTER 3 ADC and DAC
CHAPTER 3 ADC and DAC Most of the signals directly encountered in science and engineering are continuous: light intensity that changes with distance; voltage that varies over time; a chemical reaction rate that depends on temperature, etc. Analog-to-Digital Conversion (ADC) and Digital-to-Analog Conversion (DAC) are the processes that allow digital computers to interact with these everyday signals. Digital information is different from its continuous counterpart in two important respects: it is sampled, and it is quantized. Both of these restrict how much information a digital signal can contain. This chapter is about information management: understanding what information you need to retain, and what information you can afford to lose. In turn, this dictates the selection of the sampling frequency, number of bits, and type of analog filtering needed for converting between the analog and digital realms. Quantization First, a bit of trivia. As you know, it is a digital computer, not a digit computer. The information processed is called digital data, not digit data. Why then, is analog-to-digital conversion generally called: digitize and digitization, rather than digitalize and digitalization? The answer is nothing you would expect. When electronics got around to inventing digital techniques, the preferred names had already been snatched up by the medical community nearly a century before. Digitalize and digitalization mean to administer the heart stimulant digitalis. Figure 3-1 shows the electronic waveforms of a typical analog-to-digital conversion. Figure (a) is the analog signal to be digitized. As shown by the labels on the graph, this signal is a voltage that varies over time. -
Efficient Implementations of Discrete Wavelet Transforms Using Fpgas Deepika Sripathi
Florida State University Libraries Electronic Theses, Treatises and Dissertations The Graduate School 2003 Efficient Implementations of Discrete Wavelet Transforms Using Fpgas Deepika Sripathi Follow this and additional works at the FSU Digital Library. For more information, please contact [email protected] THE FLORIDA STATE UNIVERSITY COLLEGE OF ENGINEERING EFFICIENT IMPLEMENTATIONS OF DISCRETE WAVELET TRANSFORMS USING FPGAs By DEEPIKA SRIPATHI A Thesis submitted to the Department of Electrical and Computer Engineering in partial fulfillment of the requirements for the degree of Master of Science Degree Awarded: Fall Semester, 2003 The members of the committee approve the thesis of Deepika Sripathi defended on November 18th, 2003. Simon Y. Foo Professor Directing Thesis Uwe Meyer-Baese Committee Member Anke Meyer-Baese Committee Member Approved: Reginald J. Perry, Chair, Department of Electrical and Computer Engineering The office of Graduate Studies has verified and approved the above named committee members ii ACKNOWLEDGEMENTS I would like to express my gratitude to my major professor, Dr. Simon Foo for his guidance, advice and constant support throughout my thesis work. I would like to thank him for being my advisor here at Florida State University. I would like to thank Dr. Uwe Meyer-Baese for his guidance and valuable suggestions. I also wish to thank Dr. Anke Meyer-Baese for her advice and support. I would like to thank my parents for their constant encouragement. I would like to thank my husband for his cooperation and support. I wish to thank the administrative staff of the Electrical and Computer Engineering Department for their kind support. Finally, I would like to thank Dr. -
Post-Sampling Aliasing Control for Natural Images
POST-SAMPLING ALIASING CONTROL FOR NATURAL IMAGES Dinei A. F. Florêncio and Ronald W. Schafer Digital Signal Processing Laboratory School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta, Georgia 30332 f1orenceedsp . gatech . edu rws@eedsp . gatech . edu ABSTRACT Shannon sampling theorem, and can be useful in developing Sampling and recollstruction are usuaiiy analyzed under better pre- and post-filters based on non-linear techniques. the framework of linear signal processing. Powerful tools Critical Morphological Sampling is similar to traditional like the Fourier transform and optimum linear filter design techniques in the sense that it also requires pre-ifitering be- techniques, allow for a very precise analysis of the process. fore subsampling. In some applications, this pre-ifitering In particular, an optimum linear filter of any length can be maybe undesirable, or even impossible, in which case the derived under most situations. Many of these tools are not signal is simply subsampled, without any pre-filtering, or available for non—linear systems, and it is usually difficult to a very simple filter is used. An example of increasing im find an optimum non4inear system under any criteria. In portance is video processing, where the high data rate and this paper we analyze the possibility of using non-linear fil memory restrictions often limit the filtering to very short tering in the interpolation of subsampled images. We show windows. that a very simple (5x5) non-linear reconstruction filter out- In this paper we show how it is possible to reduce performs (for the images analyzed) linear filters of up to the effects of aliasing by using non-linear reconstruction 256x256, including optimum (separable) Wiener filters of techniques. -
