Algorithms for Communications Systems and their Applications
Algorithms for Communications Systems and their Applications
Nevio Benvenuto University of Padova, Italy
Giovanni Cherubini IBM Zurich Research Laboratory, Switzerland
Copyright c 2002
John Wiley & Sons Ltd, The Atrium, Southern Gate, Chichester, West Sussex PO19 8SQ, England Telephone (C44) 1243 779777
Email (for orders and customer service enquiries): csbooks@wiley.co.uk Visit our Home Page on www.wileyeurope.com or www.wiley.com Reprinted with corrections March 2003 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, electronic, mechanical, photocopying, recording, scanning or otherwise, except under the terms of the Copyright, Designs and Patents Act 1988 or under the terms of a licence issued by the Copyright Licensing Agency Ltd, 90 Tottenham Court Road, London W1T 4LP, UK, without the permission in writing of the Publisher. Requests to the Publisher should be addressed to the Permissions Department, John Wiley & Sons Ltd, The Atrium, Southern Gate, Chichester, West Sussex PO19 8SQ, England, or emailed to permreq@wiley.co.uk, or faxed to (C44) 1243 770571. Neither the author(s) nor John Wiley & Sons, Ltd accept any responsibility or liability for loss or damage occasioned to any person or property through using the material, instructions methods or ideas contained herein, or acting or refraining from acting as a result of such use. The author(s) and Publisher expressly disclaim all implied warranties, including merchantability of ﬁtness for any particular purpose. Designations used by companies to distinguish their products are often claimed as trademarks. In all instances where John Wiley & Sons is aware of a claim, the product names appear in initial capital or capital letters. Readers, however, should contact the appropriate companies for more complete information regarding trademarks and registration.
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To Adriana, and to Antonio, Claudia, and Mariuccia
Contents
Preface Acknowledgements 1 Elements of signal theory 1.1 Signal space : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Properties of a linear space : : : : : : : : : : : : : : : : Inner product : : : : : : : : : : : : : : : : : : : : : : : 1.2 Discrete signal representation : : : : : : : : : : : : : : : : : : : : The principle of orthogonality : : : : : : : : : : : : : : Signal representation : : : : : : : : : : : : : : : : : : : Gram–Schmidt orthonormalization procedure : : : : : : 1.3 Continuoustime linear systems : : : : : : : : : : : : : : : : : : : 1.4 Discretetime linear systems : : : : : : : : : : : : : : : : : : : : : Discrete Fourier transform (DFT) : : : : : : : : : : : : The DFT operator : : : : : : : : : : : : : : : : : : : : : Circular and linear convolution via DFT : : : : : : : : : Convolution by the overlapsave method : : : : : : : : : IIR and FIR ﬁlters : : : : : : : : : : : : : : : : : : : : 1.5 Signal bandwidth : : : : : : : : : : : : : : : : : : : : : : : : : : The sampling theorem : : : : : : : : : : : : : : : : : : Heaviside conditions for the absence of signal distortion 1.6 Passband signals : : : : : : : : : : : : : : : : : : : : : : : : : : : Complex representation : : : : : : : : : : : : : : : : : : Relation between x and x .bb/ : : : : : : : : : : : : : : : Baseband equivalent of a transformation : : : : : : : : : Envelope and instantaneous phase and frequency : : : : 1.7 Secondorder analysis of random processes : : : : : : : : : : : : : 1.7.1 Correlation : : : : : : : : : : : : : : : : : : : : : : : : : : Properties of the autocorrelation function : : : : : : : : 1.7.2 Power spectral density : : : : : : : : : : : : : : : : : : : : Spectral lines in the PSD : : : : : : : : : : : : : : : : : Crosspower spectral density : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
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Properties of the PSD : : : : : : : : : : : : : : : PSD of processes through linear transformations : PSD of processes through ﬁltering : : : : : : : : 1.7.3 PSD of discretetime random processes : : : : : : : Spectral lines in the PSD : : : : : : : : : : : : : PSD of processes through ﬁltering : : : : : : : : Minimumphase spectral factorization : : : : : : 1.7.4 PSD of passband processes : : : : : : : : : : : : : PSD of the quadrature components of a random process : : : : : : : : : Cyclostationary processes : : : : : : : : : : : : : The autocorrelation matrix : : : : : : : : : : : : : : : : : : Deﬁnition : : : : : : : : : : : : : : : : : : : : : Properties : : : : : : : : : : : : : : : : : : : : : Eigenvalues : : : : : : : : : : : : : : : : : : : : Other properties : : : : : : : : : : : : : : : : : : Eigenvalue analysis for Hermitian matrices : : : Examples of random processes : : : : : : : : : : : : : : : Matched ﬁlter : : : : : : : : : : : : : : : : : : : : : : : : Matched ﬁlter in the presence of white noise : : Ergodic random processes : : : : : : : : : : : : : : : : : : 1.11.1 Mean value estimators : : : : : : : : : : : : : : : : Rectangular window : : : : : : : : : : : : : : : Exponential ﬁlter : : : : : : : : : : : : : : : : : General window : : : : : : : : : : : : : : : : : : 1.11.2 Correlation estimators : : : : : : : : : : : : : : : : Unbiased estimate : : : : : : : : : : : : : : : : : Biased estimate : : : : : : : : : : : : : : : : : : 1.11.3 Power spectral density estimators : : : : : : : : : : Periodogram or instantaneous spectrum : : : : : Welch periodogram : : : : : : : : : : : : : : : : Blackman and Tukey correlogram : : : : : : : : Windowing and window closing : : : : : : : : : Parametric models of random processes : : : : : : : : : : : ARMA. p; q/ model : : : : : : : : : : : : : : : MA(q) model : : : : : : : : : : : : : : : : : : : AR(N ) model : : : : : : : : : : : : : : : : : : : Spectral factorization of an AR(N ) model : : : : Whitening ﬁlter : : : : : : : : : : : : : : : : : : Relation between ARMA, MA and AR models : 1.12.1 Autocorrelation of AR processes : : : : : : : : : : 1.12.2 Spectral estimation of an AR.N / process : : : : : : Some useful relations : : : : : : : : : : : : : : : AR model of sinusoidal processes : : : : : : : : Guide to the bibliography : : : : : : : : : : : : : : : : : :
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Bibliography : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : 1.A Multirate systems : : : : : : : : : : 1.A.1 Fundamentals : : : : : : : : 1.A.2 Decimation : : : : : : : : : 1.A.3 Interpolation : : : : : : : : : 1.A.4 Decimator ﬁlter : : : : : : : 1.A.5 Interpolator ﬁlter : : : : : : : 1.A.6 Rate conversion : : : : : : : 1.A.7 Time interpolation : : : : : : Linear interpolation : : : : Quadratic interpolation : : 1.A.8 The noble identities : : : : : 1.A.9 The polyphase representation Efﬁcient implementations : 1.B Generation of Gaussian noise : : : :
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2 The Wiener ﬁlter and linear prediction 2.1 The Wiener ﬁlter : : : : : : : : : : : : : : : : : : : : : : : : Matrix formulation : : : : : : : : : : : : : : : : : Determination of the optimum ﬁlter coefﬁcients : : The principle of orthogonality : : : : : : : : : : : Expression of the minimum meansquare error : : : Characterization of the cost function surface : : : : The Wiener ﬁlter in the zdomain : : : : : : : : : 2.2 Linear prediction : : : : : : : : : : : : : : : : : : : : : : : : Forward linear predictor : : : : : : : : : : : : : : Optimum predictor coefﬁcients : : : : : : : : : : : Forward “prediction error ﬁlter” : : : : : : : : : : Relation between linear prediction and AR models First and second order solutions : : : : : : : : : : 2.2.1 The Levinson–Durbin algorithm : : : : : : : : : : : Lattice ﬁlters : : : : : : : : : : : : : : : : : : : : 2.2.2 The Delsarte–Genin algorithm : : : : : : : : : : : : 2.3 The least squares (LS) method : : : : : : : : : : : : : : : : Data windowing : : : : : : : : : : : : : : : : : : : Matrix formulation : : : : : : : : : : : : : : : : : Correlation matrix : : : : : : : : : : : : : : : : Determination of the optimum ﬁlter coefﬁcients : : 2.3.1 The principle of orthogonality : : : : : : : : : : : : : Expressions of the minimum cost function : : : : : The normal equation using the T matrix : : : : : : Geometric interpretation: the projection operator : : 2.3.2 Solutions to the LS problem : : : : : : : : : : : : : Singular value decomposition of T : : : : : : : : : Minimum norm solution : : : : : : : : : : : : : :
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Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : 2.A The estimation problem : : : : : : : : : : : : : : : : : The estimation problem for random variables MMSE estimation : : : : : : : : : : : : : : : Extension to multiple observations : : : : : : MMSE linear estimation : : : : : : : : : : : MMSE linear estimation for random vectors : 3
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Adaptive transversal ﬁlters 3.1 Adaptive transversal ﬁlter: MSE criterion : : : : : : : : : : : : : : 3.1.1 Steepest descent or gradient algorithm : : : : : : : : : : : Stability of the steepest descent algorithm : : : : : : : : Conditions for convergence : : : : : : : : : : : : : : : : Choice of the adaptation gain for fastest convergence : : Transient behavior of the MSE : : : : : : : : : : : : : : 3.1.2 The least meansquare (LMS) algorithm : : : : : : : : : : Implementation : : : : : : : : : : : : : : : : : : : : : : Computational complexity : : : : : : : : : : : : : : : : Canonical model : : : : : : : : : : : : : : : : : : : : : Conditions for convergence : : : : : : : : : : : : : : : : 3.1.3 Convergence analysis of the LMS algorithm : : : : : : : : Convergence of the mean : : : : : : : : : : : : : : : : : Convergence in the meansquare sense (real scalar case) Convergence in the meansquare sense (general case) : : Basic results : : : : : : : : : : : : : : : : : : : : : : : : Observations : : : : : : : : : : : : : : : : : : : : : : : Final remarks : : : : : : : : : : : : : : : : : : : : : : : 3.1.4 Other versions of the LMS algorithm : : : : : : : : : : : : Leaky LMS : : : : : : : : : : : : : : : : : : : : : : : : Sign algorithm : : : : : : : : : : : : : : : : : : : : : : Sigmoidal algorithm : : : : : : : : : : : : : : : : : : : Normalized LMS : : : : : : : : : : : : : : : : : : : : : Variable adaptation gain : : : : : : : : : : : : : : : : : LMS for lattice ﬁlters : : : : : : : : : : : : : : : : : : : 3.1.5 Example of application: the predictor : : : : : : : : : : : : 3.2 The recursive least squares (RLS) algorithm : : : : : : : : : : : : Normal equation : : : : : : : : : : : : : : : : : : : : : Derivation of the RLS algorithm : : : : : : : : : : : : : Initialization of the RLS algorithm : : : : : : : : : : : : Recursive form of E min : : : : : : : : : : : : : : : : : : Convergence of the RLS algorithm : : : : : : : : : : : : Computational complexity of the RLS algorithm : : : : : Example of application: the predictor : : : : : : : : : : 3.3 Fast recursive algorithms : : : : : : : : : : : : : : : : : : : : : : 3.3.1 Comparison of the various algorithms : : : : : : : : : : :
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Block adaptive algorithms in the frequency domain : : : : : : : : : 3.4.1 Block LMS algorithm in the frequency domain: the basic scheme : : : : : : : : : : : : : : : : : : : : : : : : Computational complexity of the block LMS algorithm via FFT : : : : : : : : : : : : : : : 3.4.2 Block LMS algorithm in the frequency domain: the FLMS algorithm : : : : : : : : : : : : : : : : : : : : : : Computational complexity of the FLMS algorithm : : : : : Convergence in the mean of the coefﬁcients for the FLMS algorithm : : : : : : : : : : : : 3.5 LMS algorithm in a transformed domain : : : : : : : : : : : : : : : 3.5.1 Basic scheme : : : : : : : : : : : : : : : : : : : : : : : : : On the speed of convergence : : : : : : : : : : : : : : : : 3.5.2 Normalized FLMS algorithm : : : : : : : : : : : : : : : : : 3.5.3 LMS algorithm in the frequency domain : : : : : : : : : : : 3.5.4 LMS algorithm in the DCT domain : : : : : : : : : : : : : : 3.5.5 General observations : : : : : : : : : : : : : : : : : : : : : 3.6 Examples of application : : : : : : : : : : : : : : : : : : : : : : : 3.6.1 System identiﬁcation : : : : : : : : : : : : : : : : : : : : : Linear case : : : : : : : : : : : : : : : : : : : : : : : : : Finite alphabet case : : : : : : : : : : : : : : : : : : : : : 3.6.2 Adaptive cancellation of interfering signals : : : : : : : : : : General solution : : : : : : : : : : : : : : : : : : : : : : : 3.6.3 Cancellation of a sinusoidal interferer with known frequency 3.6.4 Disturbance cancellation for speech signals : : : : : : : : : : 3.6.5 Echo cancellation in subscriber loops : : : : : : : : : : : : : 3.6.6 Adaptive antenna arrays : : : : : : : : : : : : : : : : : : : : 3.6.7 Cancellation of a periodic interfering signal : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 3.A PN sequences : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Maximallength sequences : : : : : : : : : : : : : : : : : CAZAC sequences : : : : : : : : : : : : : : : : : : : : : Gold sequences : : : : : : : : : : : : : : : : : : : : : : : 3.B Identiﬁcation of a FIR system by PN sequences : : : : : : : : : : : 3.B.1 Correlation method : : : : : : : : : : : : : : : : : : : : : : Signaltoestimation error ratio : : : : : : : : : : : : : : : 3.B.2 Methods in the frequency domain : : : : : : : : : : : : : : : System identiﬁcation in the absence of noise : : : : : : : : System identiﬁcation in the presence of noise : : : : : : : 3.B.3 The LS method : : : : : : : : : : : : : : : : : : : : : : : : Formulation using the data matrix : : : : : : : : : : : : : Computation of the signaltoestimation error ratio : : : : 3.B.4 The LMMSE method : : : : : : : : : : : : : : : : : : : : : 3.B.5 Identiﬁcation of a continuoustime system : : : : : : : : : : 3.4
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Transmission media 4.1 Electrical characterization of a transmission system : : : : : Simpliﬁed scheme of a transmission system : : : Characterization of an active device : : : : : : : Conditions for the absence of signal distortion : : Characterization of a 2port network : : : : : : : Measurement of signal power : : : : : : : : : : 4.2 Noise generated by electrical devices and networks : : : : : Thermal noise : : : : : : : : : : : : : : : : : : : Shot noise : : : : : : : : : : : : : : : : : : : : : Noise in diodes and transistors : : : : : : : : : : Noise temperature of a twoterminal device : : : Noise temperature of a 2port network : : : : : : Equivalentnoise models : : : : : : : : : : : : : Noise ﬁgure of a 2port network : : : : : : : : : Cascade of 2port networks : : : : : : : : : : : : 4.3 Signaltonoise ratio (SNR) : : : : : : : : : : : : : : : : : SNR for a twoterminal device : : : : : : : : : : SNR for a 2port network : : : : : : : : : : : : Relation between noise ﬁgure and SNR : : : : : 4.4 Transmission lines : : : : : : : : : : : : : : : : : : : : : : 4.4.1 Fundamentals of transmission line theory : : : : : : Ideal transmission line : : : : : : : : : : : : : : Nonideal transmission line : : : : : : : : : : : : Frequency response : : : : : : : : : : : : : : : : Conditions for the absence of signal distortion : : Impulse response of a nonideal transmission line Secondary constants of some transmission lines : 4.4.2 Crosstalk : : : : : : : : : : : : : : : : : : : : : : Nearend crosstalk : : : : : : : : : : : : : : : : Farend crosstalk : : : : : : : : : : : : : : : : : 4.5 Optical ﬁbers : : : : : : : : : : : : : : : : : : : : : : : : : Description of a ﬁberoptic transmission system : 4.6 Radio links : : : : : : : : : : : : : : : : : : : : : : : : : : 4.6.1 Frequency ranges for radio transmission : : : : : : Radiation masks : : : : : : : : : : : : : : : : : : 4.6.2 Narrowband radio channel model : : : : : : : : : : Equivalent circuit at the receiver : : : : : : : : : Multipath : : : : : : : : : : : : : : : : : : : : : 4.6.3 Doppler shift : : : : : : : : : : : : : : : : : : : : : 4.6.4 Propagation of wideband signals : : : : : : : : : : Channel parameters in the presence of multipath : Statistical description of fading channels : : : : : 4.6.5 Continuoustime channel model : : : : : : : : : : : Power delay proﬁle : : : : : : : : : : : : : : : :
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Doppler spectrum : : : : : : : : : : : : : : : : : : : Doppler spectrum models : : : : : : : : : : : : : : : Shadowing : : : : : : : : : : : : : : : : : : : : : : Final remarks : : : : : : : : : : : : : : : : : : : : : 4.6.6 Discretetime model for fading channels : : : : : : : : Generation of a process with a preassigned spectrum 4.7 Telephone channel : : : : : : : : : : : : : : : : : : : : : : : : 4.7.1 Characteristics : : : : : : : : : : : : : : : : : : : : : : Linear distortion : : : : : : : : : : : : : : : : : : : Noise sources : : : : : : : : : : : : : : : : : : : : : Nonlinear distortion : : : : : : : : : : : : : : : : : Frequency offset : : : : : : : : : : : : : : : : : : : Phase jitter : : : : : : : : : : : : : : : : : : : : : : Echo : : : : : : : : : : : : : : : : : : : : : : : : : : 4.8 Transmission channel: general model : : : : : : : : : : : : : : Power ampliﬁer (HPA) : : : : : : : : : : : : : : : : Transmission medium : : : : : : : : : : : : : : : : : Additive noise : : : : : : : : : : : : : : : : : : : : : Phase noise : : : : : : : : : : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 5 Digital representation of waveforms 5.1 Analog and digital access : : : : : : : : : : : : : : : : : 5.1.1 Digital representation of speech : : : : : : : : : : Some waveforms : : : : : : : : : : : : : : : : Speech coding : : : : : : : : : : : : : : : : : : The interpolator ﬁlter as a holder : : : : : : : : Sizing of the binary channel parameters : : : : 5.1.2 Coding techniques and applications : : : : : : : : 5.2 Instantaneous quantization : : : : : : : : : : : : : : : : : 5.2.1 Parameters of a quantizer : : : : : : : : : : : : : 5.2.2 Uniform quantizers : : : : : : : : : : : : : : : : Quantization error : : : : : : : : : : : : : : : : Relation between 1, b and −sat : : : : : : : : Statistical description of the quantization noise Statistical power of the quantization error : : : Design of a uniform quantizer : : : : : : : : : Signaltoquantization error ratio : : : : : : : : Implementations of uniform PCM encoders : : 5.3 Nonuniform quantizers : : : : : : : : : : : : : : : : : : Three examples of implementation : : : : : : : 5.3.1 Companding techniques : : : : : : : : : : : : : : Signaltoquantization error ratio : : : : : : : : Digital compression : : : : : : : : : : : : : : : Signaltoquantization noise ratio mask : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
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5.6
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Optimum quantizer in the MSE sense : : : : : : : : : : : : Max algorithm : : : : : : : : : : : : : : : : : : : : : : Lloyd algorithm : : : : : : : : : : : : : : : : : : : : : : Expression of 3q for a very ﬁne quantization : : : : : : Performance of nonuniform quantizers : : : : : : : : : Adaptive quantization : : : : : : : : : : : : : : : : : : : : : : : : General scheme : : : : : : : : : : : : : : : : : : : : : : 5.4.1 Feedforward adaptive quantizer : : : : : : : : : : : : : : : Performance : : : : : : : : : : : : : : : : : : : : : : : : 5.4.2 Feedback adaptive quantizers : : : : : : : : : : : : : : : : Estimate of ¦s .k/ : : : : : : : : : : : : : : : : : : : : : Differential coding (DPCM) : : : : : : : : : : : : : : : : : : : : : 5.5.1 Conﬁguration with feedback quantizer : : : : : : : : : : : 5.5.2 Alternative conﬁguration : : : : : : : : : : : : : : : : : : 5.5.3 Expression of the optimum coefﬁcients : : : : : : : : : : : Effects due to the presence of the quantizer : : : : : : : 5.5.4 Adaptive predictors : : : : : : : : : : : : : : : : : : : : : Adaptive feedforward predictors : : : : : : : : : : : : : Sequential adaptive feedback predictors : : : : : : : : : Performance : : : : : : : : : : : : : : : : : : : : : : : : 5.5.5 Alternative structures for the predictor : : : : : : : : : : : Allpole predictor : : : : : : : : : : : : : : : : : : : : : Allzero predictor : : : : : : : : : : : : : : : : : : : : : Polezero predictor : : : : : : : : : : : : : : : : : : : : Pitch predictor : : : : : : : : : : : : : : : : : : : : : : APC : : : : : : : : : : : : : : : : : : : : : : : : : : : : Delta modulation : : : : : : : : : : : : : : : : : : : : : : : : : : 5.6.1 Oversampling and quantization error : : : : : : : : : : : : 5.6.2 Linear delta modulation (LDM) : : : : : : : : : : : : : : : LDM implementation : : : : : : : : : : : : : : : : : : : Choice of system parameters : : : : : : : : : : : : : : : 5.6.3 Adaptive delta modulation (ADM) : : : : : : : : : : : : : Continuously variable slope delta modulation (CVSDM) ADM with secondorder predictors : : : : : : : : : : : : 5.6.4 PCM encoder via LDM : : : : : : : : : : : : : : : : : : 5.6.5 Sigma delta modulation (6DM) : : : : : : : : : : : : : : : Coding by modeling : : : : : : : : : : : : : : : : : : : : : : : : : Vocoder or LPC : : : : : : : : : : : : : : : : : : : : : : RPE coding : : : : : : : : : : : : : : : : : : : : : : : : CELP coding : : : : : : : : : : : : : : : : : : : : : : : Multipulse coding : : : : : : : : : : : : : : : : : : : : : Vector quantization (VQ) : : : : : : : : : : : : : : : : : : : : : : 5.8.1 Characterization of VQ : : : : : : : : : : : : : : : : : : : Parameters determining VQ performance : : : : : : : : : Comparison between VQ and scalar quantization : : : : 5.3.2
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Optimum quantization : : : : : : : : : : : : : : : Generalized Lloyd algorithm : : : : : : : : : : 5.8.3 LBG algorithm : : : : : : : : : : : : : : : : : : Choice of the initial codebook : : : : : : : : : Description of the LBG algorithm with splitting Selection of the training sequence : : : : : : : 5.8.4 Variants of VQ : : : : : : : : : : : : : : : : : : Tree search VQ : : : : : : : : : : : : : : : : : Multistage VQ : : : : : : : : : : : : : : : : : Product code VQ : : : : : : : : : : : : : : : : 5.9 Other coding techniques : : : : : : : : : : : : : : : : : : Adaptive transform coding (ATC) : : : : : : : Subband coding (SBC) : : : : : : : : : : : : : 5.10 Source coding : : : : : : : : : : : : : : : : : : : : : : : 5.11 Speech and audio standards : : : : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : 5.8.2
