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    Partial-Update Adaptive Signal Processing: Design Analysis and Implementation
    Partial-Update Adaptive Signal Processing: Design Analysis and Implementation
    Partial-Update Adaptive Signal Processing: Design Analysis and Implementation
    Ebook451 pages2 hours

    Partial-Update Adaptive Signal Processing: Design Analysis and Implementation

    By Kutluyil Doğançay

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    About this ebook

    Partial-update adaptive signal processing algorithms not only permit significant complexity reduction in adaptive filter implementations, but can also improve adaptive filter performance in telecommunications applications. This book gives state-of-the-art methods for the design and development of partial-update adaptive signal processing algorithms for use in systems development.Partial-Update Adaptive Signal Processing provides a comprehensive coverage of key partial updating schemes, giving detailed information on the theory and applications of acoustic and network echo cancellation, channel equalization and multiuser detection. It also examines convergence and stability issues for partial update algorithms, providing detailed complexity analysis and a unifying treatment of partial-update techniques.Features:• Advanced analysis and design tools• Application examples illustrating the use of partial-update adaptive signal processing• MATLAB codes for developed algorithms This unique reference will be of interest to signal processing and communications engineers, researchers, R&D engineers and graduate students."This is a very systematic and methodical treatment of an adaptive signal processing topic, of particular significance in power limited applications such as in wireless communication systems and smart ad hoc sensor networks. I am very happy to have this book on my shelf, not to gather dust, but to be consulted and used in my own research and teaching activities" – Professor A. G. Constantinides, Imperial College, LondonAbout the author:Kutluyil Dogançay is an associate professor of Electrical Engineering at the University of South Australia. His research interests span statistical and adaptive signal processing and he serves as a consultant to defence and private industry. He was the Signal Processing and Communications Program Chair of IDC Conference 2007, and is currently chair of the IEEE South Australia Communications and Signal Processing Chapter.
    • Advanced analysis and design tools
    • Algorithm summaries in tabular format
    • Case studies illustrate the application of partial update adaptive signal processing
    LanguageEnglish
    PublisherAcademic Press
    Release dateSep 17, 2008
    ISBN9780080921150
    Partial-Update Adaptive Signal Processing: Design Analysis and Implementation
    Author

    Kutluyil Doğançay

    Kutluyil Dogançay received the BS degree with honors in electrical and electronic engineering from Bogaziçi University, Istanbul, Turkey, in 1989, the MSc degree in communications and signal processing from Imperial College, The University of London, UK, in 1992, and the PhD degree in telecommunications engineering from The Australian National University, Canberra, ACT, Australia, in 1996. Since November 1999, he has been with the School of Engineering, University of South Australia, where he is a professor and discipline leader of electrical and mechatronic engineering. His research interests span statistical and adaptive signal processing with applications in defence and communication systems. Dr Dogançay received the Best Researcher Award of School of Engineering, University of South Australia, in 2015, and Tall Poppy Science Award of the Australian Institute of Political Science in 2005. He was the Tutorials Chair of the IEEE Statistical Signal Processing Workshop (SSP 2014), and the Signal Processing and Communications Program Chair of the 2007 Information, Decision and Control Conference. He serves on the Editorial Board of Signal Processing and the EURASIP Journal on Advances in Signal Processing. From 2009-2015 he was an elected member of the Signal Processing Theory and Methods (SPTM) Technical Committee of the IEEE Signal Processing Society. He is currently an associate member of the Sensor Array and Multichannel (SAM) Technical Committee and a member of the IEEE ComSoc Signal Processing for Communications and Electronics Technical Committee. Dr Dogançay is the EURASIP liaison for Australia.

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      Partial-Update Adaptive Signal Processing - Kutluyil Doğançay

      Index

      Chapter 1

      Introduction

      1.1 Adaptive signal processing

      In practice most systems are inherently time-varying and/or nonlinear. The signals associated with these systems often have time-varying characteristics. Adaptive signal processing is a branch of statistical signal processing that deals with the challenging problem of estimation and tracking of time-varying systems. By virtue of its applicability to time-varying and/or nonlinear systems, adaptive signal processing finds application in a broad range of practical fields such as telecommunications, radar and sonar signal processing, biomedical engineering and entertainment systems. In order to make the estimation and tracking task tractable, the unknown system is usually modelled as a time-varying linear system or in some cases as a finitely parameterized nonlinear system such as the Volterra filter. This simplified system modelling is guided by prior knowledge of system characteristics. An important objective of adaptive signal processing is to learn the unknown and possibly time-varying signal statistics in conjunction with system estimation.

      This chapter presents a brief overview of fundamental principles of adaptive signal processing and motivates partial-update adaptive signal processing as a low-complexity implementation option in the face of resource constraints. Adaptive system identification is shown to be the central theme of adaptive signal processing. In the context of adaptive system identification partial coefficient updating is proposed as an attractive approach to complexity reduction. The chapter sets the scene for the remainder of the book by presenting simple adaptive system identification examples that illustrate the potential benefits of partial-update adaptive signal processing in addition to allowing compliance with the existing resource constraints.