Euclidean Algorithm for Laurent Polynomial Matrix Extension
Euclidean Algorithm for Laurent Polynomial Matrix Extension |A note on dual-chain approach to construction of wavelet filters Jianzhong Wang May 11, 2015 Abstract In this paper, we develop a novel and effective Euclidean algorithm for Laurent polynomial matrix extension (LPME), which is the key of the construction of perfect reconstruction filter banks (PRFBs). The algorithm simplifies the dual-chain approach to the construction of PRFBs in the paper [5]. The algorithm in this paper can also be used in the applications where LPME plays a role. Key words. Laurent polynomial matrix extension, perfect reconstruction filter banks, multi-band filter banks, M-dilation wavelets, B´ezoutidentities, Euclidean division algo- rithm. 2010 AMS subject classifications. 42C40, 94A12, 42C15, 65T60, 15A54 1 Introduction In recent years, the theory of filter banks has been developed rapidly since they are widely used in many scientific areas such as signal and image processing, data mining, data feature extraction, and compression sensing. In the theory, the perfect reconstruction filter banks (PRFBs) play an important role [7, 8, 17, 18, 19, 20]. A PRFB consists of two sub-filter banks: an analysis filter bank, which is used to decompose signals into different bands, and a synthetic filter bank, which is used to compose a signal from its different band components. Either an analysis filter bank or a synthetic one consists of several band-pass filters. Figure 1 illustrates the role of an M-band PRFB, in which, the filter set fH0;H1; ··· ;HM−1g forms an analysis filter bank and the filter set fB0;B1; ··· ;BM−1g forms a synthetic one. -
Arxiv:1902.09040V1 [Cs.IT] 24 Feb 2019 Acting on a Space of Vector-Valued Discrete-Time Signals, E.G., `2 Z, C2
FACTORING PERFECT RECONSTRUCTION FILTER BANKS INTO CAUSAL LIFTING MATRICES: A DIOPHANTINE APPROACH CHRISTOPHER M. BRISLAWN Abstract. The theory of linear Diophantine equations in two unknowns over polynomial rings is used to construct causal lifting factorizations for causal two- channel FIR perfect reconstruction multirate filter banks and wavelet trans- forms. The Diophantine approach generates causal lifting factorizations sat- isfying certain polynomial degree-reducing inequalities, enabling a new lifting factorization strategy called the Causal Complementation Algorithm. This provides an alternative to the noncausal lifting scheme based on the Ex- tended Euclidean Algorithm for Laurent polynomials that was developed by Daubechies and Sweldens. The new approach, which can be regarded as Gauss- ian elimination in polynomial matrices, utilizes a generalization of polynomial division that ensures existence and uniqueness of quotients whose remainders satisfy user-specified divisibility constraints. The Causal Complementation Al- gorithm is shown to be more general than the Extended Euclidean Algorithm approach by generating causal lifting factorizations not obtainable using the polynomial Euclidean Algorithm. 1. Introduction Figure1 depicts the Z-transform representation of a two-channel multirate digital def i filter bank with input X(z) = i x(i)z− [1,2,3,4,5]. It is a perfect reconstruction (PR) filter bank if the transfer function X(z)=X(z) is a monomial (i.e., a constant P multiple of a delay) in the absence of additional processing or distortion. For suit- ably chosen polyphase transfer matrices bH(z) and G(z) the system in Figure1 is mathematically equivalent to the polyphase-with-delay (PWD) filter bank represen- tation in Figure2[2,4]. -
Compensation of Reconstruction Filter Effect in Digital Signal Processing System
Journal of Signal Processing, Vol.18, No.3, pp.121-134, May 2014 PAPER Compensation of Reconstruction Filter Effect in Digital Signal Processing System Sukanya Praesomboon1, Kobchai Dejhan1 and Surapun Yimman2 1Faculty of Engineering, King Mongkut’s Institute of Technology Ladkrabang, Bangkok 10520, Thailand 2Faculty of Applied Science, King Mongkut’s University of Technology North Bangkok, Bangkok 10800, Thailand E-mail: [email protected], [email protected], [email protected] Abstract This paper proposes the reduction effect of reconstruction filter for digital signal processing system. It is well known that the reconstruction filter limits the bandwidth of output signal processing and the output signal is staircase form or has high frequency components. For effect reduction we use the nonrecursive system which corresponds with the inverse system function of the reconstruction filter. The initial step starts with the analog system function of the