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6 Modulation theory 6.1 Theory of optimum detection : : : : : : : : : : : : : : : : : : Statistics of the random variables fwi g : : : : : : : : Sufﬁcient statistics : : : : : : : : : : : : : : : : : : Decision criterion : : : : : : : : : : : : : : : : : : : Theorem of irrelevance : : : : : : : : : : : : : : : : Implementations of the maximum likelihood criterion Error probability : : : : : : : : : : : : : : : : : : : 6.1.1 Examples of binary signalling : : : : : : : : : : : : : : Antipodal signals (² D 1) : : : : : : : : : : : : : : Orthogonal signals (² D 0) : : : : : : : : : : : : : : Binary FSK : : : : : : : : : : : : : : : : : : : : : : 6.1.2 Limits on the probability of error : : : : : : : : : : : : Upper limit : : : : : : : : : : : : : : : : : : : : : : Lower limit : : : : : : : : : : : : : : : : : : : : : : 6.2 Simpliﬁed model of a transmission system and deﬁnition of binary channel : : : : : : : : : : : : : : : : : : : : : : : : Parameters of a transmission system : : : : : : : : : Relations among parameters : : : : : : : : : : : : : 6.3 Pulse amplitude modulation (PAM) : : : : : : : : : : : : : : : 6.4 Phaseshift keying (PSK) : : : : : : : : : : : : : : : : : : : : Binary PSK (BPSK) : : : : : : : : : : : : : : : : : Quadrature PSK (QPSK) : : : : : : : : : : : : : : : 6.5 Differential PSK (DPSK) : : : : : : : : : : : : : : : : : : : : 6.5.1 Error probability for an MDPSK system : : : : : : : : 6.5.2 Differential encoding and coherent demodulation : : : : Binary case (M D 2, differentially encoded BPSK) : Multilevel case : : : : : : : : : : : : : : : : : : : :
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AMPM or quadrature amplitude modulation (QAM) : : : : : : : : : : Comparison between PSK and QAM : : : : : : : : : : : : : 6.7 Modulation methods using orthogonal and biorthogonal signals : : : : 6.7.1 Modulation with orthogonal signals : : : : : : : : : : : : : : : Probability of error : : : : : : : : : : : : : : : : : : : : : : Limit of the probability of error for M increasing to inﬁnity 6.7.2 Modulation with biorthogonal signals : : : : : : : : : : : : : : Probability of error : : : : : : : : : : : : : : : : : : : : : : 6.8 Binary sequences and coding : : : : : : : : : : : : : : : : : : : : : : Optimum receiver : : : : : : : : : : : : : : : : : : : : : : 6.9 Comparison between coherent modulation methods : : : : : : : : : : : Tradeoffs for QAM systems : : : : : : : : : : : : : : : : : Comparison of modulation methods : : : : : : : : : : : : : 6.10 Limits imposed by information theory : : : : : : : : : : : : : : : : : Capacity of a system using amplitude modulation : : : : : : Coding strategies depending on the signaltonoise ratio : : : Coding gain : : : : : : : : : : : : : : : : : : : : : : : : : : Cutoff rate : : : : : : : : : : : : : : : : : : : : : : : : : : 6.11 Optimum receivers for signals with random phase : : : : : : : : : : : ML criterion : : : : : : : : : : : : : : : : : : : : : : : : : Implementation of a noncoherent ML receiver : : : : : : : Error probability for a noncoherent binary FSK system : : : Performance comparison of binary systems : : : : : : : : : 6.12 Binary modulation systems in the presence of ﬂat fading : : : : : : : : Diversity : : : : : : : : : : : : : : : : : : : : : : : : : : : 6.13 Transmission methods : : : : : : : : : : : : : : : : : : : : : : : : : : 6.13.1 Transmission methods between two users : : : : : : : : : : : : Three methods : : : : : : : : : : : : : : : : : : : : : : : : 6.13.2 Channel sharing: deterministic access methods : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 6.A Gaussian distribution function and Marcum function : : : : : : : : : : 6.A.1 The Q function : : : : : : : : : : : : : : : : : : : : : : : : : 6.A.2 The Marcum function : : : : : : : : : : : : : : : : : : : : : : 6.B Gray coding : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 6.C Baseband PPM and PDM : : : : : : : : : : : : : : : : : : : : : : : : Signaltonoise ratio : : : : : : : : : : : : : : : : : : : : : 6.D Walsh codes : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 6.6 7 Transmission over dispersive channels 7.1 Baseband digital transmission (PAM systems) : : Transmitter : : : : : : : : : : : : : : : Transmission channel : : : : : : : : : : Receiver : : : : : : : : : : : : : : : : : Power spectral density of a PAM signal : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
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Passband digital transmission (QAM systems) : : : : : : : : : Transmitter : : : : : : : : : : : : : : : : : : : : : : Power spectral density of a QAM signal : : : : : : : Three equivalent representations of the modulator : : Coherent receiver : : : : : : : : : : : : : : : : : : : 7.3 Baseband equivalent model of a QAM system : : : : : : : : : 7.3.1 Signal analysis : : : : : : : : : : : : : : : : : : : : : : Signaltonoise ratio : : : : : : : : : : : : : : : : : 7.3.2 Characterization of system elements : : : : : : : : : : Transmitter : : : : : : : : : : : : : : : : : : : : : : Transmission channel : : : : : : : : : : : : : : : : : Receiver : : : : : : : : : : : : : : : : : : : : : : : : 7.3.3 Intersymbol interference : : : : : : : : : : : : : : : : : Discretetime equivalent system : : : : : : : : : : : Nyquist pulses : : : : : : : : : : : : : : : : : : : : Eye diagram : : : : : : : : : : : : : : : : : : : : : : 7.3.4 Performance analysis : : : : : : : : : : : : : : : : : : Symbol error probability in the absence of ISI : : : : Matched ﬁlter receiver : : : : : : : : : : : : : : : : 7.4 Carrierless AM/PM (CAP) modulation : : : : : : : : : : : : : 7.5 Regenerative PCM repeaters : : : : : : : : : : : : : : : : : : : 7.5.1 PCM signals over a binary channel : : : : : : : : : : : Linear PCM coding of waveforms : : : : : : : : : : Overall system performance : : : : : : : : : : : : : 7.5.2 Regenerative repeaters : : : : : : : : : : : : : : : : : : Analog transmission : : : : : : : : : : : : : : : : : Digital transmission : : : : : : : : : : : : : : : : : : Comparison between analog and digital transmission Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 7.A Line codes for PAM systems : : : : : : : : : : : : : : : : : : 7.A.1 Line codes : : : : : : : : : : : : : : : : : : : : : : : : Nonreturntozero (NRZ) format : : : : : : : : : : : Returntozero (RZ) format : : : : : : : : : : : : : : Biphase (B ) format : : : : : : : : : : : : : : : : : Delay modulation or Miller code : : : : : : : : : : : Block line codes : : : : : : : : : : : : : : : : : : : Alternate mark inversion (AMI) : : : : : : : : : : : 7.A.2 Partial response systems : : : : : : : : : : : : : : : : : The choice of the PR polynomial : : : : : : : : : : : Symbol detection and error probability : : : : : : : : Precoding : : : : : : : : : : : : : : : : : : : : : : : Error probability with precoding : : : : : : : : : : : Alternative interpretation of PR systems : : : : : : :
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7.B Computation of Pe for some cases of interest 7.B.1 Pe in the absence of ISI : : : : : : : 7.B.2 Pe in the presence of ISI : : : : : : Exhaustive method : : : : : : : : Gaussian approximation : : : : : Worstcase limit : : : : : : : : : : Saltzberg limit : : : : : : : : : : GQR method : : : : : : : : : : : 7.C Coherent PAMDSB transmission : : : : : : General scheme : : : : : : : : : : Transmit signal PSD : : : : : : : Signaltonoise ratio : : : : : : : 7.D Implementation of a QAM transmitter : : : 7.E Simulation of a QAM system : : : : : : : :
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8 Channel equalization and symbol detection 8.1 Zeroforcing equalizer (LEZF) : : : : : : : : : : : : : : : : : 8.2 Linear equalizer (LE) : : : : : : : : : : : : : : : : : : : : : : 8.2.1 Optimum receiver in the presence of noise and ISI : : : Alternative derivation of the IIR equalizer : : : : : : Signaltonoise ratio : : : : : : : : : : : : : : : : 8.3 LE with a ﬁnite number of coefﬁcients : : : : : : : : : : : : : Adaptive LE : : : : : : : : : : : : : : : : : : : : : 8.4 Fractionally spaced equalizer (FSE) : : : : : : : : : : : : : : : Adaptive FSE : : : : : : : : : : : : : : : : : : : : : 8.5 Decision feedback equalizer (DFE) : : : : : : : : : : : : : : : Adaptive DFE : : : : : : : : : : : : : : : : : : : : : Design of a DFE with a ﬁnite number of coefﬁcients Design of a fractionally spaced DFE (FSDFE) : : : Signaltonoise ratio : : : : : : : : : : : : : : : : Remarks : : : : : : : : : : : : : : : : : : : : : : : : 8.6 Convergence behavior of adaptive equalizers : : : : : : : : : : Adaptive LE : : : : : : : : : : : : : : : : : : : : : Adaptive DFE : : : : : : : : : : : : : : : : : : : : : 8.7 LEZF with a ﬁnite number of coefﬁcients : : : : : : : : : : : 8.8 DFE: alternative conﬁgurations : : : : : : : : : : : : : : : : : DFEZF : : : : : : : : : : : : : : : : : : : : : : : : DFEZF as a noise predictor : : : : : : : : : : : : : DFE as ISI and noise predictor : : : : : : : : : : : : 8.9 Benchmark performance for two equalizers : : : : : : : : : : : Performance comparison : : : : : : : : : : : : : : : Equalizer performance for two channel models : : : 8.10 Optimum methods for data detection : : : : : : : : : : : : : : 8.10.1 Maximum likelihood sequence detection : : : : : : : : Lower limit to error probability using the MLSD criterion : : : : : : : : : : :
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The Viterbi algorithm (VA) : : : : : : : : : : : : : : : : : : : Computational complexity of the VA : : : : : : : : : : : : : : 8.10.2 Maximum a posteriori probability detector : : : : : : : : : : : : Statistical description of a sequential machine : : : : : : : : : The forwardbackward algorithm (FBA) : : : : : : : : : : : : Scaling : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Likelihood function in the absence of ISI : : : : : : : : : : : Simpliﬁed version of the MAP algorithm (MaxLogMAP) : : Relation between MaxLogMAP and LogMAP : : : : : : : : 8.11 Optimum receivers for transmission over dispersive channels : : : : : : Ungerboeck’s formulation of the MLSD : : : : : : : : : : : : 8.12 Error probability achieved by MLSD : : : : : : : : : : : : : : : : : : : Computation of the minimum distance : : : : : : : : : : : : : 8.13 Reduced state sequence detection : : : : : : : : : : : : : : : : : : : : : Reduced state trellis diagram : : : : : : : : : : : : : : : : : : RSSE algorithm : : : : : : : : : : : : : : : : : : : : : : : : : Further simpliﬁcation: DFSE : : : : : : : : : : : : : : : : : : 8.14 Passband equalizers : : : : : : : : : : : : : : : : : : : : : : : : : : : : 8.14.1 Passband receiver structure : : : : : : : : : : : : : : : : : : : : Joint optimization of equalizer coefﬁcients and carrier phase offset : : : : : : : : : : : : : : Adaptive method : : : : : : : : : : : : : : : : : : : : : : : : 8.14.2 Efﬁcient implementations of voiceband modems : : : : : : : : : 8.15 LE for voiceband modems : : : : : : : : : : : : : : : : : : : : : : : : : Detection of the training sequence : : : : : : : : : : : : : : : Computations of the coefﬁcients of a cyclic equalizer : : : : : Transition from training to data mode : : : : : : : : : : : : : Example of application: a simple modem : : : : : : : : : : : 8.16 LE and DFE in the frequency domain with data frames using cyclic preﬁx 8.17 Numerical results obtained by simulations : : : : : : : : : : : : : : : : QPSK transmission over a minimum phase channel : : : : : : QPSK transmission over a nonminimum phase channel : : : : 8PSK transmission over a minimum phase channel : : : : : : 8PSK transmission over a nonminimum phase channel : : : 8.18 Diversity combining techniques : : : : : : : : : : : : : : : : : : : : : : Antenna arrays : : : : : : : : : : : : : : : : : : : : : : : : : Combining techniques : : : : : : : : : : : : : : : : : : : : : Equalization and diversity : : : : : : : : : : : : : : : : : : : Diversity in transmission : : : : : : : : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 8.A Calculus of variations and receiver optimization : : : : : : : : : : : : : 8.A.1 Calculus of variations : : : : : : : : : : : : : : : : : : : : : : : Linear functional : : : : : : : : : : : : : : : : : : : : : : : : Quadratic functional : : : : : : : : : : : : : : : : : : : : : :
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8.A.2 Receiver optimization : : : : : : : : : : : : : : 8.A.3 Joint optimization of transmitter and receiver : : 8.B DFE design: matrix formulations : : : : : : : : : : : : 8.B.1 Method based on correlation sequences : : : : : 8.B.2 Method based on the channel impulse response and i.i.d. symbols : : : : : : : : : : : : : : : : 8.B.3 Method based on the channel impulse response and any symbol statistic : : : : : : : : : : : : : 8.B.4 FSDFE : : : : : : : : : : : : : : : : : : : : : 8.C Equalization based on the peak value of ISI : : : : : : 8.D Description of a ﬁnite state machine (FSM) : : : : : : :
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9 Orthogonal frequency division multiplexing 9.1 OFDM systems : : : : : : : : : : : : : : : : : : : : : : : : 9.2 Orthogonality conditions : : : : : : : : : : : : : : : : : : : : Time domain : : : : : : : : : : : : : : : : : : : : Frequency domain : : : : : : : : : : : : : : : : : ztransform domain : : : : : : : : : : : : : : : : : 9.3 Efﬁcient implementation of OFDM systems : : : : : : : : : OFDM implementation employing matched ﬁlters : Orthogonality conditions in terms of the polyphase components : : : : : OFDM implementation employing a prototype ﬁlter 9.4 Noncritically sampled ﬁlter banks : : : : : : : : : : : : : : 9.5 Examples of OFDM systems : : : : : : : : : : : : : : : : : Discrete multitone (DMT) : : : : : : : : : : : : : Filtered multitone (FMT) : : : : : : : : : : : : : : Discrete wavelet multitone (DWMT) : : : : : : : : 9.6 Equalization of OFDM systems : : : : : : : : : : : : : : : : Interpolator ﬁlter and virtual subchannels : : : : : Equalization of DMT systems : : : : : : : : : : : Equalization of FMT systems : : : : : : : : : : : : 9.7 Synchronization of OFDM systems : : : : : : : : : : : : : : 9.8 Passband OFDM systems : : : : : : : : : : : : : : : : : : : Passband DWMT systems : : : : : : : : : : : : : Passband DMT and FMT systems : : : : : : : : : Comparison between OFDM and QAM systems : : 9.9 DWMT modulation : : : : : : : : : : : : : : : : : : : : : : Transmit and receive ﬁlter banks : : : : : : : : : : Approximate interchannel interference suppression Perfect interchannel interference suppression : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
10 Spread spectrum systems 10.1 Spread spectrum techniques : : : : : : : : : : : : : : : : : : : : : : : : 10.1.1 Direct sequence systems : : : : : : : : : : : : : : : : : : : : : :
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Classiﬁcation of CDMA systems : : : : : : : : : Synchronization : : : : : : : : : : : : : : : : : : 10.1.2 Frequency hopping systems : : : : : : : : : : : : : Classiﬁcation of FH systems : : : : : : : : : : : 10.2 Applications of spread spectrum systems : : : : : : : : : : 10.2.1 Antijam communications : : : : : : : : : : : : : : 10.2.2 Multipleaccess systems : : : : : : : : : : : : : : : 10.2.3 Interference rejection : : : : : : : : : : : : : : : : 10.3 Chip matched ﬁlter and rake receiver : : : : : : : : : : : : Number of resolvable rays in a multipath channel Chip matched ﬁlter (CMF) : : : : : : : : : : : : 10.4 Interference : : : : : : : : : : : : : : : : : : : : : : : : : Detection strategies for multipleaccess systems : 10.5 Equalizers for singleuser detection : : : : : : : : : : : : : Chip equalizer (CE) : : : : : : : : : : : : : : : : Symbol equalizer (SE) : : : : : : : : : : : : : : 10.6 Block equalizer for multiuser detection : : : : : : : : : : : 10.7 Maximum likelihood multiuser detector : : : : : : : : : : : Correlation matrix approach : : : : : : : : : : : Whitening ﬁlter approach : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : :
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11 Channel codes 11.1 System model : : : : : : : : : : : : : : : : : : : : : : : : : : : : 11.2 Block codes : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 11.2.1 Theory of binary codes with group structure : : : : : : : : Properties : : : : : : : : : : : : : : : : : : : : : : : : : Parity check matrix : : : : : : : : : : : : : : : : : : : : Code generator matrix : : : : : : : : : : : : : : : : : : Decoding of binary parity check codes : : : : : : : : : : Cosets : : : : : : : : : : : : : : : : : : : : : : : : : : : Two conceptually simple decoding methods : : : : : : : Syndrome decoding : : : : : : : : : : : : : : : : : : : : 11.2.2 Fundamentals of algebra : : : : : : : : : : : : : : : : : : Modulo q arithmetic : : : : : : : : : : : : : : : : : : : Polynomials with coefﬁcients from a ﬁeld : : : : : : : : The concept of modulo in the arithmetic of polynomials Devices to sum and multiply elements in a ﬁnite ﬁeld : : Remarks on ﬁnite ﬁelds : : : : : : : : : : : : : : : : : : Roots of a polynomial : : : : : : : : : : : : : : : : : : Minimum function : : : : : : : : : : : : : : : : : : : : Methods to determine the minimum function : : : : : : Properties of the minimum function : : : : : : : : : : : 11.2.3 Cyclic codes : : : : : : : : : : : : : : : : : : : : : : : : : The algebra of cyclic codes : : : : : : : : : : : : : : : :
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Properties of cyclic codes : : : : : : : : : : : : : : : : Encoding method using a shift register of length r : : Encoding method using a shift register of length k : : Hard decoding of cyclic codes : : : : : : : : : : : : : Hamming codes : : : : : : : : : : : : : : : : : : : : : Burst error detection : : : : : : : : : : : : : : : : : : 11.2.4 Simplex cyclic codes : : : : : : : : : : : : : : : : : : : Relation to PN sequences : : : : : : : : : : : : : : : : 11.2.5 BCH codes : : : : : : : : : : : : : : : : : : : : : : : : An alternative method to specify the code polynomials Bose–Chaudhuri–Hocquenhem (BCH) codes : : : : : : Binary BCH codes : : : : : : : : : : : : : : : : : : : Reed–Solomon codes : : : : : : : : : : : : : : : : : : Decoding of BCH codes : : : : : : : : : : : : : : : : Efﬁcient decoding of BCH codes : : : : : : : : : : : : 11.2.6 Performance of block codes : : : : : : : : : : : : : : : : 11.3 Convolutional codes : : : : : : : : : : : : : : : : : : : : : : : : 11.3.1 General description of convolutional codes : : : : : : : : Parity check matrix : : : : : : : : : : : : : : : : : : : Generator matrix : : : : : : : : : : : : : : : : : : : : Transfer function : : : : : : : : : : : : : : : : : : : : Catastrophic error propagation : : : : : : : : : : : : : 11.3.2 Decoding of convolutional codes : : : : : : : : : : : : : Interleaving : : : : : : : : : : : : : : : : : : : : : : : Two decoding models : : : : : : : : : : : : : : : : : : Viterbi algorithm : : : : : : : : : : : : : : : : : : : : Forwardbackward algorithm : : : : : : : : : : : : : : Sequential decoding : : : : : : : : : : : : : : : : : : : 11.3.3 Performance of convolutional codes : : : : : : : : : : : 11.4 Concatenated codes : : : : : : : : : : : : : : : : : : : : : : : : Softoutput Viterbi algorithm (SOVA) : : : : : : : : : 11.5 Turbo codes : : : : : : : : : : : : : : : : : : : : : : : : : : : : Encoding : : : : : : : : : : : : : : : : : : : : : : : : The basic principle of iterative decoding : : : : : : : : The forwardbackward algorithm revisited : : : : : : : Iterative decoding : : : : : : : : : : : : : : : : : : : : Performance evaluation : : : : : : : : : : : : : : : : : 11.6 Iterative detection and decoding : : : : : : : : : : : : : : : : : 11.7 Lowdensity parity check codes : : : : : : : : : : : : : : : : : : Encoding procedure : : : : : : : : : : : : : : : : : : : Decoding algorithm : : : : : : : : : : : : : : : : : : : Example of application : : : : : : : : : : : : : : : : : Performance and coding gain : : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
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11.A Nonbinary parity check codes : : : : : : : : : : : : : Linear codes : : : : : : : : : : : : : : : : Parity check matrix : : : : : : : : : : : : : Code generator matrix : : : : : : : : : : : Decoding of nonbinary parity check codes : Coset : : : : : : : : : : : : : : : : : : : : Two conceptually simple decoding methods Syndrome decoding : : : : : : : : : : : : :
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12 Trellis coded modulation 12.1 Linear TCM for one and twodimensional signal sets : : : : 12.1.1 Fundamental elements : : : : : : : : : : : : : : : : : Basic TCM scheme : : : : : : : : : : : : : : : : : Example : : : : : : : : : : : : : : : : : : : : : : : 12.1.2 Set partitioning : : : : : : : : : : : : : : : : : : : : 12.1.3 Lattices : : : : : : : : : : : : : : : : : : : : : : : : 12.1.4 Assignment of symbols to the transitions in the trellis 12.1.5 General structure of the encoder/bitmapper : : : : : Computation of dfree : : : : : : : : : : : : : : : : 12.2 Multidimensional TCM : : : : : : : : : : : : : : : : : : : : Encoding : : : : : : : : : : : : : : : : : : : : : : Decoding : : : : : : : : : : : : : : : : : : : : : : 12.3 Rotationally invariant TCM schemes : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 13 Precoding and coding techniques for dispersive channels 13.1 Capacity of a dispersive channel : : : : : : : : : : : : : 13.2 Techniques to achieve capacity : : : : : : : : : : : : : : Bit loading for OFDM : : : : : : : : : : : : : Discretetime model of a single carrier system : Achieving capacity with a single carrier system 13.3 Precoding and coding for dispersive channels : : : : : : : 13.3.1 Tomlinson–Harashima (TH) precoding : : : : : : 13.3.2 TH precoding and TCM : : : : : : : : : : : : : : 13.3.3 Flexible precoding : : : : : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