      1.2 Examples of adaptive filtering

      A fundamental building block for an adaptive signal processing system is the adaptive filter. The objective of an adaptive filter is to learn an unknown system from observations of the system input and/or output signals utilizing any a priori knowledge of the system and signal characteristics. The task of learning an unknown system is fundamental to many signal processing problems and comes in many disguises in applications of adaptive filters. In this section we take a brief look at two broad examples of adaptive filters. These examples serve the purpose of elucidating the basic components of adaptive filters.

      1.2.1 Adaptive system identification

      Most adaptive filtering problems are either (1) a special case of adaptive system identification or (2) utilize adaptive system identification as a means of solving another signal processing problem. In this sense, adaptive system identification provides the basis for a wide range of adaptive signal processing applications. It is, therefore, essential that we have a good understanding of the underlying principles and assumptions relating to adaptive system identification.

      As depicted in Figure 1.1, in adaptive system identification, the objective is to estimate an unknown system from its input and output observations given by x(k, respectively. Throughout this book we restrict our attention to discrete-time signals and systems, so the independent time index k is an integer. A model for the adaptive filter is chosen based on prior knowledge of the unknown system characteristics, as well as complexity considerations. In its simplest and most preferred form, the adaptive filter is a finite impulse response (FIR) filter of length N with adjustable impulse response coefficients (adaptive filter coefficients):

      (1.1)

      Here T denotes the transpose operator. Equation are assumed to be real-valued unless otherwise specified. It is often straightforward to extend the analysis to adaptive filters with complex coefficients.

      Figure 1.1 Adaptive system identification.

      is called the error signal e(k) :

      (1.2)

      At each iteration k the adaptive filter updates its coefficients in order to minimize an appropriate norm of the error signal e(kgives an estimate of the unknown system parameters. If the unknown system is time-varying, i.e. its parameters change with time, the adaptive filter can track these changes by updating its coefficients in accordance with the error signal. It can take several iterations for the adaptation process to converge (i.e. to learn unknown system parameters). The time taken by the adaptation process to converge provides an indication of the convergence rate.

      There are two main tasks performed by the adaptive filter; viz. adaptation process and filtering process. In Figure 1.1 these processes are identified by the adaptation process and adaptive filter blocks. For a linear adaptive filter as given by (1.1), the filtering process involves convolution. If the number of adaptive filter coefficients is large, the convolution operation may prove to be computationally expensive. Reduced complexity convolution techniques based on fast Fourier transform (FFT), such as overlap-add and overlap-save, may be used to ease computational demand. The adaptation process can also become computationally expensive for long adaptive filters due to the arithmetic operations required to update the adaptive filter coefficients. The computational complexity of the adaptation process depends on the adaptation algorithm employed.

      Prediction of random signals and noise cancellation are two special cases of adaptive system identification. Figure 1.2(a) shows a one-step predictor which estimates the present value of the random signal x(k. If x(k) is a stable autoregressive (AR) process of order N:

      (1.3)

      where v(kin . After convergence the prediction error e(k) is equal to v(k), which implies whitening of the coloured noise signal x(kat the input of the adaptive filter. Swapping x(kin Figure 1.1 and referring to x(kas the interfering noise changes the system identification problem to a noise cancellation problem with e(kmay undergo before interfering with the signal of interest x(k). The sum of x(kis the primary signal. The unknown system is identified by an adaptive filter. Subtracting the adaptive filter output from the reference signal gives the error signal. Minimization of the error norm implies minimization of the difference between the adaptive filter output and the filtered reference signal. If the adaptive filter provides a perfect estimate of the unknown system, then the error signal becomes identical to the signal of interest. This provides perfect noise removal.

      Figure 1.2 Special cases of system identification. (a) One-step prediction of a random signal by an adaptive filter which identifies the autoregressive model of the random signal with y ( k ) giving the prediction output and e ( k ) the prediction error; (b) Adaptive noise cancellation where the adaptive filter estimates the unknown system filtering the input noise signal n ( k ) with x ( k ) denoting the signal of interest and e ( k ) the ‘cleaned’ signal.

      1.2.2 Adaptive inverse system identification

      Figure 1.3 illustrates the adaptive inverse system identification problem. Comparison of Figures 1.2 and 1.3 reveals that adaptive inverse system identification requires an adaptive filter to be connected to the input and noisy output of the unknown system in the reverse direction. The use of the D for the desired adaptive filter response ensures that the adaptive filter will be able to approximate the inverse system for non-minimum-phase or maximum-phase linear systems. For such systems the stable inverse system is non-causal and has infinite impulse response (IIR). A (causal) FIR adaptive filter can approximate the stable inverse only if it is sufficiently long (i.e. N is sufficiently large) and D ).

      Figure 1.3 Adaptive inverse system identification.

      as a measure of accuracy for inverse system identification. The adaptive filter coefficients are adjusted iteratively in order to minimize the error norm in a statistical sense. At the end of the minimization process the adaptive filter converges to an estimate of the inverse of the unknown system. Due to the presence of noise at the unknown system output, the inverse system estimate is not identical to the zero-forcing solution that ignores the output

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