reconstruction filter and is converted to a discrete-time system using the approximation of derivative technique and inverse system. This inverse discrete-time system function is a nonrecursive system to compensate the reduction effect of the reconstruction filter implemented by using the software. For hardware experiments, we use TMS320C31 digital signal processor, D/A, PIC-32 microcontroller, PWM analog output. The experimental results of the proposed principle show that it can be used to compensate the bandwidth and reduce the total harmonic distortion (THD) of high frequency signal. In addition, this experiment shows that the nonrecursive system for compensation uses only first-order system and it does not utilizes any more processor. Furthermore, it is able to increase the performance. -
Oversampled Perfect Reconstruction Filter Bank Transceivers
Oversampled Perfect Reconstruction Filter Bank Transceivers Siavash Rahimi Department of Electrical & Computer Engineering McGill University Montreal, Canada March 2014 A thesis submitted to McGill University in partial fulfillment of the requirements for the degree of Doctor of Philosophy. c 2014 Siavash Rahimi 2014/03/05 i Contents 1 Introduction 1 1.1 Multicarrier Modulation ............................ 1 1.2 Literature Review ................................ 3 1.2.1 Oversampled perfect reconstruction filter bank ............ 5 1.2.2 Synchronization and equalization ................... 6 1.2.3 Extension to multi-user applications .................. 8 1.3 Thesis Objectives and Contributions ..................... 9 1.4 Thesis Organization and Notations ...................... 12 2 Background on Filter Bank Multicarrier Modulation 14 2.1 OFDM ...................................... 15 2.1.1 The Cyclic Prefix ............................ 17 2.1.2 OFDM Challenges ........................... 19 2.2 Multirate Filter Banks ............................. 22 2.2.1 Basic Multirate Operations ...................... 24 2.2.2 Transceiver Transfer Relations and Reconstruction Conditions . 29 2.2.3 Modulated Filter Banks ........................ 32 2.2.4 Oversampled Filter Banks ....................... 37 2.3 A Basic Survey of Different FBMC Types .................. 39 2.3.1 Cosine Modulated Multitone ...................... 39 2.3.2 OFDM/OQAM ............................. 41 2.3.3 FMT ................................... 45 2.3.4 DFT modulated OPRFB ........................ 46 Contents ii 3 OPRFB Transceivers 47 3.1 Background and Problem Formulation .................... 47 3.2 A Factorization of Polyphase Matrix P(z) .................. 51 3.2.1 Preliminary Factorization of P(z)................... 51 3.2.2 Structure of U(z) ........................... 52 3.2.3 Expressing U(z) in terms of Paraunitary Building Blocks ...... 54 3.3 Choice of a Parameterization for Paraunitary Matrix B(z) ........ -
M-Ary Orthogonal Modulation Using Wavelet
M-ARY ORTHOGONAL MODULATION USING WAVELET BASIS FUNCTIONS A Thesis Presented to The Faculty of the School of Electrical Engineering and Computer Science Fritz J. and Dolores H. Russ College of Engineering and Tech~~olog~. Ohio University In Partial Fulfillment of the Requirements for the Degree Master of Science by Xiaoyun Pan No\~ember,3000 THIS THESIS ENTITLED "M-ARY 0:RTHOGONAL MODULATION USING WAVELET BASIS FUNCTIONS" by Xiaoyun Pan has been approved for the School of Electrical Engineering and Computer Science and the Russ College of Engineering and Technology Jerrel R. Mitchell, Dean :/ ,I Fritz J. and Dolores H. Russ College of Engineering and Technology Acknowledgement I would like to express my sincere gratitude to my advisor, Dr. Jeffrey C. Dill, for his instruction and guidance during the development of this thesis. His knowledge, patience and insightful direction and continuous encouragement greatly contributed to the completion of this research. I also wish to thank all the thesis committee members, Dr. David Matolali, Dr. Joseph Essman and Dr. Thomas Hogan for their interest in this thesis, their suggestions and instructions. 'Their willingness to be part of the review process is greatly appreciated. I would like to take this opportunity to express my deepest appreciation to my parents and my husband, Ming, for the love and encouragement they have given me o\ er the years. They are always the indispensable support in my emotion and spirit. Special thanks are given to Jim, my good friend for helping nle check the grammar and spelling of this thesis. His continuous friendship and encouragement remain in my mc!noiy along ith thc two year 2nd eight nlonth's stiidy life in Athcns. -
Generic Invertibility of Multidimensional FIR Filter Banks and MIMO Systems Ka L