14 Synchronization 14.1 The problem of synchronization for QAM systems : : : : : : 14.2 The phaselocked loop : : : : : : : : : : : : : : : : : : : : : 14.2.1 PLL baseband model : : : : : : : : : : : : : : : : : Linear approximation : : : : : : : : : : : : : : : : 14.2.2 Analysis of the PLL in the presence of additive noise Noise analysis using the linearity assumption : : : 14.2.3 Analysis of a secondorder PLL : : : : : : : : : : : : 14.3 Costas loop : : : : : : : : : : : : : : : : : : : : : : : : : :
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14.3.1 PAM signals : : : : : : : : : : : : : : : : : : : : : : : : 14.3.2 QAM signals : : : : : : : : : : : : : : : : : : : : : : : 14.4 The optimum receiver : : : : : : : : : : : : : : : : : : : : : : : Timing recovery : : : : : : : : : : : : : : : : : : : : Carrier phase recovery : : : : : : : : : : : : : : : : : 14.5 Algorithms for timing and carrier phase recovery : : : : : : : : : 14.5.1 ML criterion : : : : : : : : : : : : : : : : : : : : : : : : Assumption of slow time varying channel : : : : : : : 14.5.2 Taxonomy of algorithms using the ML criterion : : : : : Feedback estimators : : : : : : : : : : : : : : : : : : Earlylate estimators : : : : : : : : : : : : : : : : : : 14.5.3 Timing estimators : : : : : : : : : : : : : : : : : : : : : Nondata aided : : : : : : : : : : : : : : : : : : : : : Nondata aided via spectral estimation : : : : : : : : : Dataaided and datadirected : : : : : : : : : : : : : : Data and phasedirected with feedback: differentiator scheme : : : : : : : : : : : : Data and phasedirected with feedback: Mueller & Muller scheme : : : : : : : : : Nondata aided with feedback : : : : : : : : : : : : : 14.5.4 Phasor estimators : : : : : : : : : : : : : : : : : : : : : Data and timingdirected : : : : : : : : : : : : : : : : Nondata aided for MPSK signals : : : : : : : : : : : Data and timingdirected with feedback : : : : : : : : 14.6 Algorithms for carrier frequency recovery : : : : : : : : : : : : 14.6.1 Frequency offset estimators : : : : : : : : : : : : : : : : Nondata aided : : : : : : : : : : : : : : : : : : : : : Nondata aided and timingindependent with feedback : Nondata aided and timingdirected with feedback : : : 14.6.2 Estimators operating at the modulation rate : : : : : : : : Dataaided and datadirected : : : : : : : : : : : : : : Nondata aided for MPSK : : : : : : : : : : : : : : : 14.7 Secondorder digital PLL : : : : : : : : : : : : : : : : : : : : : 14.8 Synchronization in spread spectrum systems : : : : : : : : : : : 14.8.1 The transmission system : : : : : : : : : : : : : : : : : Transmitter : : : : : : : : : : : : : : : : : : : : : : : Optimum receiver : : : : : : : : : : : : : : : : : : : : 14.8.2 Timing estimators with feedback : : : : : : : : : : : : : Nondata aided: noncoherent DLL : : : : : : : : : : : Nondata aided MCTL : : : : : : : : : : : : : : : : : Data and phasedirected: coherent DLL : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
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15 Selftraining equalization 15.1 Problem deﬁnition and fundamentals : : : : : : : : : : : : : : : : : : : Minimization of a special function : : : : : : : : : : : : : : :
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15.2 Three algorithms for PAM systems : : : : : : : : : : : : : The Sato algorithm : : : : : : : : : : : : : : : : Benveniste–Goursat algorithm : : : : : : : : : : Stopandgo algorithm : : : : : : : : : : : : : : Remarks : : : : : : : : : : : : : : : : : : : : : : 15.3 The contour algorithm for PAM systems : : : : : : : : : : Simpliﬁed realization of the contour algorithm : : 15.4 Selftraining equalization for partial response systems : : : The Sato algorithm for partial response systems : Contour algorithm for partial response systems : 15.5 Selftraining equalization for QAM systems : : : : : : : : The Sato algorithm for QAM systems : : : : : : 15.5.1 Constant modulus algorithm : : : : : : : : : : : : : The contour algorithm for QAM systems : : : : Joint contour algorithm and carrier phase tracking 15.6 Examples of applications : : : : : : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : 15.A On the convergence of the contour algorithm : : : : : : : :
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16 Applications of interference cancellation 16.1 Echo and near–end crosstalk cancellation for PAM systems : Crosstalk cancellation and full duplex transmission Polyphase structure of the canceller : : : : : : : : Canceller at symbol rate : : : : : : : : : : : : : : Adaptive canceller : : : : : : : : : : : : : : : : : Canceller structure with distributed arithmetic : : : 16.2 Echo cancellation for QAM systems : : : : : : : : : : : : : 16.3 Echo cancellation for OFDM systems : : : : : : : : : : : : : 16.4 Multiuser detection for VDSL : : : : : : : : : : : : : : : : : 16.4.1 Upstream power backoff : : : : : : : : : : : : : : : 16.4.2 Comparison of PBO methods : : : : : : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : :
17 Wired and wireless network technologies 17.1 Wired network technologies : : : : : : : : : : : : : : : : : : : : : 17.1.1 Transmission over unshielded twisted pairs in the customer service area : : : : : : : : : : : : : : : : : : : : : : : : : Modem : : : : : : : : : : : : : : : : : : : : : : : : : : Digital subscriber line : : : : : : : : : : : : : : : : : : 17.1.2 High speed transmission over unshielded twisted pairs in local area networks : : : : : : : : : : : : : : : : : : : : 17.1.3 Hybrid ﬁber/coaxial cable networks : : : : : : : : : : : : : Ranging and power adjustment for uplink transmission : : : : : : : : : : : : : : : : :
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Contents
17.2 Wireless network technologies : : : : : : : 17.2.1 Wireless local area networks : : : Medium access control protocols 17.2.2 MMDS and LMDS : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : 17.A Standards for wireless systems : : : : : : 17.A.1 General observations : : : : : : : Wireless systems : : : : : : : : Modulation techniques : : : : : Parameters of the modulator : : Cells in a wireless system : : : 17.A.2 GSM standard : : : : : : : : : : : System characteristics : : : : : : Radio subsystem : : : : : : : : GSMEDGE : : : : : : : : : : : 17.A.3 IS136 standard : : : : : : : : : : 17.A.4 JDC standard : : : : : : : : : : : 17.A.5 IS95 standard : : : : : : : : : : : 17.A.6 DECT standard : : : : : : : : : : 17.A.7 HIPERLAN standard : : : : : : :
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18 Modulation techniques for wireless systems 18.1 Analog frontend architectures : : : : : : : : : : : : : : : : : Conventional superheterodyne receiver : : : : : : : Alternative architectures : : : : : : : : : : : : : : Direct conversion receiver : : : : : : : : : : : : : Single conversion to lowIF : : : : : : : : : : : : Double conversion and wideband IF : : : : : : : : 18.2 Three noncoherent receivers for phase modulation systems : 18.2.1 Baseband differential detector : : : : : : : : : : : : : 18.2.2 IFband (1 Bit) differential detector (1BDD) : : : : : Performance of MDPSK : : : : : : : : : : : : : : 18.2.3 FM discriminator with integrate and dump ﬁlter (LDI) 18.3 Variants of QPSK : : : : : : : : : : : : : : : : : : : : : : : 18.3.1 Basic schemes : : : : : : : : : : : : : : : : : : : : : QPSK : : : : : : : : : : : : : : : : : : : : : : : : Offset QPSK or staggered QPSK : : : : : : : : : : Differential QPSK (DQPSK) : : : : : : : : : : : : ³=4DQPSK : : : : : : : : : : : : : : : : : : : : 18.3.2 Implementations : : : : : : : : : : : : : : : : : : : : QPSK, OQPSK, and DQPSK modulators : : : : : ³=4DQPSK modulators : : : : : : : : : : : : : : 18.4 Frequency shift keying (FSK) : : : : : : : : : : : : : : : : : 18.4.1 Power spectrum of MFSK : : : : : : : : : : : : : :
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Power spectrum of noncoherent binary FSK : : Power spectrum of coherent MFSK : : : : : : : 18.4.2 FSK receivers and corresponding performance : : : Coherent demodulator : : : : : : : : : : : : : : Noncoherent demodulator : : : : : : : : : : : : Limiterdiscriminator FM demodulator : : : : : : 18.5 Minimum shift keying (MSK) : : : : : : : : : : : : : : : : 18.5.1 Power spectrum of continuousphase FSK (CPFSK) 18.5.2 The MSK signal viewed from two perspectives : : Phase of an MSK signal : : : : : : : : : : : : : MSK as binary CPFSK : : : : : : : : : : : : : : MSK as OQPSK : : : : : : : : : : : : : : : : : Complex notation of an MSK signal : : : : : : : 18.5.3 Implementations of an MSK scheme : : : : : : : : 18.5.4 Performance of MSK demodulators : : : : : : : : : MSK with differential precoding : : : : : : : : : 18.5.5 Remarks on spectral containment : : : : : : : : : : 18.6 Gaussian MSK (GMSK) : : : : : : : : : : : : : : : : : : 18.6.1 GMSK via CPFSK : : : : : : : : : : : : : : : : : 18.6.2 Power spectrum of GMSK : : : : : : : : : : : : : 18.6.3 Implementation of a GMSK scheme : : : : : : : : Conﬁguration I : : : : : : : : : : : : : : : : : : Conﬁguration II : : : : : : : : : : : : : : : : : : Conﬁguration III : : : : : : : : : : : : : : : : : 18.6.4 Linear approximation of a GMSK signal : : : : : : Performance of GMSK demodulators : : : : : : Performance of a GSM receiver in the presence of multipath : : : : : : : : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : 18.A Continuous phase modulation (CPM) : : : : : : : : : : : : Alternative deﬁnition of CPM : : : : : : : : : : Advantages of CPM : : : : : : : : : : : : : : : 19 Design of high speed transmission systems over unshielded twisted pair cables 19.1 Design of a quaternary partial response classIV system sion at 125 Mbit/s : : : : : : : : : : : : : : : : : : : : Analog ﬁlter design : : : : : : : : : : : : : : Received signal and adaptive gain control : : Nearend crosstalk cancellation : : : : : : : Decorrelation ﬁlter : : : : : : : : : : : : : : Adaptive equalizer : : : : : : : : : : : : : : Compensation of the timing phase drift : : :
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2 Design of a dual duplex transmission system at 100 Mbit/s : : Dual duplex transmission : : : : : : : : : : : : : : Physical layer control : : : : : : : : : : : : : : : : Coding and decoding : : : : : : : : : : : : : : : : 19.1.A Interference suppression : : : : : : : : : : : : : : : : : : : : Index : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 1253 1253 1255 1255 1258 1261 1262 1263 1263 1265 1266 1269 1269 1270 1272 1273 1274 1274 1277 .xxviii Contents Adaptive equalizer coefﬁcient adaptation : : : : : : Convergence behavior of the various algorithms : : 19.2.1 VLSI implementation : : : : : : : : : : : : : : : : : Adaptive digital NEXT canceller : : : : : : : : : : Adaptive digital equalizer : : : : : : : : : : : : : : Timing control : : : : : : : : : : : : : : : : : : : Viterbi detector : : : : : : : : : : : : : : : : : : : 19.1 Signal processing functions : : : : : : : : : : : : : : The 100BASET2 transmitter : : : : : : : : : : : : The 100BASET2 receiver : : : : : : : : : : : : : Computational complexity of digital receive ﬁlters : Bibliography : : : : : : : : : : : : : : : : : : : : : : : : : : : : : Appendices : : : : : : : : : : : : : : : : : : : : : : : : : : : : : 19.
Preface The motivation for this book is twofold. Italy. The ﬁrst three sections of Chapter 11. Track 1. It covers Chapter 1. we present a discussion of fundamental algorithms and structures for telecommunication technologies. where a sequence of information symbols is sent over a transmission channel. The representation of waveforms by sequences of binary symbols is treated in Chapter 5. First. for a ﬁrst course it is suggested that emphasis be placed on PCM. Chapter 6 examines the fundamental principles of a digital transmission system. we provide a didactic tool to students of communications systems. The book is divided into nineteen chapters. together with a brief introduction to computer simulations. On the other hand. The Wiener ﬁlter and the linear prediction theory. as well as our professional experience acquired in industrial research laboratories. We brieﬂy indicate four tracks corresponding to speciﬁc areas and course work offered. A discussion of the characteristics of transmission media follows in Chapter 4. Track 2. focuses on modulation techniques. On the one hand. The contents reﬂect our experience in teaching courses on Algorithms for Telecommunications at the University of Padova. Track 1 includes the basic elements for a ﬁrst course on telecommunications. The text explains the procedures for solving problems posed by the design of systems for reliable communications over wired or wireless channels. we focus on fundamental developments in the ﬁeld in order to provide the reader with the necessary insight to design essential elements of various communications systems. are dealt with in Chapter 2. which is an extension of Track 1. In particular. Track 2. We refer to Shannon theorem to establish the maximum bit rate that can be transmitted reliably over a noisy channel. parametric models of random processes are analyzed in Chapter 1. In this track we focus on the description of noise in electronic devices and on the laws of propagation in transmission lines and radio channels. with an emphasis on secondorder statistical descriptions. which constitute fundamental elements for receiver design. Signal dispersion caused by a transmission channel is then analyzed in Chapter 7. Chapter 3 lists iterative methods to achieve the objectives stated . where we introduce methods for increasing transmission reliability by exploiting the redundancy added to the information bits. which we regard as an introduction to the remaining tracks. conclude the ﬁrst track. which recalls fundamental concepts on signals and random processes. Next. Examples of elementary and practical implementations of transmission systems are presented.
The principles of multicarrier and spread spectrum modulation techniques. and considerably reduces development time. are essential for this track. ﬁxed ﬁltering. Track 3. and speciﬁc examples of system design are given in Chapters 18 and 19. are investigated in depth in Chapters 9 and 10. which are increasingly being adopted in communications systems. the desired signal at the receiver is often disturbed by other transmissions taking place simultaneously. and of Chapter 8. Therefore the algorithmic aspects of a transmission system are becoming increasingly important. channel equalization is examined as a further application of the Wiener ﬁlter.xxx Preface in Chapter 2. An overview of wired and wireless access technologies appears in Chapter 17. are also discussed. The assumption that the transmission channel characteristics are known a priori is removed in Chapter 15. we introduce multicarrier modulation techniques. Hardware devices are assigned wherever possible only the functions of analog frontend. which can be utilized for more than one transmission standard. A further method to mitigate channel dispersion is precoding. We observe the trend towards implementing transceiver functions using digital signal processors. Applications of interference cancellation and multiuser detection are addressed in Chapter 16. which are preferable for transmission over very dispersive channels and/or applications that require ﬂexibility in spectral allocation. Track 4 addresses various challenges encountered in designing wired and wireless communications systems. where blind equalization techniques are discussed. which illustrates speciﬁc modulation techniques developed for mobile radio applications. with emphasis to applications for simultaneous channel access by several users that share a wideband channel. This approach enhances the ﬂexibility of transceivers. These applications are further developed in the ﬁrst two sections of Chapter 16. The operations of systems that employ joint precoding and channel coding are explained in Chapter 13. more sophisticated methods of equalization and symbol detection. The design of the receiver frontend. In line with the above considerations. as well as the algorithms introduced in Chapter 8. for example channel identiﬁcation and interference cancellation. are dealt with in Chapter 14. Track 3 begins with a review of Chapters 2 and 3. This is followed by Chapter 18. In Chapter 10 spread spectrum systems are examined. In the second part of the chapter. The inherent narrowband interference rejection capabilities of spread spectrum systems. and digitaltoanalog and analogtodigital conversion. . respectively. which illustrate the fundamental principles of transmission system design. are analyzed. Cancellation techniques to suppress interference signals are treated in Chapter 16. Track 4. In the ﬁrst part of Chapter 8. as well as various methods for timing and carrier recovery. as well as various applications of the Wiener ﬁlter. Channel coding techniques to improve the reliability of transmission are investigated in depth in Chapters 11 and 12. In Chapter 9. The elements introduced in Chapters 2 and 3. Initially singlecarrier modulation systems are considered. as well as their implementations. Because of electromagnetic coupling. which rely on the Viterbi algorithm and on the forwardbackward algorithm. which investigates individual building blocks for channel equalization and symbol detection.
A special acknowledgment goes to our colleagues Werner Bux and Evangelos Eleftheriou of the IBM Zurich Research Laboratory. the editing of the various chapters would never have been completed without the contributions of numerous students in our courses on Algorithms for Telecommunications. Stefano Tomasin. Antonio Mian. Andrea Galtarossa. we nevertheless express our sincere gratitude. Nevio Benvenuto Giovanni Cherubini . Charlotte Bolliger and Lilli M. for their continuing support. Giovanna Sostrato. our thanks also go to Jane Frankenﬁeld Zanin for her help in translating the text into English. We are pleased to thank the following colleagues for their invaluable assistance throughout the revision of the book: Antonio Assalini.Acknowledgements We gratefully acknowledge all who have made the realization of this book possible. Elena Costa. In particular. the contribution of Barbara Sicoli was indispensable. and Silvano Pupolin of the University of Padua. Antonio Salloum. We gratefully acknowledge our colleague and mentor Jack Wolf for letting us include his lecture notes in the chapter on channel codes. Roberto Rinaldo. Paola Bisaglia. Andrea Scaggiante. We also thank Christian Bolis and Chiara Paci for their support in developing the software for the book. Roberto Corvaja. Fortunato Santucci. Although space limitations preclude mentioning them all by name. For text processing of the Italian version. Carlo Monti. Ezio Obetti. Giancarlo Calvagno. Giulio Colavolpe. and Urs Bitterli and Darja Kropaci for their help with the graphics editing. Pavka for their assistance in administering the project. Alberto Bononi. and Luciano Tomba. Riccardo Rahely.
" 1 E Z Á Â.' N 4 O 5 P P T Y 8 X 9 nu xi omicron pi rho sigma tau upsilon phi chi psi omega . a table containing the Greek alphabet is included.To make the reading of the adopted symbols easier. # Ã Ä ½ ¼ H 2 I K 3 M alpha beta gamma delta epsilon zeta eta theta iota kappa lambda mu ! ¹ ¾ o ³ ². & − × . The Greek alphabet Þ þ A B 0 Ž ž. % ¦.
1 Signal space Deﬁnition 1. MA and especially AR models). 1. There exists a unique vector 0.3) (1. Addition is associative x C . ARMA. z and 0 be elements of a linear space.Chapter 1 Elements of signal theory In the present chapter we recall fundamental concepts of signal theory and random processes. those with complexvalued components. and the product of a vector by a scalar coincides with the vector obtained by multiplying each component for that scalar. while others will ﬁnd it a useful incentive for further indepth study.. Addition is commutative xCyDyCx 2.2) (1.and secondorder ergodic processes (periodogram. we will begin with the deﬁnition of signal space and its discrete representation. 1. Properties of a linear space Let x. called null. such that 0CxDx (1. In any event.e. IIR and FIR impulse responses) and signals (complex representation of passband signals and the baseband equivalent). We will conclude with the study of random processes. and Þ and þ be complex numbers (scalars). In our case of particular interest is the set of complex vectors. The Euclidean space is an example of linear space in which the sum of two vectors coincides with the vector obtained by adding the individual components. then move to the study of discretetime linear systems (discrete Fourier transforms. the sum between vectors and the multiplication of a vector by a scalar. with emphasis on the statistical estimation of ﬁrst. for which we recommend the items in the bibliography. A majority of readers will simply ﬁnd this chapter a review of known principles. in an Euclidean space.1 A linear space is a set of elements called vectors. together with two operators deﬁned over the elements of the set.y C z/ D .x C y/ C z 3. y.1) . correlogram. i.
whose elements are the signals fx.6) (1.5) (1. t 2< (1.9) where Tc is the sampling period or interval. Multiplication by scalars is associative Þ.k/g. 1 Later a discretetime signal will be indicated simply as fx. omitting the indication of the sampling period.8) A geometrical interpretation of the two elementary operations in a twodimensional Euclidean space is given in Figure 1. .þx/ D . Two other examples of linear spaces are: the discretetime signal space (an Euclidean space with inﬁnite dimensions). called additive inverse.Þþ/x In particular. the Euclidean space is an example of a linear space.x C y/ D Þx C Þy .Þ C þ/x D Þx C þx 0x D 0 (1.1. we have 1x D x 6.t/ where < denotes the set of real numbers.7) (1. there is a unique vector x.4) x C .10) Figure 1. For each x.kTc /g k integer (1. Distributive laws Þ. In general. such that (1.1. Elements of signal theory 4. we will indicate by fxk g a sequence of real or complex numbers not necessarily generated at instants kTc . As previously mentioned.1 and the continuoustime signal space.2 Chapter 1. whose elements are the signals x. x/ D 0 5. Geometrical interpretation in the twodimensional space of the sum of two vectors and the multiplication of a vector by a scalar.
2.2. we indicate with hx. . yi D jjxjj cos Â jjyjj is the length of the projection of x onto y. yi D 0.13) Observation 1. y I ]T . jjxjj is the norm of x. Note that hx. yi is real.2 Two vectors x and y are orthogonal (x ? y) if hx. yi D jjxjj jjyjj cos Â where jjxjj denotes the norm or length of the vector x. that is the vector length. : : : . (I=2) (1.1.1. Signal space 3 Inner product In an I dimensional Euclidean space.12). : : : . Deﬁnition 1. Geometrical representation of the inner product between two vectors.11) If hx. x I ]T and y D [y1 . that is if the angle they form is 90Ž .12) jxi j2 D jjxjj2 (1. 2 Henceforth: T stands for transpose.2 given the two vectors x D [x1 . Ł for complex conjugate and H for transpose complex conjugate or Hermitian. represented in Figure 1. hx. xi D I X i D1 (1. that is obtained from the relation: hx. yi the inner product: hx.1 From (1. there is an important geometrical interpretation of the inner product in the Euclidean space.14) y y x θ x Figure 1. yi D I X i D1 xi yiŁ (1.
t/j2 dt < 1 (1. is deﬁned by ² Z C1 1 if i D j Ł (1.16) for continuoustime signals. In both cases it is assumed that the energy of signals is ﬁnite. 4. deﬁning the inner product as C1 X kD 1 x. xi D 1 C1 jx.t/g. xiŁ . Given a ﬁniteenergy signal x. i 2 I. 2.t/ y Ł . 5. zi.k/ (1. hÞx. zi C hy. .t/j2 dt < 1 Z and C1 1 jy. yi.17) Recall that the inner product enjoys the following properties: 1. Hence. yi D Þhx. with K a complex scalar.t/j2 dt < 1 (1. Equality holds if and only if x D Ky.t/ denotes the Dirac delta. hx.2 Discrete signal representation Let us consider the problem of associating a sequence (possibly ﬁnite) of numbers with a continuoustime signal.3 A basis of orthonormal signals (orthogonal signals with unit norm) f i . t 2 <. whereas Ž.15) for discretetime signals. for continuoustime signals it must be: Z C1 1 jx. Žn is the Kronecker delta.t/ dt (1.18) h i. 3. where I is a ﬁnite or numerable set. xi > 0 8x 6D 0.t/ dt D Ži j D 0 if i 6D j 1 In this text. t 2 <. yij Ä jjxjjjjyjj. Z E x D hx.4 Chapter 1. (Schwarz inequality) jhx. Elements of signal theory We can extend these concepts to a signal space. yi D hy. and Z C1 1 x. hx.t/ j .k/ y Ł . ji D i .19) 3 This procedure can easily be extended to discretetime signals.t/. zi D hx. 1. hx C y.
is a complete basis for x.t/g.xi ci /.28) If E e D 0. The minimum of (1. i i D 1 and Ee D E x where Ex D O X i 2I Ex O jxi j2 (1. xi D Ex C By adding and subtracting jci j2 .t/.t/ (1. Discrete signal representation 5 we want to express x.t/ dt i 2I (1.27) (1. xi O O O xi ciŁ xiŁ ci / (1. i 2 I.t/ iŁ .1. xi O X i 2I .20) is by minimizing the energy of e.26) xi D ci D hx. as a linear combination of the functions f i . Consider the signal X x.t/ D i 2I .25) is obtained if the second term is zero.t/ D O xi i .25) as jxi j2 xi ciŁ xiŁ ci C jci j2 D jxi ci j2 D .t/þ dt (1.20) i 2I If we deﬁne the error as e. Hence the coefﬁcients to be determined are given by Z C1 x. and x.t/ x.21) the most common method to determine the coefﬁcients fxi g in (1.t/ D x. and express E e as E e D hx x.t/g. then f i . x O xi O hx.23) D hx. deﬁned as þ2 Z C1 þ þ þ X þ þ E e D he.29) x. i 2 I. t 2 <.t/ (1. hx.2.22) þx.xi xiŁ one ﬁnds X jxi Ee D E x C i 2I ci j2 X i 2I jci j2 (1.24) ii i 2I (1.t/ can be expressed as X xi i .t/ O (1.t/. ei D xi i .xi ci /Ł . xi C hx.t/ þ 1 þ i 2I Let ci D hx. t 2 <.
20) with fxi D ci g. given by xi D hx.t/.t/g. As an example. i 2 I.26) satisﬁes the following important property: he. which form a complete basis for x. i 2 I. ii D hx x. from (1. t 2 <. then x.6 Chapter 1. t 2 <.t/g.t/. onto the space spanned by the signals f i . Moreover. i 2 I.3. 1 and 2. Therefore. i 2 I. the space whose signals are expressed as linear combinations of f i .32) φ2 x e ^ x φ 1 Figure 1. ii (1. for a generic signal x and for a basis formed by two signals one gets the geometrical representation in Figure 1.31) that is e ? i .29) X jxi j2 (1.t/.t/ can be represented by a sequence of numbers fxi g. O ii D xi xi D 0 8i 2 I (1. If E e D 0.3 The signal x. i 2 I.t/g. is called the projection of O x. .3. the signal x. Signal representation Deﬁnition 1. belongs to the space spanned by f i . that is. given by (1. Geometrical representation of the projection of x onto the space spanned by 1 and 2 . t 2 <. 8i 2 I. Elements of signal theory where equality must be intended in terms of quadratic norm. given a sequence of orthonormal signals.t/.30) Ex D i 2I that is the energy of the signal coincides with the sum of the squares of the coefﬁcient amplitudes. The principle of orthogonality The vector identiﬁed by the optimum coefﬁcients (1.
35) In particular. the Euclidean distance between two signals coincides with the Euclidean distance between the corresponding vectors.x.2. Then (1.37) D X i 2I jxi yi j2 2 D d. Moreover. observing (1. with Â the angle formed by x and y. . x 0 .t/ D yi i . y/ D E x C E y 2Re hx. yi (1. yi jjxjj jjyjj (1. it is sufﬁcient that their inner product is real. we have the following correspondence between a signal and its vector representation: x. yi D 1 C1 x. x/. xi D jjxjj2 D E x (1.38) In other words. If ² is real.x.x.t/ y Ł . Z Ex D 1 C1 jx.40) be the correlation coefﬁcient between x and y. yi (1.x.t/ !1 y.41) We note that if ² is real. Let ²D hx.12) ² D cos Â . the real and the imaginary part of c.1.39) O In particular we have E e D d 2 .t/ t 2< ! x D [: : : .36) We introduce now the Euclidean distance between two signals: d.33) It is useful to analyze the inner product between signals in terms of the corresponding vector representations.t/ and y.x. the signals are not necessarily real. x 1 . x 1 . the following relation holds:4 ð Ł d 2 . Discrete signal representation 7 In short.t/j2 dt D X i 2I jxi j2 D hx.t/ D xi i .t/j dt 2 Ã1 2 (1.34) i 2I i 2I then Z hx. y/ (1. 4 The symbols Re [c] and Im [c] denote.39) becomes: p d 2 . Let X X x. y/ D E x C E y 2² E x E y (1.t/ dt D hx. y/ D p ÂZ Ex y C1 D 1 jx.t/ (1. respectively. : : :] T (1.
t/ D 1 2B ³ .t/ at the time instants t D i Tc .42) Tc D i . Let 0 1 . from (1.t/ is the projection of s2 upon 0 . Elements of signal theory Example 1. in fact.t/ D s2 . in (1.t/ D qi (1. 1 i 1 .50) . the orthonormal functions obtained from f i0 . M (1. 1i D hs2 D hs2 . 1.8 Chapter 1.t/.44) a procedure to derive an orthonormal basis for this set is now outlined.48) it is easy to see that h 0 2.46) Then it follows 1 .2.t/g a set of orthogonal functions and by 8 9 < 0 .47) 2.t/ D q2 0 E 2 (1. 1i 1. : : : . The coefﬁcients are given by the samples of x.t/ 0 .t/g.t/ D q1 0 E 1 (1.48) 2 .45) : i E i0 .t/ hs2 . with ﬁnite bandwidth B (see (1. 1i 0 2 ? 1.4. xi D x.t/g m D 1.t i Tc / Tc forms a complete orthogonal basis for x.t/ 1 i 1 . Let 0 2 . it can be shown that for a real valued signal x. As illustrated in Figure 1.48) hs2 . Then.140)).t/ D s1 .i Tc / (1. t 2 <.t/ = .t/ (1.43) Gram–Schmidt orthonormalization procedure Given a set of M signals fsm .1 Resorting to the sampling theorem. 1i hs2 .t i Tc / 1 Tc (1. 2. hs2 . the sequence of functions Â Ã 1 sin ³ .t/ (1.49) 1. 1i D0 (1. We indicate by f i0 .
m D 1. in any case the reciprocal distances between signals remain unchanged. we choose i .54) In such a case.t/ (1.1.3 The number of dimensions I of f i . Let us look at a few examples of discrete representation of a set of signals. limited to φ 0 and φ 0 .t/ D 0 8t (1. : : : . Obviously.t/ is not identically zero.52) and if 0 i . not all equal to zero. : : : . 2.4.51) then one gets In general 0 3 ? 1 and ? 0 i . are linearly dependent. i in Figure 1. 1 i 1 . such that M X mD1 cm sm .4.t/ (1.t/g is identically zero.t/ 0 3 hs3 .t/g is not unique. Let 0 3 .t/ D s3 . ji j . 3.2.t/ hs3 .53) 1. Discrete signal representation 9 Figure 1.t/ 0 .t/ D si . M. it happens that in (1. that is if there exists a set of coefﬁcients. a null signal cannot be an element of the basis.t/ D qi E i0 (1. Observation 1.t/ i 1 X jD1 hsi . 2. Geometrical representation of the GramSchmidt orthonormalization procedure.52) for some i the signal f i0 .t/g.2 The set of f i .t/g can be lower than M if the signals fsm . The procedure is represented geometrically It follows that i ? j for j D 1. 2 3 Observation 1. 2 i 2 . .
2.5.6. is the following: 8 T 2 < p sin 2³ t 0<t < T 2 (1.t/ D : 0 8 < A sin 2³ t T s2 .t/ D : 0 8 < A sin 2³ t T s3 .10 Chapter 1. an orthonormal basis.55) (1.56) (1. . The three signals.t/ D : 0 elsewhere 8 2 T < p sin 2³ t <t <T T 2 (1.57) which are depicted in Figure 1.2 For the three signals 8 < A sin 2³ t T s1 .58) T 1 .5.t/ D : 0 T 2 elsewhere 0<t < 0<t <T elsewhere T <t <T 2 elsewhere (1. represented in Figure 1.t/ D : 0 elsewhere A A s1(t) 0 s (t) 0 T 0 −A 2 −A 0 T t t A s (t) 0 3 −A 0 T t Figure 1. Elements of signal theory Example 1.59) T 2 .
Vector representation or constellation of signals of Figure 1.t/ (1.t/ D T 2 1 . Moreover. .t/ (1.t/ D Ap T 2 Ap s2 . Discrete signal representation 11 _ _ 2/ \T _ _ 2/ \T φ1(t) φ2(t) 0 0 _ _ −2/ \T 0 _ _ −2/ \T T 0 T t t Figure 1.t/ D T 2 Ap s3 .64) 2 2 Ä ½ Ap T s3 .t/ ! s3 D 0.5.7. T (1.0 (1.1. T (1.60) Ap T 2 2 .6.7): Ä ½T Ap s1 . s1 .63) 2 Ä ½ Ap Ap T s2 .t/ ! s1 D T.t/ ! s2 D T.5.65) 2 φ2 s3 A T 2 s2 s1 0 A T 2 φ1 Figure 1.t/ 1 .61) (1.2.62) from which the correspondence between signals and their vector representation is (see Figure 1.t/ C 2 . Orthonormal basis for signals of Figure 1.
67) 0<t <T elsewhere A A s (t) 0 s (t) 0 T/2 0 1 −A 2 −A T 0 T/2 T t t A A s (t) 0 s (t) 0 0 3 −A 4 −A T/2 T 0 T/2 T t t Figure 1. 3. to determine the basis functions we write sm .2. Elements of signal theory We note that the three signals are represented as a linear combination of only two functions (I D 2).t/ D A cos m cos.3 (4PSK) Given the set of four signals 8 Â Â < A cos 2³ f 0 t C m sm .2³ f 0 t/ 2 2 2 2 We choose the following basis: 8 r > < C 2 cos. 2. Deﬁnition 1.2³ f 0 t/ A sin m sin.t/ D > : 0 (1.8.66) depicted in Figure 1.2³ f t/ 0 T 1 .t/ D : 0 1 2 Ã ³ 2 Ã 0<t <T elsewhere m D 1.t/ as ÄÂ ÄÂ Ã ½ Ã ½ 1 ³ 1 ³ sm . Example 1. 4 (1.12 Chapter 1.4 The vector representation of a set of M signals is often called a signal constellation. Modulated 4PSK signals for f0 D 1=T.8. .
69) D sin2 2³ f 0 T 2³ f 0 T (1.3.1. 2. . and E i ' 1 for i D 1.t/ D 8 > < > : 0 r 2 sin.t/ form an othonormal basis for the four signals in Figure 1. also called analog ﬁlter. Continuoustime linear systems 13 φ2 s2 A 2 s1 T 0 A 2 T φ1 s3 s4 Figure 1.2³ f 0 t/ T 0<t <T elsewhere (1.68) One ﬁnds that E and h Hence if f0 D k (k integer) 2T or f0 × 1 T (1. 1. and 2 . where x and y are the input and output signals.9.8. whose constellation is given in Figure 1. respectively.71) 1.10. 1 .9. and h denotes the ﬁlter impulse response. 2 i ' 0.t/ and 2 . 4PSK constellation.3 Continuoustime linear systems A timeinvariant continuoustime continuousamplitude linear system. is represented in Figure 1. 2i 1 D1C sin ³ 4 f 0 T ³ 4 f0 T E 2 D1 sin ³ 4 f 0 T ³ 4 f0 T (1. Under these conditions for f 0 .70) then it results h 1 .
t 2 <.74) The inverse Fourier transform is given by Z 1 x. f / D X .t/ D X .t/ e j2³ ft dt f 2< 1 (1.75) In the frequency domain.73) becomes Y.5 Deﬁnition 1. f / f 2< (1.t/] D x. f / e j2³ ft d f 1 (1.− / x.t − / d− 1 1 (1.t/.73) (1.72).76) where H is the ﬁlter frequency response. f /j.14 Chapter 1.5 We introduce two functions that will be extensively used: 8 F Â Ã < 1 jfj< f 2 D rect : F 0 elsewhere (1. f / H.t/ where the symbol ‘Ł’ denotes the convolution operation (1. Z C1 X . We also introduce the Fourier transform of the signal x.t/ D 1 0 1 1 t >0 t <0 t >0 t <0 . Analog ﬁlter as a timeinvariant linear system with continuous domain.10. Elements of signal theory x(t) h y(t) Figure 1. General properties of the Fourier transform are given in Table 1.72) In short we will use the notation y.77) 5 Two important functions that will be used very often are: ( step function: 1. (1. is usually called the magnitude response or amplitude response.t/ D h. f / D F[x.t − / x.t/ D ( sign function: sgn. jH.− / d− D h.1.t/ D x Ł h. The magnitude of the frequency response. The output at a certain instant t 2 < is given by the convolution integral Z 1 Z 1 y.
t/ x. f / D X .2³ f 0 tC'/ ] 1 j' f 0 / C e j' X .t/ D Z C1 1 C1 X Parseval’s theorem Poisson sum formula Ex D jX . X Hermitian. f / D X Ł . f C f 0 /] [e X . f/ f /. f / C X Ł . f / [X .t/ x.t/ x. X Ł.1 Some general properties of the Fourier transform.t/ C b y. X. t/ x. f / C 2 j2³ f X . f / x.t/ sin.t/ cos. − /]. f /] 2j Â Ã f 1 X jaj a e j2³ f t0 X .t/j2 dt D . f C f 0 /] [e j' X . f /] 2 1 [X .− / d− D 1 Ł x.t t0 / 1 [X .− / Ł y. f f 0 /] [e X . f / X .t/ x. f / X. X.− / Ł y Ł . f / X Ł . X real and even X. f /] odd.t/ D x Ł . f / D X Ł. f /. f /] even. f / C b Y. f 2j 1 j' f 0 / C e j' X Ł .t/ D x Ł . f 2 1 f 0 / e j' X . f / a X .1.t/ X . f / X .− /].t/ C x Ł . f / D X imaginary and odd differentiation integration convolution correlation product real signal imaginary signal real and even signal real and odd signal d x. Im[X . t/ x Ł . jX .kTc / D X Tc `D 1 Tc kD 1 1 C1 X jx.t/ e j2³ f 0 t x. f f0 / x.Á/ Ł Y.t/ e j . a 6D 0 x.t/ D x Ł .at/. f / 1 X .t/ 2j Fourier transform X.0/ Ž.t/ y. f /j2 even X.t/ D x Ł . f/ f/ f/ x.t/] D Im[x. Property Signal x. f / Y. f / D X Ł. t/ x.t/ x. Re[X .t/ x Ł .t/ D x. f / X.3. f 2 j2³ f X .2³ f 0 t C '/ Re[x.t/] D x.t/ 1 [x.2³ f 0 t C '/ x.t/ [x. f / D Z C1 X Ł.t/ dt Z t x. f /j2 d f D E X Â Ã 1 ` x.t/ Re[x.t/ 2 x.Á/]. Continuoustime linear systems 15 Table 1. X . f /.t/ linearity duality time inverse complex conjugate real part imaginary part scaling time shift frequency shift modulation a x. f / Y Ł.
81) 4/F 3/F 2/F 1/F 0 1/F 2/F 3/F 4/F t 1/F·rect(f/F) 1/F F/2 0 F/2 f Figure 1.s/jsD j2³ f sinc(tF) 1 (1. f / D H .Ft/] D rect F as illustrated in Figure 1.s/ to indicate the Laplace transform of h. H .s/ often used in practice is characterized by the ratio of two polynomials in s.s/ is also called the transfer function of the ﬁlter. Example of signal and Fourier transform pair.11. .t/ D The following relation holds: sin.16 Chapter 1. t 2 <: H .80) with s complex variable. f F (1.79) Further examples of signals and relative Fourier transforms are given in Table 1.s/ by H. f / is related to H . It is easy to observe that if the curve s D j2³ f in the splane belongs to the convergence region of the integral in (1. then H. Elements of signal theory sinc. We reserve the notation H .2.80).t/e 1 st dt (1.t/.11.³ t/ ³t Â Ã (1.s/ D Z C1 h. A class of functions H . each with a ﬁnite number of coefﬁcients.78) 1 F[sinc.