IEEE TRANSACTIONS ON SIGNAL PROCESSING 1 Generic Invertibility of Multidimensional FIR Filter Banks and MIMO Systems Ka L. Law, Robert M. Fossum, Member, IEEE, and Minh N. Do, Senior Member, IEEE, Abstract—We study the invertibility of M-variate Laurent original signal. Using the polyphase representation in the z- polynomial N × P matrices. Such matrices represent multi- domain [1], [2], we can represent the analysis part as an dimensional systems in various settings such as filter banks, N × P matrix H(z) (shown in Fig.1(b)) with entries in a multiple-input multiple-output systems, and multirate systems. −1 −1 Laurent polynomial ring C[z1,z2, ..., zM ,z1 , ..., zM ]. In Given an N × P Laurent polynomial matrix H(z1, ..., zM ) of degree at most k, we want to find a P × N Laurent polynomial this case M is the dimension of signals, N is the number left inverse matrix G(z) of H(z) such that G(z)H(z) = I. of channels in the filter bank, and P is the sampling factor We provide computable conditions to test the invertibility and at each channel. An application of this setting may arise in propose algorithms to find a particular inverse. multichannel acquisition. In such an application we collect The main result of this paper is to prove that H(z) is X z generically invertible when N −P ≥ M; whereas when N −P < data about an unknown multidimensional signal ( ) as M, then H(z) is generically noninvertible. As a result, we output of the analysis part in Fig. 1(a). -
Multidimensional Filter Banks and Multiscale Geometric Representations Full Text Available At
Full text available at: http://dx.doi.org/10.1561/2000000012 Multidimensional Filter Banks and Multiscale Geometric Representations Full text available at: http://dx.doi.org/10.1561/2000000012 Multidimensional Filter Banks and Multiscale Geometric Representations Minh N. Do University of Illinois at Urbana-Champaign Urbana, IL 61801 USA [email protected] Yue M. Lu Harvard University Cambridge, MA 02138 USA [email protected] Boston { Delft Full text available at: http://dx.doi.org/10.1561/2000000012 Foundations and Trends R in Signal Processing Published, sold and distributed by: now Publishers Inc. PO Box 1024 Hanover, MA 02339 USA Tel. +1-781-985-4510 www.nowpublishers.com [email protected] Outside North America: now Publishers Inc. PO Box 179 2600 AD Delft The Netherlands Tel. +31-6-51115274 The preferred citation for this publication is M. N. Do and Y. M. Lu, Multidimen- sional Filter Banks and Multiscale Geometric Representations, Foundations and Trends R in Signal Processing, vol 5, no 3, pp 157{264, 2011 ISBN: 978-1-60198-584-2 c 2012 M. N. Do and Y. M. Lu All rights reserved. No part of this publication may be reproduced, stored in a retrieval system, or transmitted in any form or by any means, mechanical, photocopying, recording or otherwise, without prior written permission of the publishers. Photocopying. In the USA: This journal is registered at the Copyright Clearance Cen- ter, Inc., 222 Rosewood Drive, Danvers, MA 01923. Authorization to photocopy items for internal or personal use, or the internal or personal use of specific clients, is granted by now Publishers Inc for users registered with the Copyright Clearance Center (CCC). -
Reconstruction of Nonuniformly Sampled Bandlimited Signals
RECONSTRUCTION OF NONUNIFORMLY SAMPLED BANDLIMITED SIGNALS USING TIME-VARYING DISCRETE-TIME FIR FILTERS Håkan Johansson and Per Löwenborg Division of Electronics Systems Department of Electrical Engineering SE-581 83 Linköping University, Sweden Abstract – This paper deals with reconstruction of non- (a) xa(t) (b) xa(t) uniformly sampled bandlimited continuous-time signals using time-varying discrete-time FIR filters. The points of departures are that the signal is slightly oversampled as to the average sampling frequency and that the sampling t t t t t t t instances are known. Under these assumptions, a representa- 0 T4T2T3T 5T t 0 1 2 3 4 5 tion of the reconstructed sequence is derived that utilizes a (c) xa(t) time-frequency function. This representation enables a proper utilization of the oversampling and reduces the reconstruction problem to a design problem that resembles an ordinary filter design problem. Furthermore, for an d0 2T+d0 4T+d0 t important special case, corresponding to a certain type of d1 2T+d1 4T+d1 periodic nonuniform sampling, it is shown that the recon- igure 1. (a) Uniform sampling. (b) Nonuniform sampling. struction problem can be posed as a filter-bank design prob- c) Periodic nonuniform sampling. lem, thus with requirements on a distortion transfer function and a number of aliasing transfer functions. 1. INTRODUCTION cut-off frequency. Although different weights (windows) Nonuniform sampling occurs in many practical applications can improve the situation, it is recognized that other design either intentionally or unintentionally [1]. An example of the methods are usually preferred [8]. When designing filters, latter is found in time-interleaved analog-to-digital convert- one does normally not attempt to approximate a desired ers (ADCs) where static time-skew errors between the dif- function over the whole frequency range ωT ≤ π , ωT ferent subconverters give rise to a class of periodic being the frequency variable.