Discretetime linear system (ﬁlter). is shown in Figure 1. k 2 Z. f f 0 / Ž. a > 0 t e at 1. x(k) Tc y(k) Tc h Figure 1.t/ 1 e j2³ f 0 t cos. a > 0 Fourier transform X. where Z denotes the set of integers. f f 0 / C Ž. Discretetime linear systems 17 Table 1.k/ and y.t/ Ž. The impulse response of the system is denoted by fh.a C j2³ f /2 2a a 2 C . f / Ž. f T / T sinc2 . f / C 2 j2³ f 1 j³ f T sinc.2³ f /2 r ³ ³ Á exp ³ f 2 a a 1. k 2 Z. f / 1 Ž.t/ Â Ã t rect T Â Ã t sinc T Â Ã Â Ã t jtj 1 rect T 2T e at 1.2 Examples of Fourier transform signal pairs.12. where x. or more simply by h. f T / T rect. Signal x. a > 0 2 e at .4 Discretetime linear systems A discrete–time timeinvariant linear system.t/.k/g. f f0 / 1 [Ž.4. f T / 1 a C j2³ f 1 .2³ f 0 t/ 1.2³ f 0 t/ sin. f C f 0 /] 2 1 [Ž.k/ are respectively the input and output signals at the time instants kTc . a > 0 e ajtj .t/ sgn.1.12.t/. . f C f 0 /] 2j 1 1 Ž. with sampling period Tc .
k/ D [x.z/ D under the condition ja=zj < 1. given by H .k/ D C1 X nD 1 h.3 some further properties of the ztransform are summarized.83) we ﬁnd H . For discretetime linear systems. f / where all functions are periodic of period 1=Tc .z/ zDe j2³ f Tc (1.D/ D C1 1 h.88) Sometimes the D transform is used instead of the ztransform. in the frequency domain (1.82) becomes Y.k/ D 0. f /H.k/.k/z k (1. We say the system is causal (anticausal ) if h. We deﬁne as transfer function of the ﬁlter the ztransform6 of the impulse response h. the output is given by y.k/ D 0. f / the frequency response of the ﬁlter. f / D X . Elements of signal theory The relation between the input sequence fx.86) (1. f / D F[h.k/e j2³ f kTc D H .m/].k/ D Tc 1 C 2T 1 2Tc c H. We list some deﬁnitions that are valid for timeinvariant linear systems.1 A fundamental example of z–transform is that of the sequence: ( ak k½0 jaj < 1 h.83) We indicate with H.84) The inverse Fourier transform of the frequency response yields Z h.k/ D x Ł h.k/ D 0 k<0 Applying the transform formula (1. k < 0 (if h.4.85) We note the property that.k/g is given by the convolution operation: y. Example 1.b/ bk . where b is a complex constant. k > 0).k/ D bk .z/ D C1 X kD 1 h.k/] D C1 X kD 1 h.82) In short. we will use the notation y.k/D k .k/g and the output sequence fy. kD . and H .k n/x. f /e j2³ f kTc d f (1. deﬁned as H.n/ (1.k/ D H .18 Chapter 1. for x.m/ Ł h.87) 1 1 az 1 D z z a (1. In Table 1.z/ is replaced by P h. 6 (1. where D D z 1 .
k/ scaling convolution correlation real sequence a k x. 1. k D 0.89) Using the Poisson formula of Table 1. one demonstrates that the Fourier transform of the sequence fh k g is related to Q. m/]. f / D F[h k ] D H e j2³ f Tc D Tc `D 1 Tc Discrete Fourier transform (DFT) For a sequence with a ﬁnite number of samples. we obtain: Gm D N 1 X kD0 km gk W N WN D e 2³ j N (1.k/ Example 1.az/ linearity ax.4. and setting Gm D Evaluating G.z/ Â Ã 1 [x. : : : .m=.92) .k/ C by.N Tc //.k/ D x Ł .k/ delay x.z/ C bY . N of the Fourier transform becomes G. be a continuoustime signal with Fourier transform Q.t/. f / D N 1 X kD0 1.kTc / k2Z (1.k/ X .z/ z m X .z/ a X .k/ X .z Ł / Â Ã 1 X z Â Ã 1 XŁ Ł z X .91) 1.z/Y Ł Ł z X .z/Y . f / by Ã Â 1 Á 1 X 1 (1. Discretetime linear systems 19 Table 1.t/.k/ inverse time x. f /. m D 0. Y .3 Properties of the ztransform. : : : .z Ł / x.k/ z transform X . y.k/.1.2 Let q. We now consider the sequence obtained by sampling q.k/ [x. fgk g. that is h k D q. the expression gk e j2³ f kTc (1. f / at the points f D m=. 1. N G. Property Sequence x.N Tc /.90) Q f l H.m/].m/ Ł y.m/ Ł y Ł .4.k m/ complex conjugate x Ł .z/. f 2 <. t 2 <.1.z/ X Ł . k/ x Ł .z/ D X Ł .
97) observing (1. the expression of the inverse DFT (IDFT) coincides with that of the DFT. N 1.98) 7 8 The computational complexity of the FFT is often expressed as N log2 N . is called the DFT of fgk g. Elements of signal theory The sequence fGm g.N 1/2 . X 1 N 1 Gm W N km N mD0 k D 0.94) 1.N 1/. A square matrix A is unitary if A H A D I where I is the identity matrix. k D 0.N 1/ complex additions and N 2 complex multiplications. G N G D Fg 1] 1] (1. g N and the vector of transform coefﬁcients G T D [G0 . 1. We also observe that direct computation of (1. a matrix for which all elements are zero except the elements on the main diagonal that are all equal to one.1= N /F is a unitary matrix. n D 0. besides the factor 1=N .N 2 4 WN ::: W N 1/ F D 6 1 WN 6 6 : : : : :: : : : 6 : : : : : 4 : . : : : . . : : : . N The inverse of (1.7 The DFT operator The DFT operator can be expressed in matrix form as 2 1 1 1 ::: 1 6 .93) We note that.e. 1.8 The following property holds: if C is a right circulant square matrix whose rows are obtained by successive shift to the right of the ﬁrst row.96) D DFT[g] (1. 1. : : : . N 1. gT D [g0 . i.N 1/ 2 6 1 WN WN ::: WN 6 6 2.92) it is immediate to verify the following relation: (1. : : : . G1 . : : : .n D W N .92) is given by gk D 1. : : : .20 Chapter 1. g1 . the algorithm known as fast Fourier transform Ð (FFT) allows computation of the DFT by N log2 N complex additions and N log2 N N 2 complex multiplications. and . : : : . m D 0.N 1/ 1 WN WN : : : WN in with elements [F]i. 1. then FCF 1 is a diagonal matrix whose elements are given by the DFT of the ﬁrst row of C.N 1/ . The inverse operator (IDFT) is given by 1 Ł F N (1. however. N 3 7 7 7 7 7 7 7 7 5 (1.95) F 1 D p We note that F D FT . Introducing the vector formed by the samples of the sequence fgk g. 1. k D 0.92) requires N . provided W N 1 is substituted with W N . i. N 1 (1.
13) with L x > N : x.k/ D L 1 X i D0 h rep L .circ/ .k `L/ (1. : : : .13.99) Circular and linear convolution via DFT Let the two sequences x and h have a ﬁnite support of L x and N samples respectively (see Figure 1.105) with main period corresponding to k D 0. N .i/ xrepL . : : : . 1. k D 0. based on (1. xrep L .k/ D 0 and h. Timelimited signals: fx(k)g.k/ D and h rep L .104) Deﬁnition 1.102) h.k/ D h L x.4. Discretetime linear systems 21 Moreover.6 The circular convolution between x and h is a periodic sequence of period L deﬁned as y .k/ D C1 X `D 1 x. : : : . Lx 1.k/ D 0 k<0 C1 X `D 1 k<0 k > Lx 1 (1. L x(k) 0 Lx 1 k 0 1 N1 k 1. h(k) i/ (1. Figure 1. 1.k `L/ (1. and fh(k)g.95). k D 0.k 1.101) We deﬁne the periodic signals of period L.100) k>N 1 (1.103) where in order to avoid time aliasing it must be L ½ Lx ed L½N (1. we obtain gD 1 Ł F G N (1.1. 1.
i/x. and fYm g. 1. Y L 1 D diagfDFT[x]gH (1.14. respectively.k/ D y.k/ only for k D N 1.circ/ .82): y. Elements of signal theory .111) Indeed.97).108) with (1. the result of circular convolution coincides with fy. and we write y . : : : . : : : . : : : . N .circ/ . and taking the inverse transform of the result. We give below two relations between the circular convolution y .k/ k D 0. one obtains the desired linear convolution. 1. Relation 1. . We verify that the two convolutions y .110) require that both sequences be completed with zeros (zero padding) to get a length of L D L x C N 1 samples. fHm g. This is achieved only for k ½ N 1 and k Ä L 1.k/ D y . the Lpoint (1.k/g. and y . : : : . : : : . : : : . h. respectively. L 1 (1.22 Chapter 1. Y1 . if we indicate with fXm g.109) L ½ Lx C N then y.circ/ . 1.110) To compute the convolution between the two ﬁnitelength sequences x and h. L 1 (1. taking the Lpoint DFT of the two sequences.105).circ/ D Y0 .106) m D 0.circ/ . L . : : : . indicated by ž and Ž. m D 0. 9 The notation diagfvg denotes a diagonal matrix whose elements on the diagonal are equal to the components of the vector v. with reference to Figure 1.108) 2. L x C N to see that only if N 1 X i D0 h. L 1. N . By comparing (1. L 1 In vector notation (1.k/ D x Ł h.112) (1. it is easy 1 (1. We are often interested in the linear convolution between x and h given by (1. we obtain DFT of sequences x. output of the linear convolution. 1. (1.k i/ (1.circ/ and the linear convolution y.circ/ Ym D Xm Hm 1. (1.circ/ and y coincide only for the instants k D N 1.106) becomes9 i h . Then.circ/ Then. in both cases we use L D Lx with L > N .circ/ T Y .106).circ/ .107) where H is the column vector given by the Lpoint DFT of the sequence h. performing the product (1.k/ D whose support is k D 0.109) and (1. only for a delay k such that it is avoided the product between nonzero samples of the two periodic sequences h r ep L and xr ep L .
N 1/ samples of fy. : : : .1.k/ D y .113) k D N px .14. N 1. and fh(k)g. L x C N 2g. An alternative to (1. : : : .k/ Let us deﬁne z. k D 0. : : : .k/g for instants k D N 1. with support f N px . L x 1 . k D 0. 1. px/ x . Illustration of the circular convolution operation between fx(k)g. Discretetime linear systems 23 hrep L (i) xrep L (ki) 0 N1 L1 i k(L1) k i Figure 1.k/g. 2. px/ be the linear convolution between x .116) Convolution by the overlapsave method For a very long sequence x.k/ D (1.114) 1 (1. the application of (1. 1.k/g are zero. : : : . 1 x.115) then from (1. If this were not true it would be sufﬁcient to shift the input by . N .L x C k/ Let y .112) leads to the overlapsave method to determine the linear convolution between x and h.112) is to consider. px/ that is obtained by partially repeating x with a cyclic preﬁx of N px samples: ( x.106) the following relation between the corresponding L x –point DFTs is obtained: Zm D Xm Hm m D 0. L x elsewhere 1 (1. px/ . . Let us now subdivide the sequence fx. : : : . : : : .k/ D ( y . 1. instead of the ﬁnite sequence x. it is easy to prove the following relation y . A fast procedure to compute the linear convolution fy. px/ and h. If N px ½ N 1. 1. 1.114) and (1. L x 1 (1.4.k/ 0 k D 0. It is not restrictive to assume that the ﬁrst . : : : .circ/ .N 1/ samples of the sequence fy.k/ k D 0.k/g into blocks of L samples such that adjacent blocks are characterized by an overlapping of . L 1. : : : .N 1/ samples. L x k D 0. : : : . L 1 is the following:10 10 In this section the superscript 0 indicates a vector of L components.N 1/ samples. an extended sequence x . px/ . and neglect the ﬁrst . 1. Relation 2.
x.N 2/ C 1. : : : . h. 0.1/. k D 0.L 1/] (1.N N 1 terms vector matrix (1.119) (1.k/ D 0.2.126) The algorithm proceeds until the entire input sequence is processed.L 1/ .L 1/ 2. Loading z } { 1/.0/.N D [x. Elements of signal theory 1.118) in which we have assumed x. : : : . x.L 2.k/ The third loading contains x 0T D [x. 3. : : : .2.123) and the desired output samples will be y. : : : .N 2//. 2. Transform H0 D DFT[h0 ] X 0 D diagfDFT[x 0 ]g 3.N 2/ (1.24 Chapter 1..L 1/ 2. N 2.N (1. .L 1/ . 1/] L N zeros h x 0T 0T D [h.125) k D L . : : : . x.120) vector (1. h. The second loading contains x 0T D [x. Matrix product Y 0 D X 0 H0 4.N /. x.3.k/ k D 2.122) where the symbol ] denotes a component that is neglected. : : : . : : : . 1. ] . y. y.0/. : : : . : : : . Inverse transform h i z } { D [ ]. : : : . : : : . x.N 2//] (1.121) y 0T D DFT 1 Y 0T 1/.N 2/ (1.L 1/ . x. 0 ] 1/.N 2//] (1.124) and will yield the desired output samples y. y.1/.N 2//.117) (1.L 1/ .L 1/ 2.N /.
k/ D 8 p >X < > : nD1 k rn pn k½0 k<0 (1. 8n.4. (1.k n/ D q X nD0 bn x.n/ D bn .131) 0 where rn D H .127) becomes y.z/[1 We give now two deﬁnitions.1 zn z pn z 1 1 / (1. n D 0.z/ D q X nD0 bn z n (1. To get the impulse response coefﬁcients.k n/ C q X nD0 bn x.7 A causal system is stable (bounded inputbounded output stability) if jpn j < 1. assuming known the ztransform H .132) . pn z 1 ]jzDpn (1. In the case in which an D 0. Deﬁnition 1.129) generally deﬁnes an inﬁnite impulse response (IIR) ﬁlter. : : : . p.128) and the transfer function for such systems assumes the form q X Y . q. Equation (1.z/ D 1 rn pn z 1 H) h. If the system is causal.k/ D p X nD1 an y. Discretetime linear systems 25 IIR and FIR ﬁlters An important class of linear systems is identiﬁed by the input–output relation p X nD0 an y. n D 1.129) nD0 1C p X nD1 an z nD1 p Y nD1 . : : : .z/. (1.z/.1 / where fzn g and fpn g are.z/ bn z n b0 D n q Y . 1.z/ D X .3.k n/ (1. we can expand H .z/ D H .k n/ k½0 (1.127) where we can set a0 D 1 without loss of generality. respectively. If q < p and assuming that all poles are distinct.1.130) and we obtain a ﬁnite impulse response (FIR) ﬁlter with h. 2.z/ in partial fractions and apply the linear property of the ztransform (see Table 1.129) reduces to H . the zeros and poles of H . page 19). we obtain p X nD1 H .
1 zn z 1 / (1. a minimum (maximum) phase system concentrates all its energy on the ﬁrst (last) samples of the impulse response. The coefﬁcients of the impulse responses h 1 . which is below (above) the phase response of all other systems. In other words. 8n.15.z/ is minimum phase. is given in Figure 1.2/ 0 0 0:15e j1:66 h.z/ and impulse response h 1 .z/ D Hmin .8 The system is minimum phase (maximum phase) if jpn j < 1 and jzn j Ä 1 (jpn j > 1 and jzn j > 1).4. H1 . h 2 . we get a maximumphase system H2 . We now observe that the magnitude of Ł the frequency response does not change if 1=z n is replaced with z n in (1.3/ 0:4e 0 0:58e j0:51 j0:31 h. Among all systems having the same magnitude response jH.0/ h 1 (minimum phase) h 2 (maximum phase) h 3 (general case) 0:9e 0:7e j1:57 h. If we move all the zeros outside the unit circle. that is a transfer function with some zeros inside and others outside the unit circle. h.4/ 0:3e 0:4e j0:63 0:3e j0:63 j1:57 0:9e j1:57 j0:63 .k/g. The coefﬁcients are normalized so that the three impulse responses have equal energy.k/ D k X i D0 jh.z/ as: H1 .15b.133). argH.26 Chapter 1. the minimum (maximum) phase system presents a phase response.15a.4.e j2³ f Tc /j. After determining the zeros of the transfer function. the magnitude of the frequency responses being equal.e j2³ f Tc /.3 It is interesting to determine the phase of a system for a given impulse response. Extending our previous considerations also to IIR ﬁlters.1/ 0 0:4e j0:31 0:24e j2:34 h.e.4 Impulse responses of systems having the same magnitude of the frequency response.15a.z/ whose impulse response is shown in Figure 1.134) Comparing the partialenergy sequences for the three impulse responses of Figure 1. and h 3 are given in Table 1.z/ is a ratio of polynomials in z 1 with poles and zeros Table 1.k/ shown in Figure 1.z/ D b 0 4 Y nD1 . if h 1 is a causal minimumphase ﬁlter. We deﬁne the partial energy of a causal impulse response as E. H1 . one ﬁnds that the minimum (maximum) phase system yields the largest (smallest) fE. Let us consider the system with transfer function H1 .i/j2 (1.133) As shown in Figure 1. A general case.15c. i. Elements of signal theory Deﬁnition 1. we factorize H1 . Example 1.
e. In the case of a minimumphase FIR ﬁlter with impulse response h min . where K is a constant. : : : .4. . i.z/ is a ratio of polynomials in z with poles and zeros outside the unit circle.z/ D K Hmin z1Ł . q. is satisﬁed. Discretetime linear systems 27 Figure 1. Hmax . In this text we use the notation fh 2 . Impulse response magnitudes and zero locations for three systems having the same frequency response magnitude. q. 1.z/ D z q Hmin z1Ł is a causal maximumphase ﬁlter. is an anticausal maximumphase ﬁlter. Á Ł H2 .n/g D fh 1 . n D 0. the relation fh 2 . : : : .15. Moreover.n/g D BŁ fh Ł . 1 where B is the backward operator that orders the elements of a sequence from the last to the ﬁrst.n/g. Á Ł inside the unit circle. n D 0.q n/g. then Hmax . 1.1.n/.
is the set of values ¾ 2 < for which jx. f /.¾ /.16 is usually done. then H is Hermitian. 1. In particular. in which the ﬁlter is a lowpass ﬁlter (LPF) if the support jH. f / D HŁ . in which the time domain of the input is different from that of the output. the terminology is less standard. f /j is an even function. decimator and interpolator ﬁlters are introduced. . the classiﬁcation of Figure 1.f/j. Classiﬁcation of real valued analog ﬁlters on the basis of the support of jH. otherwise it is a passband ﬁlter (PBF).16.28 Chapter 1. Let us consider a ﬁlter with impulse response h and frequency response H. and jH. f /j includes the origin. Depending on the support of jH. We adopt the classiﬁcation of Figure 1. Elements of signal theory In Appendix 1.A multirate transformations for systems are described.5 Signal bandwidth Deﬁnition 1.9 The support of a signal x. together with their efﬁcient implementations. f /j.17. Figure 1.¾ /j 6D 0. H. If h assumes real values. If h assumes complex values. ¾ 2 <.
1. Signal bandwidth 29 Figure 1. or as max f jH. f /j 6D 0. a) First zero: B D minf f > 0 : H.5. Classiﬁcation of complex valued analog ﬁlters on the basis of support of jH. Let us consider an LPF having frequency response H. f 2 <.136) .10 In general. f /j. the set of positive frequencies such that jX . f 2 B. for a signal x.f/j. The ﬁlter gain H0 is usually deﬁned as H0 D jH. f /j is an even function.140) BD B 11 The signal bandwidth may be given different deﬁnitions. Deﬁnition 1. f /j 6D 0 is called passband or simply band B: B D f f ½ 0 : jX .17. we have jX . includes or not the origin. f /j. We note that B is equivalent to the support of X limited to positive frequencies. f /j 6D 0g (1. We give the following four deﬁnitions for the bandwidth B of h. Analogously. for a realvalued signal x. The bandwidth11 of x is given by the measure of B: Z df (1.0/j.135) As jX . f / D 0g (1. we will use the same denomination and we will say that x is a baseband (BB) or passband (PB) signal depending on whether the support of jX . f /. other deﬁnitions are as average gain of the ﬁlter in the passband B.
f /j within a period. bandwidth at A dB: ¦ ² A jH.t/. t 2 <. As discretetime signals are often obtained by sampling continuoustime signals. in general complexvalued. t 2 < be a continuoustime signal. The sampling theorem Let q.137) jH.30 Chapter 1. For discretetime ﬁlters. h k D q. we refer the reader to [1]. B is equivalent to the support of X .18 illustrates the various deﬁnitions for a particular jH.16. c) Based on energy. . we will state the following fundamental theorem. If 1=Tc < B0 the signal cannot be perfectly reconstructed from its samples. f /j.139) Figure 1. Elements of signal theory For example. whereas for a PBF B D f 2 f 1 . The samples of the signal q.t/. In the case of a complexvalued signal x.142) For the proof. f /j B D max f > 0 : D 10 20 H0 Typically A D 3. with the caution of considering the support of jH. and B is thus given by the measure of the entire support. f /j2 d f p 100 (1. bandwidth at p%: Z B Z 01 0 (1. B0 is often referred to as the minimum sampling frequency.90) between a signal and its samples. In the case of discretetime highpass ﬁlters (HPF). let’s say between 1=.kTc / (1. 40. taken with period Tc .138) Typically p D 90 or 99. with regard to the signals of Figure 1. f /j2 d f D jH. under the condition that the sampling frequency 1=Tc satisﬁes the relation 1 ½ B0 Tc (1. which is based on the relation (1. f /j2 d f H2 0 (1.t/. d) Equivalent noise bandwidth: Z 1 BD 0 jH.2Tc /.2Tc / and 1=. we have that for an LPF B D f 2 . f / has support within an interval B of ﬁnite measure B0 . originating the socalled aliasing phenomenon in the frequencydomain signal representation.141) univocally represent the signal q. for which H is periodic of period 1=Tc .2Tc /. b) Based on amplitude. the same deﬁnitions hold. whose Fourier transform Q. the passband will extend from a certain frequency f 1 to 1=. or 60.
t/.19.4 0. where it is employed as an interpolation ﬁlter having an ideal frequency response given by ( GI . The real signal bandwidth following the deﬁnitions of: 1) Bandwidth at ﬁrst zero: Bz D 0:652 Hz.6 1.8 1 f (Hz) 1. f / D 1 0 f 2B elsewhere (1. 4) Equivalent noise bandwidth: Breq D 0:5 Hz. B50 dB D 1:62 Hz. .18. Operation of (a) sampling and (b) interpolation.pD99/ D 1:723 Hz. care must be taken in the choice of B0 ½ 2B to avoid aliasing between the positive and negative frequency components of Q. the signal q. t 2 <.143) We note that for realvalued baseband signals B0 D 2B.5. B40 dB D 0:87 Hz. Figure 1.4 1.19. f /.6 0.2 0.8 2 Figure 1. can be reconstructed from its samples fh k g according to the scheme of Figure 1. Signal bandwidth 31 0 −10 B3dB Breq −20 −30 H(f) (dB) −40 −50 B z −60 B40dB −70 BE (p=90) −80 B50dB −90 BE (p=99) −100 0 0. 2) Amplitudebased bandwidth: B3 dB D 0:5 Hz.pD90/ D 1:362 Hz.2 1. BE. For passband signals. In turn. 3) Energybased bandwidth: BE.1.
144) where H0 and t0 are two nonnegative constants. and B is the passband of the ﬁlter input signal x. A ﬁlter of the type (1. f / D t0 −. Figure 1.145) or. f / D H0 X . linear phase 12 arg H. 12 For a complex number c.144) satisﬁes the Heaviside conditions for the absence of signal distortion and is characterized by 1. f /j D H0 2. also called envelope delay 1 d arg H. the signal at the input is reproduced at the output with a gain factor H0 and a delay t0 . as jX . Then the output is given by Y. f / D 2³ f t0 f 2B f 2B (1.146) In other words. f / e j2³ f t0 (1. constant magnitude jH.32 Chapter 1. We show in Figure 1. .149) We emphasize that it is sufﬁcient that the Heaviside conditions are veriﬁed within the support of X .147) 3. Elements of signal theory Heaviside conditions for the absence of signal distortion Let us consider a ﬁlter having frequency response H.144).t/ D H0 x. that satisﬁes the conditions stated by Heaviside. f2 ). y.148) f 2B (1.10 or Figure 1.12) given by H. f / D 2³ d f (1. page 441). Characteristics of a ﬁlter satisfying the conditions for the absence of signal distortion in the frequency interval (f1 .20 the frequency response of a PBF. arg c denotes the phase of c (see note 3. f / D H0 e j2³ f t0 f 2B (1. the ﬁlter frequency response may be arbitrary. constant group delay.t t0 / (1.20. in the time domain. with bandwidth B D f 2 f 1 . f /X . f /j D 0 outside the support. for a ﬁlter of the type (1. f / D H. f / (see Figure 1.
6. Passband signals 33 1.a/ be a complex PB ﬁlter. The ﬁlter output.1. The following two procedures can be adopted to obtain x . in which jH. Referring to Figure 1. it is sufﬁcient that h . f / D (1. that extends from f 1 to f 2 .21 and to the transformations illustrated in Figure 1. with passband.150) 0 f <0 In practice.a/ H . given x we extract its positive frequency components using an analytic ﬁlter or phase splitter.bb/ . that extends from f 2 to f 1 . It is now convenient to introduce a suitable frequency f 0 . is Figure 1.a/ is called the analytic signal or preenvelope of x. Illustration of transformations to obtain the baseband equivalent signal x. f /. which usually belongs to the passband .21.22.bb/ . PB ﬁlter.6 Passband signals Complex representation For a passband signal x it is convenient to introduce an equivalent representation in terms of a baseband signal x . having the following ideal frequency response ( 2 f >0 .a/ . in which H. f 1 . f /j ' 0.a/ . f / ' 2.21. called reference carrier frequency. .a/ .a/ . f 2 / of x. f / D 2 Ð 1. x .bb/ around the carrier frequency f0 using a phase splitter. equal to that of X . and stopband. Let x be a PB realvalued signal with Fourier transform as illustrated in Figure 1. The signal x . h .
f /1. f / D X .153) In other words. x.34 Chapter 1. f /1.t/ C x .t/ D Re[x .152) ( .bb/ . f /H.bb/ j2³ f 0 t F F ! X .bb/ . BB ﬁlter. One gets the same result using a frequency shift of x followed by a lowpass ﬁlter (see Figures 1. we have x . making use of the property X . f / D X .a/ . f0 t f / D X . f /. also called complex envelope of x around the carrier frequency f 0 .bb/ is given by the components of x at positive frequencies.158) .23: H. f / D H.a/ .152) it also follows x. f / D X .bb/ is the baseband equivalent of x. It is immediate to determine the relation between the frequency responses of the ﬁlters of Figure 1. f / C X .f/ D 2X .t/ e j2³ f 0 t (1. or.a/ .t/ e and in the frequency domain X .bb/ around the carrier frequency f0 using a phase splitter. scaled by 2 and frequency shifted by f 0 .154) From (1. frequency shifted by f 0 to obtain a BB signal. Elements of signal theory phase splitter x(t) h(a) x (a) (t) x(bb)(t) e j2 πf 0 t Figure 1. f / D X Ł .a/ .a/ . In fact.a/ .a/ .t/e j2³ as illustrated in Figure 1.t/ D h.156) x .a/ .25.22. Analytically. it follows X .a/ .bb/ .t/ D x . f /1. The signal x .t/ D Using (1.t/ D x Ł h . f / C X Ł .21 and Figure 1.155) Relation between x and x(bb) A simple analytical relation exists between a real signal x and its complex envelope. f C f 0 / 0 (1.23 and 1.a/Ł . f / ! X . f /1.24). f C f 0 / (1. x .t/] 2 ] (1. Transformations to obtain the baseband equivalent signal x.t/ D Re[x .151) (1. equivalently. f C f 0 / for f > for f < f0 f0 (1.a/ .bb/ . x .157) (1.t/ x .154) one can derive the relation between the impulse response of the analytic ﬁlter and the impulse response of the lowpass ﬁlter: h . f/ (1.
6.1. x (bb) (t) x (a) (t) Re[ .bb/ around the carrier frequency f0 using a lowpass ﬁlter.24. Passband signals 35 Figure 1. Transformations to obtain the baseband equivalent signal x.25. ] x(t) e j2π f0 t Figure 1. Illustration of transformations to obtain the baseband equivalent signal x.23. LPF x(t) h e j2 πf 0 t x(bb)(t) Figure 1. .bb/ around the carrier frequency f0 using a lowpass ﬁlter. Relation between a signal. its complex envelope and the analytic signal.
h is real and the scheme of Figure 1. Substituting (1.163) Magnitude and phase of H. f / (1.159) .bb/ x . respectively. in block diagrams a Hilbert ﬁlter is indicated as “ ³=2”.t/ cos 2³ f 0 t . Elements of signal theory Baseband components of a PB signal. f / are shown in Figure 1. 13 We note that the ideal Hilbert ﬁlter in Figure 1.t/] (1. We note that h .36 Chapter 1. given x.26. We introduce the notation (1.bb/ . cos(2 π f 0 t) x (bb) (t) I x(t) x (bb) (t) Q sin(2 π f 0 t) Figure 1.h/ .t/ D x I .2 on page 17): h .bb/ .160) and . In practice these ﬁlter speciﬁcations are imposed only on the passband of the input signal. Conversely.t/ D Re[x .bb/ .160) and (1.26.161) to get the baseband components.t/ D 1 ³t (1.28.164) .28 has an impulse response given by (see Table 1. called inphase and quadrature components of x.h/ .t/ C j x Q .bb/ x.bb/ x I .24 and the relations (1.t/] (1.161) are realvalued baseband signals.bb/ x Q .t/ D Im[x . f / D j sgn. one can use the scheme of Figure 1.159) in (1.t/ D x I .158) we obtain .t/ where .13 To simplify the notation. If the frequency response H. f / has Hermitiansymmetric characteristics with respect to the origin.bb/ .27b employs instead an ideal Hilbert ﬁlter with frequency response given by H. Relation between a signal and its baseband components.162) as illustrated in Figure 1.bb/ x Q .t/ sin 2³ f 0 t (1.h/ . The scheme of Figure 1.h/ phaseshifts by ³=2 the positivefrequency components of the input and by ³=2 the negativefrequency components.27a holds. f / D e ³ j 2 sgn.
we obtain the relation H.1.h/ . f / D 1 C jH. f / (1.t/ D d− (1.167) Consequently. the output of the Hilbert ﬁlter (also denoted as Hilbert transform of x) is Z C1 x.h/ .165) ³ − 1 t . Passband signals 37 Figure 1.a/ .− / 1 x .150) and of the Hilbert ﬁlter (1. Comparing the frequency responses of the analytic ﬁlter (1. Relations to derive the baseband signal components.27. if x is the input signal.6.163).
38 Chapter 1. also x.166) ³ t − 1 14 We recall that the design of a ﬁlter.t/ D x Ł h . noting that from (1. j sgn f /.161): . j sgn f / D 1.g.bb/ x. Elements of signal theory  H (h)(f) 1 0 f arg H (h) (f) π 2 0 π − 2 f Figure 1. the analytic signal.t/ C j x .t/ Consequently.h/ x . x . taking the Hilbert transform of a signal we get the initial signal with the sign changed.163) .bb/ x I . Magnitude and phase responses of the ideal Hilbert ﬁlter.h/ .t t D /.t/ D x.h/ .14 We note that in practical systems. the complex envelope.160) and (1.t/ sin 2³ f 0 t as illustrated in Figure 1.169) (1.h/ . letting x . in the block diagram of Figure 1.t/ the analytic signal can be expressed as x .27b.t/ D x.152)..27.t/ cos 2³ f 0 t . are implemented by Moreover. In other words. Then it results: Z C1 .t/ D x . and in particular of a Hilbert ﬁlter.t/ and the various sinusoidal waveforms must be delayed. transformations to obtain.t/ cos 2³ f 0 t C x .170) (1.− / 1 x.a/ . (1. .h/ . or the Hilbert transform of a given signal. Then. from (1.28.h/ . e.168) (1.t/ sin 2³ f 0 t (1. we are only able to produce an output with a delay t D .t/ D d− (1. Consequently.h/ .171) x Q . requires the introduction of a suitable delay.
it is usually more convenient to perform signal analysis in the frequency domain by the Fourier transform. f / D Ae j'0 Ž.t/ D Ae j'0 e j2³ f 0 t (1.Bt/ (1.179) (1.6.Bt/ cos.bb/ .bb/ .2³ f 0 t C '0 / Then X.a/ . f / D rect X B B and x . Then. Â Ã f A .176) (1.t/ D A sinc.a/ .bb/ .173) The analytic signal is given by: X .172) Ž.6.1.1 Let x.177) (1. f C f 0 / (1. using f 0 as reference carrier frequency. Passband signals 39 ﬁltering operations.2 Let x.178) Another analytical technique to get the expression of the signal after the various transformations is obtained by applying the following theorem. f 2 f0/ C A e 2 j'0 (1.bb/ .172). . Example 1.175) We note that we have chosen as reference carrier frequency of the complex envelope the same carrier frequency as in (1.6. In the following two examples we use frequencydomain techniques to obtain the complex envelope of a PB signal. f / D Ae j'0 Ž. x.t/ D A sinc.29. f / F 1 f0 / F 1 ! x .t/ D A cos.174) ! x . However.t/ be a sinusoidal signal. f and X . f / D A j'0 e Ž.2³ f 0 t/ with the Fourier transform given by Ä Â Ã Â Ã½ f f0 f C f0 A rect C rect X. f / D 2B B B as illustrated in Figure 1.t/ D Ae j'0 (1. Example 1.
t/ D a.a/ .t/ and x .t/ 2 c .157).29.t/ D a.bb/ .t/ (1.183) have disjoint support in the frequency domain and (1.h/ . Corollary 1.1 Let the product of two real signals be x. Under the hypothesis that f 0 ½ B.184) ∋ ∋ B 2 B f R 2 (1. then the analytic signal of x is related to that of c by: x .bb/ . while that of the second is equal to .181) we obtain x .185) (1.183) is given by the interval [ f 0 B.a/Ł .180) where a is a BB signal with Ba D [0.t/ Proof. C1/.183) 1 .a/ .t/c.t/ c.180) yields x.t/ D a.40 Chapter 1.t/ 2 2 (1. f 0 C B].a/Ł . Theorem 1.t/ 1 c. 1.t/ (1.t/ 1 c.a/ 1 .1 From (1.t/ D Substituting (1. If f 0 > B.t/ D a. the two terms in (1.a/ .t/ C a.h/ . B/ and c is a PB signal with Bc D [ f 0 . Frequency response of a PB signal and corresponding complex envelope.181) (1.t/c.t/ c.t/ C 2 c In the frequency domain the support of the ﬁrst term in (1.181) is immediately obtained.182) . valid for every real signal c.182) in (1.t/ D a. C1/. Elements of signal theory A 2B −f 0 − B B − f0 −f 0 + 2 2 A B X (f) 0 X (bb)(f) f0 − f0 f0 + − B 2 0 B 2 f R Figure 1. We consider the general relation (1.
t/j2 dt D E x .a/ .187) (1. (1.t/ c.t/ D x .h/ . odd x .h/ .2³ f 0 t C '0 / cos.t/e in (1.t/ c.t/ x .bb/ .a/ .t/ D Im[x . t/ (Real) Hilbert transform x . Finally.188) (1.184).t/ D a.h/ .t/e j .t/ x.186) is in the design of a Hilbert ﬁlter h .t/ D x.5 some properties of the Hilbert transformation (1.t/ D x. An interesting application of (1.h/ . t/ x.h/ . if f 0 > B.t/ sin.152).5 Some properties of the Hilbert transform.h/ . Passband signals 41 In fact.a/ .t/ D a.191) (1.189) where a is a BB signal with bandwidth B.t/ D a. t/.t/ cos. from (1.6.t/] (1. t/.t/ D x .t/ x . Property (Real) signal x.bb/ j2³ f 0 t (1.2³ f 0 t C '0 / (1.t/ D a.h/ 1 1 hx.168) that are easily obtained by using the Fourier transform and the properties of Table 1.2³ f 0 t/ Example 1.h/ .1.155) and (1.t/ D h. from (1. t/ x .2³ f 0 t C '0 / Z C1 Z C1 Ex D jx.h/ i D Z C1 1 x.t/ D a.t/ x .3 Let a modulated double sideband (DSB) signal be expressed as x.t/j2 dt D jx .1) cosinusoidal signal energy orthogonality a.185) is obtained by substituting (1.186) which substituted in (1.h/ .t/ sin.186) we get h .1.169) we get x .h/ .t/ x .h/ .h/ .2³ f 0 tC'0 / x .2³ f 0 t C '0 / x . even duality inverse time even signal odd signal product (see Theorem 1.190) (1.181).t/ sin. x .t/ dt D 0 .192) . Table 1.t/e j'0 We list in Table 1. Then.h/ .6.h/ .h/ starting from a lowpass ﬁlter h.t/ x. In fact.181) yields (1. x . from the above theorem we have the following relations: x . t/ x.
cos.− / Ł Re[x . Three transformations are given in Figure 1.30.158). Passband transformations and their baseband equivalent. Assuming h is the realvalued impulse response of an LPF and using (1. y.30.t/ D fh. We will prove the relation illustrated in Figure 1.2³ f 0 − C '1 //]g.30b. Elements of signal theory Baseband equivalent of a transformation Given a transformation involving also passband signals.bb/ .42 Chapter 1.t/ Figure 1. together with their baseband equivalent.− /e j2³ f 0 − . it is often useful to determine an equivalent relation between baseband complex representations of input and output signals. .
bb/Ł .− / Ł x .195) . . with reference to the analytic signal we deﬁne: 1.a/ .bb/ and quadrature component h Q that are related to H.t/ C h. Given a PB signal x. f / D 1 [H.197) (1.6. f C f 0 / I and H.a/ by (see (1.194) D 1 [H.a/Ł Consequently.t/ D jx . H f /] (1.− / Ł x .30c has inphase component h I . I 2 f /] f C f 0 /] (1.t/ D arg x .bb/ .a/ has symmetrical frequency speciﬁcations around f 0 .a/ has Hermitian symmetry around f 0 then .161)) H. that the ﬁlter h .a/Ł .bb/ .− / 2 ! # .bb/ eC j . f / 2j H. f C f 0 /] 1 [H.bb/ is simpliﬁed.bb/ . f C f 0 / C H. f / D Ha . Instantaneous phase 'x . f / D 0 Q In other words h . f / C H.a/ H. 2 and H.bb/ .t/ D h . Passband signals 43 " D Re ÄÂ D Re hŁx h.bb/ .bb/ e j'1 Ã 2 where the last equality follows because the term with frequency components around 2 f 0 is ﬁltered by the LPF. Envelope and instantaneous phase and frequency We will conclude this section with a few deﬁnitions. if H.bb/ . f / D Q 1 [H. I 2 In practice this condition is veriﬁed by ensuring that the ﬁlter h .bb/ .t/ is real and the realization of the ﬁlter 1 h .t/.t/ (1.t/ (1.196) .193) .a/ .160) and (1.bb/ . Envelope Mx .1.a/ . moreover.bb/ .− / e j'1 2 ½ .bb/ in Figure 1.2³ 2 f 0 − C'1 / .bb/Ł .t/j 2.bb/ We note. f C f 0 / D 2j .a/ .
t/ D jx .t/ C 2³ f 0 t f x .t/ D arg x .202) becomes x.t/ D A cos.a/ .157).207) A D jx . 1.201) Then. Elements of signal theory 3. For example if x.t/j A (1. a PB signal x can be written as x.206) (1.bb/ .t/ Then (1.t/] C f 0 2³ dt (1.152) the equivalent relations follow: Mx .t/ D 1 d 'x .bb/ .t/ D 'x .t/ D Re[x .t/ D Mx .a/ .t/] D Mx .t/ D 1 d [arg x .203) (1. Instantaneous frequency f x .t// (1.bb/ .bb/ .2³ f 0 t C '0 / it follows that Mx .t/ D [A C 1Mx .2³ f 0 t C '0 / D arg x .t/ cos.'x . from (1. Frequency deviation 1 f x . Phase deviation 1'x .a/ . from the polar representation x .bb/ .t/ D jx .205) 1.t/ D f 0 With reference to the above relations.202) Two simpliﬁed methods to get the envelope Mx .t// (1.t/j 2.t/] cos.58 on page 514.204) (1.t/ and from (1.7 Secondorder analysis of random processes We recall the functions related to the statistical description of random processes. three other deﬁnitions follow.bb/ .208) .2³ f 0 t C '0 C 1'x .t/ D f x . Envelope deviation 1Mx .t/ '0 (1.t/ 3.t/ D 2³ f 0 t C '0 f x .198) In terms of the complex envelope signal x .200) (1.t/ e j'x .t/ D A 'x .t/j 'x .t/ are given in Figure 6.t/ 2³ dt (1.199) (1. . especially those functions concerning secondorder analysis.44 Chapter 1.t/ 2³ dt (1.209) f0 D 1 d 1'x .t/ from the PB signal x.
t 2 <. rx . be two continuoustime random processes. Secondorder analysis of random processes 45 1. t 4.t/mŁ . t ž x and y are uncorrelated if cx y .t/ D rx y .t.t − /] (1. ž x is widesense stationary (WSS) if 1.t.t/ D E[jx.210) mx . based on Deﬁnition 1. Crosscovariance cx y .x.t/mŁ . it is sufﬁcient that the inner product of the signals is zero.− /. t mx .t/ D rx .t. in the random case the crosscorrelation must be zero for all the delays and not only for zero delay. while in the deterministic case.7.t//. t − / D E[.212) (1. mx .t/ and y. In fact. − . t − / D E[. . We indicate the delay or lag with − and the expectation operator with E. 2.215) mx .t −/ − //Ł ] (1.t x mx . 1.t. In this case we write x ? y. 8t.t.t −/ −/ mx .t//. 8t.1.t/j2 ] 3. Crosscorrelation rx y . Ł In particular.2.x.7.t −/ −/ m y .t. Statistical power Mx .214) − / D E[x. t Observation 1. we note that the two random variables v1 and v2 are orthogonal if condition E[v1 v2 ] D 0 is satisﬁed.t/x Ł .15 − / D 0. 15 We observe that the notion of orthogonality between two random processes is quite different from that of orthogonality between two deterministic signals.t −/ − //Ł ] (1. t 5.t/ D E[x.t. orthogonality is equivalent to uncorrelation.t − /] (1. Autocorrelation rx .t/.t y − / D 0.211) (1.t/y Ł . ž if at least one of the two random processes has zero mean. Mean value mx . − .213) − / D E[x. t − / D rx .t. t 6. 8t. Autocovariance cx .y.t/ D mx .t/] 2.1 Correlation Let x.x.4 ž x and y are orthogonal if rx y .t. 8t.
− /] D Z C1 1 rx .217) are obtained from the Wiener–Khintchine theorem [2].− /j. yx 5.7.216) and (1.− / D rŁ .217) one gets the statistical power Mx D rx . rx Ł .− /j2 .0/ ½ jrx .− /. f / d f (1. mx .− /. 8t. t − / D rx y . f / D F[rx . x 1.t/ D m y . rx . rx . rx . 4. − / D rŁ . Properties of the autocorrelation function 1. rx .0/ D Z C1 1 Px . its power spectral density (PSD) is deﬁned as the Fourier transform of the autocorrelation function Px . 2. 3.− /.− / is a function with Hermitian symmetry. 8t.217) In particular from (1.11 The passband B of a random process x is deﬁned with reference to its PSD function.46 Chapter 1. The pair of equations (1. m y . f /e j2³ f − d f (1.− / D Z C1 1 Px .t/. x 2.t/j2 ] D Mx is the statistical power.218) hence the name PSD for the function Px .0/ D E[jx.0/ D ¦x D Mx is the variance of x.− /. Deﬁnition 1. whereas cx . rx y .t. t 2 <. rx y . − / D rŁ .t/ D mx .0/ ½ jrx y .2 Power spectral density Given the WSS random process x.t/ are jointly widesense stationary if 1. . Elements of signal theory 2 ž rx .t/.t/ and y.− /e j2³ f − d− (1. jmx j2 ž x.216) The inverse transformation is given by the following formula: rx .0/r y . f /: it represents the distribution of the statistical power in the frequency domain.
t/ C x . f / D Px .219) yields the relation rx .− / C r. With this aim in mind we give the following theorem.219) of the PSD: x.220) where I identiﬁes a discrete set of frequencies f f i g.222) The most interesting consideration is that the following random process decomposition corresponds to the decomposition (1.7. can be uniquely decomposed into a component Px .2 .c/ .1. f / D Px . x .d/ is given by x .t/ where x . f / (1.t/ D x .225) where fxi g are orthogonal random variables (r. f / C Px .t/ D X i 2I xi e j2³ fi t (1.226) .d/ Px .d/ . Secondorder analysis of random processes 47 Spectral lines in the PSD In many applications it is important to detect the presence of sinusoidal components in a random process.d/ .c/ . Theorem 1.c/ where Px is an ordinary (piecewise linear) function and .d/ .− / D r.− / D x X i 2I (1.c/ and x . Px .d/ are orthogonal processes having PSD functions .220).224) Moreover.d/ Px . f fi / (1.219) X i 2I Mi Ž. f / D Px .c/ .223) (1.c/ The PSD of a WSS process. f / D (1.− / x x with r.s) having statistical power E[jxi j2 ] D Mi where Mi is deﬁned in (1. i 2I (1. so that .221) Mi e j2³ fi − (1. f / and Px .v.d/ .d/ Px .c/ . i 2 I. f / .d/ with no impulses and a discrete component consisting of impulses (spectral lines) Px .c/ . The inverse Fourier transform of (1.d/ .
Px .− / jmx j2 is absolutely integrable : − / become uncorrelated for − ! 1. Px is a constant. f / D jmx j2 Ž.227) (1. f / D Px . This follows from property 1 of the autocorrelation. However. 5. Crosspower spectral density One can extend the deﬁnition of PSD to two jointly WSS random processes: Px y . f /j2 Ä Px .t/ is real valued.− / D jmx j2 x (1.k/] Since rx y . f / is generally not an even function. 3.228) 2/ cx . Px Ł . then both rx .48 Chapter 1. f / (1. 0 Ä jPx y .− / D rx . Deﬁnition 1. f / is a nonnegative function.t/.− / (1. − / 6D rŁ y . f / is in general a complex function.− /.231) Moreover.d/ Hence Px . Elements of signal theory Observation 1. Px .− / D cx .t/ and x. f / D Px y .232) (1. Ł 4.− / D jmx j2 − !1 (1.233) In this case 2 Px . f /. f / is a realvalued function.229) .− / and Px .− / x and r.5 The spectral lines of the PSD identify the periodic components in the process. if the process x. P yx . Px . (1. t 2 <.13 (White random process) The zeromean random process x. f /. the following inequality holds: Deﬁnition 1. f / D F[rx y .c/ . and the process exhibits at most a spectral line at the origin.d/ .t For such processes. f / are even functions. one can prove that r.e.230) Properties of the PSD 1. f /. f / D ¦x i. is called white if 2 rx .− / D ¦x Ž. x Px y . 2. . The property 1) denotes that x. f /P y .12 A WSS random process is said to be asymptotically uncorrelated if the following two properties hold: 1/ lim rx .
and Px1 x2 . We consider the scheme of Figure 1. Px2 . assuming the PSDs of the various input processes are known. Substitute the expressions of the products in the previous equations using the rule Yi Y Ł ! P yi y j j Ł X` Xm (1. P y2 .31 in which the inputs x1 and x2 have the following PSDs: Px1 . f /. the procedure consists of three steps. and P y1 y2 .237) (1.242) (1.7. f /. f /. . To determine the PSDs P y1 . f /. f /.241) (1. Construct the different products Ł Ł Ł Ł Ł Ł Ł Y1 Y1 D jH1 j2 X1 X1 C jH2 j2 X2 X2 C H1 H2 X1 X2 C H1 H2 X1 X2 (1. Determine the frequency response of the various outputs in terms of the inputs. x1 (t) x2 (t) y1 (t) y2 (t) h1 h2 h3 Figure 1. 1.239) (1.234) (1. PSD of processes through ﬁltering.31.236) Ł Ł Y2 Y2 D jH2 H3 j2 X2 X2 Ł Ł Ł Ł Ł Ł Y1 Y2 D H1 H2 H3 X1 X2 C jH2 j2 H3 X2 X2 (1. 2.238) 3. In our speciﬁc case. f /.240) ! Px` xm Then Ł Ł Ł P y1 D jH1 j2 Px1 C jH2 j2 Px2 C H1 H2 Px1 x2 C H1 H2 Px1 x2 (1. Secondorder analysis of random processes 49 PSD of processes through linear transformations By an example we will show how to determine PSDs of processes in a linear system.235) (1.449). we have Y1 D H1 X1 C H2 X2 Y2 D H2 H3 X2 in which for simplicity we omit the argument f .1.243) P y2 D jH2 H3 j2 Px2 Ł Ł Ł P y1 y2 D H1 H2 H3 Px1 x2 C jH2 j2 H3 Px2 The proof of the above method is based on the relation (1.
f /j2 P yz . Given a discretetime WSS random process x. then. f / D Tc F[rx . f / is a periodic function of period 1=Tc . the PSD is obtained as Px .50 Chapter 1. the statistical power is given by Mx D rx .0/ D Z 1 2Tc 1 2Tc Px .245) (1. The inverse transformation yields: rx . however.3 PSD of discretetime random processes Let fx.247) We note a further property: Px . from (1.1 remain valid also for discretetime processes: the only difference is that the correlation is now deﬁned on discrete time and is called autocorrelation sequence (ACS). r yz .244) (1. f /e j2³ f nTc d f (1. f /G .248) In particular. f /Pz . f / D Px . It is. f / D Px . 1. f /H.32. f / Ł (1. P y . Elements of signal theory h x y g z Figure 1.32. f / D Px .n/ D Z 1 2Tc 1 2Tc Px .249) .n/] D Tc C1 X nD 1 rx .k/g and fy. Deﬁnitions and properties of Section 1. f / P y . by applying the above method the following relations are easily obtained: P yx . f / D 0.n/e j2³ f nTc (1. through the frequency response of the ﬁlter. and y ? z.k/g be two discretetime random processes. interesting to review the properties of PSDs.− / D 0. f /jH. f / d f (1.245) is of particular interest since it relates the spectral density of the output process of a ﬁlter to the spectral density of the input process.7. PSD of processes through ﬁltering With reference to Figure 1. Reference scheme of PSD computations.246) The relation (1. i.231). f /H.7. In the particular case in which y and z have disjoint passbands.e.
c/ Px .d/ .255) r.15 If the samples of the random process fx.d.256) Ž Â f ` Tc Ã (1.n/ D ¦x Žn (1.i.i. samples.252) D jmx j Ž ` Tc Ã (1.1 We calculate the PSD of an i. In particular for a discretetime WSS asymptotically uncorrelated random process.250) In this case the PSD is a constant: 2 Px .229) and the following are true .n/ D 2 n 6D 0 jmx j it follows that ( cx .d.253) We note that. f / D ¦x Tc (1.219) is limited to a period of the PSD. provided the decomposition (1. f / D ¦x Tc 2 ¦x 0 (1.n/ D Then 2 r. if the process has nonzero mean value. f / D Tc C1 X nD 1 .k/g are statistically independent and identically distributed we say that fx.7. sequence fx.14 (White random process) A discretetime random process fx.k/g is white if 2 rx .7. f / D jmx j2 C1 X `D 1 (1.k/g.257) .n/ D jmx j2 x .251) Deﬁnition 1. Secondorder analysis of random processes 51 Deﬁnition 1. the relation (1.254) nD0 n 6D 0 (1. the PSD exhibits lines at multiples of 1=Tc .n/ e Â f j2³ f nTc (1. From ( Mx nD0 rx .1.c/ 2 Px . Example 1. Spectral lines in the PSD Even for a discrete time random process the PSD can be decomposed into ordinary components and spectral lines.c/ . f / 2 C1 X `D 1 cx .d/ Px .k/g has i.d/ Px .n/ D ¦x Žn x .
.z/ is a rational function.k n/ D [h.n/ D rx y Ł h.z/ H Ł Â 1 zŁ Ã (1.n/ PSD Pyx .z/H . then Py . Consequently the poles (and zeros) of Ph .z/ D Px y .1=z Ł / Py .n/ z n D H .260) whose ztransform is given by Ph . we obtain the relations between ACS and PSD listed in Table 1.k/h Ł .259) Using Table 1.n/ D C1 X kD 1 h.6 Relations between ACS and PSD for discretetime processes through a linear ﬁlter.z/H Ł .262) Â 1 zŁ Ã (1. assuming these processes are individually as well as jointly WSS. Elements of signal theory PSD of processes through ﬁltering Given the system illustrated in Figure 1. if Ph .261) In case Ph . the PSD of x.z/ D Px .258) with (1.n/ r y .e j2³ f Tc / (1.12.n/ D rx Ł rh .3.n/ (1.e j2³ f Tc /j2 In the case of white noise input.z/H Ł .n/z n (1.z/ Px y . e j' =jaj.n/ rx y .52 Chapter 1.z/ by Px .z/H Ł (1.m/ Ł h Ł .6.247). we want to ﬁnd a relation between the PSDs of the input and output signals.z/ come in pairs of the type e j' jaj. from (1. given by P y . m/].z/ D Px .261) one deduces that. From the last relation in Table 1.z/ has a pole (zero) of the type e j' jaj.258) From the comparison of (1. m/].z/ D C1 X nD 1 rh .z/ D Px .k/ is related to Px . ACS r yx . f / D Tc Px . We introduce the ztransform of the correlation sequence: Px .n/ D [rx .1=z Ł / 16 In this text we use the same symbol to indicate the correlation between random processes and the correlation between deterministic functions. Let the deterministic autocorrelation of h be deﬁned as 16 rh .z/ D 2 ¦ x H .n/ D rx Ł h. f /jH .z/H .m/ Ł h Ł .263) Table 1.6 one gets the relation between the PSDs of input and output signals.z/H . it also has a corresponding pole (zero) of the type e j' =jaj.z/ D C1 X nD 1 rx . f / D Px .
1g.n/g having ztransform Py .z/. P y .264). Secondorder analysis of random processes 53 and 2 P y .e j2³ f Tc / d f log f 0 D Tc 1=Tc The logarithms in (1.6.3 (Spectral factorization for discretetime processes) Let the process y be given. fQ1 .269) log Py . it is worth mentioning the process synthesis.e j2³ f Tc /: Z 2 (1.k/g and its autocorrelation sequence frh . The factor f 0 in (1. .268) is monic.266) and (1.6. Moreover. Z j log Py . and associated with a causal sequence f1.e.z/ Q is causal. f / D Tc ¦x jH . In the case of real ﬁlters Â Ã 1 Ł D H . in (1. monic and minimum phase) is unique.267) z where Q F. i.1.266) 1=Tc where the integration is over an arbitrarily chosen interval 1=Tc .267) is the geometric mean of Py .z/ may have only a discrete set of zeros Q on the unit circle. Minimumphase spectral factorization In the previous section we introduced the relation between an impulse response fh.265) Among the various applications of (1.7. For rational Py .z/ is obtained by extracting the poles and zeros of Py .z 1 / H zŁ (1. Two methods are shown in Section 4.e j2³ f Tc /j d f < 1 (1.1=z Ł / is the QŁ . which deals with the generation of a random process having a preassigned PSD.267) f 0 F Ł .526) and the considerations relative to (1. minimum phase.z/ F Ł Ł (1. which satisﬁes the Paley–Wiener condition for discretetime systems.z/ D 1 C fQ1 z 1 C fQ2 z 2 C ÐÐÐ (1.267) (with the constraint that F. Theorem 1.z/ D f 0 F. f / has the same shape as the ﬁlter frequency response. with autocorrelation sequence fr y . The Paley–Wiener criterion implies that Py . Then the function Py . fQ2 .269) may have any common base. fQŁ . associated with poles and zeros ztransform of an anticausal sequence f 0 f: : : . and that the spectral factorization (1. : : : g.z/ that lie inside the unit circle (see Q (1.z/ can be factorized as follows: Â Ã 1 2 Q Q Py .264) In other words. the function f 0 F.e j2³ f Tc /j2 (1.z/ that lie outside the unit circle. f 2 1 of Py . In many practical applications it is interesting to determine the minimumphase impulse response for a given autocorrelation function: with this intent we state the following theorem [3].n/g in terms of the ztransform.z/.261)).
277) (1. / .t/ D x . / . We assume that x does not have DC components.271) yields Px .t/.a/Ł and rx .− / D rx . / . Elements of signal theory 1. / (1. WSS process. signal x . f / and H.275) where Px . f /j2 Px . f / D jH.278) (1. f / D Px .271) (1.e.C/ .274) have nonoverlapping passbands.C/ .C/ . f / Px .16 A WSS random process x is said to be PB (BB) if its PSD is of PB (BB) type. f / D Px .279) . f / D jH.bb/ have zero mean.a/ x . Then x .C/ . f /j2 Px .t/ with x . PSD of the quadrature components of a random process Let x be a real PB.a/ . i. f / D Px .152) and rx . f / C Px .a/ .270) For the same input x.273) f/ (1. being x . We ﬁnd that x.a/ and x . f/ (1. / .a/ is equal to 2x . and Px . / . it follows that x .54 Chapter 1. / . / .− /e j2³ f 0 − (1.a/ ? x . f / Moreover.C/ . / .bb/ is related to x . frequency components at f D 0. hence (1.C/ x .− / D 0 The complex envelope x .C/ and x .C/Ł . The analytic (1.t/ D x .C/Ł . The following relations are valid Px . hence f /. Our aim is to derive the power spectral density of the inphase and quadrature components of the process.C/ ? x . f / D 0 as x .7. using Property 5 of the PSD.− / D 4rx .C/ and x .a/ .C/ . / .C/ . f / D Px . f / D Px . f /1. f / D 4Px .C/ .bb/ . f / D 1.272) (1.t/ C x .a/ by (1.− / and Px .a/Ł . f / D 1. hence its mean is zero and consequently also x . the output of the two ﬁlters is respectively x . / . / .C/ . We introduce two ﬁlters having a nonoverlapping passband and ideal frequency response given by H.4 PSD of passband processes Deﬁnition 1.276) rx . f / (1.a/Ł D 2x . f /1.
h/ x .t/] D (1.h/ .bb/ .C/ .bb/Ł .bb/ .27b. f / 4j x Q I f /] Q (1.287) Px .bb/ .288) one .− /] 2 I (1.bb/ .289) follows from Property 4 of ACS. from (1.bb/ .bb/ x . from (1. Referring to the block diagram in Figure 1.bb/ is an even function.bb/ .bb/ . (1. I Px .− / Then rx . f / C Px .− / D r.283) we obtain the following relations: rx .bb/Ł .bb/ .t/ D Re[x .− /] 2 Q I Px . f / D Px . as Px . from .284) f /] (1.− / D rx .bb/ x .h/ .278) it follows that x .bb/ .291) (1. f / D Px .bb/ .− / D 1 Im[rx .t/ and x Q .282) and .275) can be written as Px .t/ 2j x .bb/ .281) x .− / D rx .− / sin 2³ f 0 − x I Q j sgn.t/ (1.7.bb/ x .bb/Ł .− / D I Q rx . f C f 0 / D 4Px .a/ .bb/ .bb/ .h/ x .bb/ . in any case the random variables gets x I I Q I Q . − / The second equality in (1. f / D 1 [Px .t/ C x . f / D Q I 1 [P .bb/ .bb/ ? x Q only if Px .280).− / D rx .1.− / cos 2³ f 0 − C r.bb/ x .bb/ x I . f / D one gets rx .bb/ .0/ D 0.288) (1.− / Q I rx .290) and rx .285) (1. f 4 Finally. 4 I rx . Using (1.289) we note that rx . f / D 1 [Px .t/ are always orthogonal since rx .bb/ x .h/ .bb/ .bb/ .289) rx .bb/ ? x .h/ .− / D rx . f f 0 /] (1.bb/ .− / x (1.bb/ .bb/ x .286) (1. Secondorder analysis of random processes 55 hence Px .bb/ x Q .280) f 0 / C Px . f / (1.bb/ .bb/ .bb/ . f / and Px . Moreover.t/] D (1. From (1.292) .− / D 1 Re[rx .bb/ .bb/ x .bb/ .t/ D Im[x .bb/ x I .bb/ . f / Px .t/ 2 x . f C f 0 / Moreover.− / D rx .− / is an odd function.bb/ .bb/ .
0/ I Q (1.bb/ x .bb/ . f / D N0 rect 2 x Â Ã f B (1. The converse is also true: if x . It is immediate to get Px .295) (1.298) Â f B f0 Ã (1. f / D 2N0 rect B Then Px .0/ 2 rx .t/ D Re[x .− / sin 2³ f 0 − x (1.h/ .− / cos 2³ f 0 − C rx .0/ D rx .bb/Ł .303) (1.t/ e j2³ f 0 t ] is WSS with PSD given by (1.a/ .bb/ is a WSS process and x . then its complex envelope is WSS.− / D I Q r.bb/ . f / D 2 B B depicted in Figure 1.a/ .bb/ .− / 6D 0 (1. Elements of signal theory and rx .bb/ .2 Let x be a WSS process with power spectral density Ä Â Ã Â Ã½ f f0 f C f0 N0 rect C rect Px . / .bb/ .bb/ .302) .0/ rx . being Px .0/ D rx . f / D I Q (1.294) (1.0/ D rx .0/ D rx .bb/ x .bb/ ? x .7.56 Chapter 1. f / D 2N0 rect and Â Ã f Px .301) .bb/ .bb/ .0/ Example 1.0/ D 1 rx .300) 1 P .296) (1.bb/ is WSS. and x .0/ D rx . with rx .bb/ . f / D Px .bb/ .33.C/ .281).bb/Ł . f / D 0. however.bb/ Moreover. we have seen that.h/ .bb/ . we ﬁnd that x I ? x Q .bb/ ? x .299) (1. then x.C/ .293) In terms of statistical power the following relations hold: rx .bb/ x . If x .bb/Ł .297) rx .0/ D 2rx . I Q Cyclostationary processes In short.0/ D 4rx . if x is a real passband WSS process.
Secondorder analysis of random processes 57 N0 2 .− /e j2³ f 0 − e x j4³ f 0 t (1.t.− /e j2³ f 0 − j4³ f 0 t j2³ f 0 − e C rŁ .f0 B . .f0 B . t − / D 1 [rx .1.7.bb/ x .− /e 4 x C rx .bb/ x . t − / dt rx . x is cyclostationary in mean value with period T D 1= f .bb/ .− /e j2³ f 0 − C rŁ .f0 + 2 2 2N 0 P x (f) 0 P x (a) (f) f0 ∋ ∋ ∋ ∋ f0  B 2 f0 + B f R 2 0 2N 0 P x(bb) (f) f0  B 2 f0 f0 + B f R 2  B 2 N0 B 2 P x(bb) (f) .302) we ﬁnd that the autocorrelation of x.− / D N T0 0 (1.305) 17 To be precise. while it is cyclostationary in 0 0 correlation with period T0 =2. Spectral representation of a PB process and its BB components.bb/Ł . P x(bb) (f) 0 I Q f R  B 2 0 B 2 f R Figure 1.t. observing (1.17 In this case it is convenient to introduce the average correlation Z T0 1 rx . x is a cyclostationary process of period T0 D 1= f 0 .bb/ .bb/Ł .33.t/ is a periodic function in t of period 1= f 0 : rx .304) ] In other words.
58 Chapter 1.− /.2³ f 0 tC'0 / ] 2 D 1 2 a.2³ f 0 t C '0 / 1 2 a .t/ real random BB WSS process with bandwidth Ba < f 0 and autocorrelation ra . 2³ /. observing (1.t/ is the Hilbert transform of a.307) F− denotes the Fourier transform with respect to the variable − .bb/ . f C f 0 /] (1.− / is not identical to zero. in [0. In our case.t//e j .− / D ra .4 Let x. Example 1. in this case x turns out to be WSS.7. f.t/ is cyclostationary with period 1= f 0 .189)) x. it is N Px .− / and bandwidth Ba .308) with a. f / D 1 [Px .t/ e j'0 . From (1.− /] D T0 where Px .v. f.bb/ x .312) We note that one ﬁnds the same result (1.192) it results in x .2³ f 0 t C '0 / (1.h/ .t/ D a.h/ .bb/Ł .− / D ra .t/ be a modulated single sideband (SSB) with an upper sideband. t/ D F− [rx .bb/ .h/ . t/ dt (1.311) Therefore x has a bandwidth equal to 2Ba and an average statistical power N Mx D 1 2 Ma (1.307) Z 0 T0 Px .306) In (1. f 4 f 0 / C Pa . Hence we have rx . Example 1.bb/ .a.t/.2³ f 0 t C '0 / (1.− / rx .t/ D a. f / D F[rx .t/ D Re[ 1 .t/ C ja . From (1.281).t.308) the average PSD of x is given by N Px .311) assuming that '0 is a uniform r. f 4 as in the stationary case (1.bb/ . .303) we ﬁnd that x. f f 0 /] (1.3 Let x. f / D 1 [Pa .313) where a . a real WSS random process with autocorrelation ra .310) Because ra . t − /] (1.t/ cos.t/ be a modulated DSB signal (see (1.− / e j2'0 (1.t/ cos. Elements of signal theory whose Fourier transform is the average power spectral density 1 N N Px . x.t/ sin.309) f 0 / C Px .7.
34.313) is then stationary with Px .309).t/ D x.t/ C wo .t/.7.t/ and additive white noise w.t/ D 1 .314) where a .t/.a.bb/ and x . having a frequency response Ã Â f H. therefore it is one half of a.t//e j'0 2 it results that x .t/ C ja .t/ from r.319) Figure 1.316) Example 1. Coherent DSB demodulator and basebandequivalent scheme. f / D H0 rect (1.a/ .bb/Ł . In this case x has bandwidth equal to Ba and statistical power given by Mx D 1 4 Ma (1.bb/ x .317) where the signal x is modulated DSB (1.t/ C w.t/ D xo .t/ and it has spectral support only for positive frequencies. f f 0 /] (1. . one can use the coherent demodulation scheme illustrated in Figure 1.1.t/ (1. Being x .315) (1.5 (DSB and SSB demodulators) Let the signal r.t/ C ja .34 (see Figure 1.t/ be the sum of a desired part x. f / D N0 =2. Secondorder analysis of random processes 59 We note that the modulating signal . f / D 1 [Pa .C/ .t// coincides with the analytic signal a .t/ (1.t/ is deﬁned in (1.− / D 0 The process (1.bb/ .h/ .7.271).a.h/ .C/ . To obtain the signal a. r.30b) where h is an ideal lowpass ﬁlter. f 4 f 0 / C Pa . given by the sum of the desired part xo and noise wo : ro .bb/Ł have nonoverlapping passbands and rx .C/ .318) 2Ba Let ro be the output signal of the demodulator.t/ with PSD equal to Pw .
w.322) (1.t/ D a. we have r .t/ it results xo . we consider the noise weq at the output of ﬁlter h. Being h Ł a.t/ D weq.321) Mxo M wo (1. from wo .bb/ . we ﬁnd Pweq .328) (1. f /j2 2N0 1. f C f 0 / (1.325) In the same baseband equivalent scheme.323) e j'1 a.bb/ .t/ and using (1.Á/ e j'0 ].'0 2 '1 / (1.320) Using the equivalent block scheme of Figure 1. 3o D in terms of the reference ratio 0D Mx .329) .t/ e j'0 C w.Á/ Ł D Hence we get Mxo D 2 H0 Ma cos2 . f / D D 1 4 jH.t/ (1.60 Chapter 1. f / D and M w0 D 2 H0 N0 2Ba 4 (1.324) '1 / H0 a. Then.t/ D Re[h.326) Â Ã 2 H0 f N0 rect 2 2Ba Ł Being now w WSS.I .34 and (1. f C f 0 /.t/ with Pw.bb/ is uncorrelated with w.bb/ . Elements of signal theory We evaluate now the ratio between the powers of the signals in (1.t/ D H0 a. f / D 2N0 1.bb/Ł and thus weq with weq .327) Ã Â 2 H0 f N0 rect 4 2Ba (1.285) it follows Pw0 .t/ cos.'0 4 1 2 (1.192).319).N0 =2/ 2Ba (1.
would have fallen within the passband of the desired signal.'0 '1 / (1.N0 =2/ 4Ba 2 (1.'0 2 . coherently demodulated.331) It is interesting to observe that.35. Being Â Ã f Ba =2 H.H0 =4/ '1 / N0 2Ba D 0 cos2 .332) In other words.t/ (see (1.336) Figure 1. f / D 2 rect (1. f / D rect Ba Ba Note that in this scheme we have assumed the phase of the receiver carrier equal to that of the transmitter. . the ratio between the power of the desired signal and the power of the noise in the passband of x is given by 3i D For '1 D '0 then 3o D 23i (1. at the demodulator input.330) For '1 D '0 (1. Secondorder analysis of random processes 61 In conclusion. after the mixer.35. The ideal frequency response of h P B is given by Ã Â Ã Â f f 0 Ba =2 f f 0 Ba =2 C rect (1. We will now analyze the case of a SSB signal x.7.t/ D 1 2 h . (a) Coherent SSB demodulator and (b) basebandequivalent scheme.bb/ Ł h.t/ PB (1. using (1. following the scheme of Figure 1. where h P B is a ﬁlter used to eliminate the noise that otherwise.bb/ .330) becomes 3o D 0 (1. we have 3o D 2 .335) PB Ba the ﬁlter of the basebandequivalent scheme is given by h eq .312). measuring the noise power in a passband equal to that of the desired signal.334) H P B .333) Mx 0 D .1. the DSB demodulator yields a gain of 2 in signaltonoise ratio. to avoid distortion of the desired signal.H0 =4/ Ma cos2 .313)).
on the other hand. which is. even though half of the bandwidth is required. typically. must be expressed as the sum of a desired component proportional to a.62 Chapter 1. As a matter of fact. the considered systems do not introduce any distortion since xo .t/.Á// e j'0 ].t/ is proportional to a.bb/ .321) can be written as 3o D 0 (1. Using the fact x .a.t/ C j a .Á/ C j a . In the previous example.t/ 2 2 D D H0 Re[a.bb/ Ł h eq . f /j2 2N0 1.N0 =2/ 2Ba Observation 1.I we have Ã Â H2 1 f (1. it results in Mx D 3o (1.338) H0 a.t/ 4 In the basebandequivalent scheme.t/.t/] 4 (1.H0 =8/ N0 2Ba which using (1. whereas the noise is analyzed via the PSD. The demodulated signal xo .343) We note that the SSB system yields the same performance (for '1 D '0 ) as a DSB system. the noise weq at the output of h eq has a PSD given by Pweq . . f / D [Pweq .342) 2 .341) 8 Then we obtain 2 . Finally.337) We now evaluate the desired component xo .h/ .340) Pwo . f / D jHeq .t/.339) f Ba =2 N0 2 H0 rect D 2 Ba Ł and using (1.316) and (1.Á/ Ł 1 e j'0 1 .6 We note that also for the simple examples considered in this section the desired signal is analyzed via the various transformations. Elements of signal theory with frequency response Heq . which is valid because weq ? weq .344) 3i D . f / D H0 rect Â f Ba =2 Ba Ã (1.t/ D H0 x . f C f 0 / Ã Â (1.H0 =16/ Ma 3o D (1. f / C Pweq .t/.h/ . f /] D 0 N0 rect 4 8 2Ba and H2 Mwo D 0 N0 2Ba (1.t/ D Re[h eq .285). it results in xo . we are typically interested only in the statistical power of the noise at the system output.t/ and an orthogonal component that represents the distortion. small and has the same effects as noise. 1 4 From the relation wo D weq.
R is a Toeplitz matrix. x.k/g. then R is said to be positive deﬁnite and all its principal minor determinants are positive. taking an arbitrary vector vT D [v0 .347) If v H Rv > 0.e. v N 1 ].k/v] D v H Rv D N 1N 1 XX i D0 jD0 viŁ rx . we introduce the random vector with N components xT . 2.N Properties 1.k/xT .k/] D 6 : : 4 : rx .N xŁ .346) (1.k/ 1/.1 Let fw. R is positive semideﬁnite and almost always positive deﬁnite.8 The autocorrelation matrix Deﬁnition Given the discretetime widesense stationary random process fx.348) (1. Eigenvalues We indicate by det[R] the determinant of a matrix R.i j/v j ½ 0 (1. x. : : : . The autocorrelation matrix 63 1. in particular R is nonsingular.k/.k/ D [x. : : : .349) and the corresponding column eigenvectors ui satisfy the equation Rui D ½i ui Example 1. Its autocorrelation matrix R assumes the form 3 2 2 ¦w 0 Ð Ð Ð 0 2 6 0 ¦w Ð Ð Ð 0 7 7 6 RD6 : : : (1. i D 1.k/v. Indeed. all elements along any diagonal are equal.k/g be a white noise process.350) : 7 4 : : :: : 5 : : : 2 0 0 Ð Ð Ð ¦w . N C 1/ Ð Ð Ð rx . For real random processes R is symmetric: RT D R.0/ N C 1/] 3 7 7 7 5 (1. i.8. 8v.8.k/xT . R is Hermitian: R H D R.k The N ð N autocorrelation matrix of 2 rx . of the characteristic equation of order N det[R ½I] D 0 (1. and letting y D xT .345) 1/ rx .0/ : : : Ð Ð Ð rx . N . N C 2/ :: : ÐÐÐ 2/ Ð Ð Ð rx . we have E[jyj2 ] D E[v H xŁ . : : : . The eigenvalues of R are the solutions ½i .0/ 6 rx . 3.1.1/ 6 R D E[xŁ . 1/ rx .k is given by rx .
64 Chapter 1.k/ D e j .N 1/! (1. N .357) for all combinations of fci g.v. e j .353) 1Äi ÄN (1. 2. If the eigenvalues are distinct.354) (1. 2³ /. Elements of signal theory from which it follows that 2 ½1 D ½2 D Ð Ð Ð D ½ N D ¦ w (1.7 Correspondence between eigenvalues and eigenvectors of four matrices.351) and ui can be any arbitrary vector Example 1.N 1/! e : : : e j .!kC'/ 2 6 6 6 RD6 6 4 ! D 2³ f Tc 3 7 7 7 7 7 5 (1. then the eigenvectors are linearly independent: N X i D1 ci ui 6D 0 (1. the eigenvectors form a basis in < N . R Eigenvalue Eigenvector ½i ui Rm m ½i ui R 1 I .356) Other properties 1. not all equal to zero.2 We deﬁne a complexvalued sinusoid as x. A possible solution is given by ½1 D N and the relative eigenvector is T u1 D [1. The matrix R is given by 1 e j! : : : e j . in [0. Therefore.8.355) ] (1.7.N 1/! j . e j! .1 ¼R ¼½i / ui ½i 1 ui . Table 1. From Rm u D ½m u we obtain the relations of Table 1. 2. : : : .352) with ' uniform r. i D 1. in this case.N 2/! 1 : : : 1 2/! ÐÐÐ One can see that the rank of R is 1 and it will therefore have only one eigenvalue.N j! ÐÐÐ e ÐÐÐ e :: : j . : : : .
360) is deﬁned as Rayleigh quotient. ( 1 iD j 2 H jjui jj D ui ui D (1. By left multiplying both sides of (1. it enjoys the following properties. whose columns are the eigenvectors of R. 2. from (1. using (1. U D [u1 . In fact.358) Eigenvalue analysis for Hermitian matrices As previously seen.360) (1. The eigenvalues of a Hermitian matrix are real. the autocorrelation matrix R is Hermitian. it follows uiH u j D 0.363) (1. u2 . The trace of a matrix R is deﬁned as the sum of the elements of the main diagonal. uiH Rui ½ 0. : : : . and we indicate it with tr[R]. The autocorrelation matrix 65 3.e. then the eigenvectors are orthogonal.1. u N ] (1.359) The ratio (1.349) by uiH .½ j and since ½ j ½i /uiH u j (1.364) 0 i 6D j then the matrix U. valid for Hermitian matrices: 1. i.8. 3.361) (1.349) one gets: uiH Ru j D ½ j uiH u j uiH Ru j D ½i uiH u j Subtracting the second equation from the ﬁrst: 0 D . one gets ½i D uiH Rui uiH ui D uiH Rui jjui jj2 (1. If the eigenvalues of R are distinct and their corresponding eigenvectors are normalized. It holds tr R D N X i D1 ½i (1.13). As R is positive semideﬁnite. from which ½i ½ 0.362) ½i 6D 0 by hypothesis. If the eigenvalues of R are distinct.365) . Consequently. it follows uiH Rui D ½i uiH ui from which.
248) and (1.374) Z D 1 2Tc 1 2Tc Px .n e j2³ f nTc N X mD1 u i. Observing that uiH Rui D N N XX nD1 mD1 Ł u i.371) In fact.372). f /g Ä ½i Ä maxfPx .367) D6 : : : : 7 6 : : : : : 5 4 : : : 0 0 Ð Ð Ð ½N we get U H RU D From (1.370) 4. The eigenvalues of a positive semideﬁnite autocorrelation matrix R and the PSD of x are related by the inequalities.I ¼ /U H D N X i D1 N X i D1 (1.m e j2³ f mTc df (1. f /j2 d f .368) we obtain the following important relations: R D U UH D and I ¼R D U. f / N X nD1 Ł u i.n is the nth element of the eigenvector ui .373) and using (1. Elements of signal theory is a unitary matrix. f / D N X nD1 u i.n m/u i. the preceding equation can be written as uiH Rui D Z 1 2Tc 1 2Tc Px . : : : .366) This property is an immediate consequence of the orthogonality of the eigenvectors fui g.369) . that is U 1 D UH (1.372) where u i. minfPx . let Ui . if we deﬁne the matrix as 2 3 ½1 0 Ð Ð Ð 0 6 0 ½ ÐÐÐ 0 7 2 6 7 7 (1. N (1.66 Chapter 1.n rx . Moreover.m (1.1 ¼½i /ui uiH (1. f / be the Fourier transform of the sequence represented by the elements of ui : Ui . f / jUi .368) ½i ui uiH (1. f /g f f i D 1.n e j2³ f nTc (1.
E[.9. f /g ½max Ä ½min min f fPx . with a Gaussian distribution can be generated from two r.2³ / N det C N ] 1 1 1 T 2 e 2 . If we indicate with ½min and ½max .Q are uncorrelated.2 Ð Let xT D [x1 . f / jUi .Q D 1 ¦xi 2 mxi. x N . Examples of random processes 67 Substituting the latter result in (1.379) .Q mxi. 2.Q . . : : : .I and.360) one ﬁnds Z ½i D 1 2Tc 1 2Tc Px . ¾ N ].R/ assumes the minimum value of 1 for a white process.x mx /.376) From (1. the minimum and maximum eigenvalue of R. N (1. 2 2 2 ¦xi. If the inphase component xi.x mx /T ] is the covariance matrix.I and the quadrature component xi. : : : . f /j2 d f 1 2Tc 1 2Tc Z (1.3 Let xT D [x1. xi 2 N mi . Example 1. f /j2 d f from which (1.I C j x N . f / exhibits large variations.1.s with uniform distribution (see Appendix 1.xi.9 Examples of random processes Before reviewing some important random processes.375) jUi .376) we observe that .ξ mx / C N .9.xi. Example 1.I C j x1.ξ mx / (1. x N ] be a real Gaussian random vector.I D ¦xi.R/ may assume large values in the case Px . in view of the latter point we can deﬁne the eigenvalue spread as: . The joint probability density function is px . moreover. mx D E[x] is the vector of mean values and C N D E[.Q /] D 0 i D 1. Moreover.9. respectively.378) (1.R/ D max f fPx .1 A r. Example 1. we recall the deﬁnition of Gaussian complexvalued random vector. ¦i2 . 1. : : : .Q ] be a complexvalued Gaussian random vector.ξ / D [. : : : .B for an illustration of the method).377) where ξ T D [¾1 .371) follows.v.I /. f /g (1.9.v.
ξ / D [³ N det C N ] 1 e . with each element xi .t N /] be a complexvalued Gaussian (vector) process.68 Chapter 1.6 Given N realvalued sinusoidal signals x. The joint probability density function in this case results in px .t2 /.5 Let x. Elements of signal theory then the joint probability density function is px .t1 /.v.9.385) Ai sin. x N .t/] Z 2³ 1 A sin. uniform in [0.x mx /. : : : .t/ D N X i D1 (1.t 2³ 0 A2 cos.− / D A sin.382) The vector x is called a circularly symmetric Gaussian random vector. The vector x is called a circularly symmetric Gaussian random process.4 Let xT D [x1 .2³ f t C a/ da D 2³ 0 D0 and the autocorrelation function is given by Z 2³ 1 rx .9.ξ mx / (1. : : : . for which we will use the notation ' 2 U[0.ξ mx / H C N 1 .t/ D E[x.t N /]. x2 . Example 1.386) .t1 /.381) (1.383) where C is the covariance matrix of [x1 .9. Example 1.ξ / D [³ N det C] 1 e ξHC 1ξ (1. The mean of x is mx .380) with the vector of mean values and the covariance matrix given by mx D E[x] D E[x I ] C j E[x Q ] C N D E[.2³ f i t C 'i / (1.t/ D A sin.ti / having real and imaginary components that are uncorrelated Gaussian r.x mx / ] H (1. 2³ /. 2³ /.2³ f − / D 2 Example 1.2³ f t C a/A sin[2³ f . x N .384) − / C a] da (1.v.s with zero mean and equal variance for all values of ti .2³ f t C'/ be a realvalued sinusoidal signal with ' r.
9.k/ C w.s in [0.9 We consider a signal obtained by pulseamplitude modulation (PAM). from Example 1. page 48.k/g are uncorrelated.387) (1. following a similar procedure to that used in Examples 1.9. Moreover.8 Let the discretetime random process y.5 and 1.− / D N X A2 i cos.k/ with variance ¦w .7 Given N complexvalued sinusoidal signals x.− / D N X i D1 jAi j2 e j2³ fi − (1.t/ D N X i D1 Ai e j .v.390) is not asymptotically uncorrelated.9.s in [0. the process (1.389) with f'i g statistically independent uniform r.9.12.7 and white noise w.392) .2³ fi tC'i / (1.k/ D x. Example 1. expressed as y. according to the Deﬁnition 1.k/ h T x .391) Example 1.t/ D C1 X kD 1 x.386) is not asymptotically uncorrelated.6. In this case r y .t kT / (1. we assume fx.388) We note that.5 it is able to obtain the mean value mx .k/ be given by the sum of the 2 random process x. we ﬁnd rx .2³ f i − / 2 i D1 N X i D1 mxi .k/ of Example 1.t/ D 0 (1.9.9. Example 1. Examples of random processes 69 with f'i g statistically independent uniform r.v.1.390) We note that the process (1. 2³ /.9.9. 2³ /.k/g and fw.n/ D N X i D1 2 jAi j2 e j2³ fi nTc C ¦w Žn (1.t/ D and the autocorrelation function rx .
t − / dt D rx . f /j2 .− r y .t − / dt D [h T x .397) (1. where h T x is a ﬁniteenergy pulse. Average power for a white noise input For a white noise input with power Mx .t . having power spectral density Px . and fx.t/j2 dt is the energy of h T x .t. In general y is a cyclostationary process of period T . Mean m y .70 Chapter 1.t/ is the output of the system shown in Figure 1.393) rh T x . Let rh T x .− / be the deterministic autocorrelation of the signal h T x : Z C1 h T x . t −/ D C1 X iD 1 C1 X kD 1 h T x .t/ dt D mx HT x . t/]. f / is a periodic function of period 1=T .− / D N T 0 T iD 1 and þ þ2 þ1 þ N P y .t T − mT / (1.t/ D mx 2. f /þ Px .− / D T T 1 with Fourier transform equal to jHT x .i C m/ T /h Ł x .398) we observe that the modulator of a PAM system may be regarded as an interpolator ﬁlter with frequency response HT x =T . from (1. The signal y. Correlation r y . Elements of signal theory x(k) T hTx y(t) Figure 1.t kT / (1. f / D F[r y . In fact we have 1. 3.396) iT/ (1.t. We note that Px . f / N þT þ (1.394) rx .t/h Ł x .397) the average statistical power of the output signal is given by N M y D Mx where E h D R C1 1 Eh T (1.− / (1. Modulator of a PAM system as interpolator ﬁlter.k/g is a discretetime (with T spaced samples) WSS sequence.i/rh T x .0/ T 0 Z C1 1 T 1 X r y .395) If we introduce the average spectral analysis Z 1 T my D m y .36.36. .− /] D þ HT x .t/ Ł h Ł x .399) jh T x .i/ C1 X mD 1 h T x .
d. and observing the relation r yy Ł .406) can be obtained assuming the less stringent condition that x ? x Ł . then y is WSS with spectral density given by (1.i/ D rx I x Q . In particular we ﬁnd that y. this requires that the following two conditions are veriﬁed rx I .k/j2 ] 2 (1. Moments of y for a circularly symmetric i. i/ (1. t then rx x Ł .k/ x Q .k/] D E[x Q .k/ x Q .398).405) −/ D C1 X iD 1 rx x Ł .409) (1.t/. on the other hand.t/ ? y Ł .36.e.k/] D E[x I . 2 E[x 2 .e.406) (1.k/] E[jx.k/] 2 E[x Q .407) We note that the condition (1.k/ D Re[x.k/] C 2 j E[x I . Examples of random processes 71 4.i/ D E[x 2 .t.i/ C1 X mD 1 h T x .402) (1.1.t − mT / (1. E[y 2 .408) Observation 1.k/] D 0 (1. letting x I .k/g by using the scheme depicted in Figure 1.t.i/ D rx Q .k/] D 0 These two relations can be merged into one.i.t .378) and (1.k/ D Im[x. i.401) and E[x I . t −/ D 0 (1.t/ is circularly symmetric.7 It can be shown that if the ﬁlter h T x has a bandwidth smaller than 1=.404) that is y.k/]Ž.i/ and rx I x Q .t/] D 0 (1.i/ D 0 and r yy Ł . .i.k/ be a complexvalued random circularly symmetric sequence with zero mean (see (1.403) Filtering the i. input signal fx. i.2T / and x is a WSS sequence.400) (1. input Let x.i C m/T /h T x .9.d.k/] D x Q .379)).k/] we have 2 2 E[x I .
84)) and with rh .416) Consequently. We also assume that fx.q/ D mx 2.k/g is a WSS random sequence with mean mx and autocorrelation rx .n/ D Mx N N M y D Mx Eh Q0 (1.72 Chapter 1.415) i Q0/ (1. fyq g is a cyclostationary random sequence of period Q 0 with 1.10 Let us consider a PAM signal sampled with period TQ D T =Q 0 . In general.i Cm/Q 0 Ł hq n m Q0 (1. page 52. Let h p D h T x .n/ D N X 1 Q0 1 r y . where Q 0 is a positive integer number. f / P N þQ þ 0 If fx.417) (1.q TQ / from (1.411) describes the input–output relation of an interpolator ﬁlter (see (1.n/] D þ 1 H.n/ the deterministic autocorrelation (see (1.392) it follows C1 X yq D x.6.419) . Correlation r y .411) kD 1 If Q 0 6D 1.i/ hq .410) yq D y. We denote with H.413) By the average spectral analysis we obtain X H.260)) of the sequence fh p g.n Q0 i D 1 C1 X pD 1 (1.412) rx .414) hp (1.0/ 1 Q0 1 my D N m y . the average PSD is given by þ þ2 þ þ N y . Mean m y .q/ D mx Q 0 qD0 Q0 where H. (1. f / the Fourier transform (see (1.q. q Q 0 qD0 n/ D C1 1 X rx .k/ h q k Q 0 (1. p TQ / (1.0/ D and r y . We recall the statistical analysis given in Table 1.q.9.n/. f /þ Px . from (1.416) it results in rh .418) (1.i/ rh .n/ Q0 In particular the average power of the ﬁlter output signal is given by r y . f / D TQ F[r y . q n/ D C1 X iD 1 C1 X kD 1 C1 X mD 1 hq k Q0 (1.k/g is white noise with power Mx . Elements of signal theory Example 1.609)).
t/ D g M Ł g.37. f /G. We point out that the condition M y D Mx pD is satisﬁed if the energy of the ﬁlter impulse response is equal to the interpolation factor Q 0 . i. we consider a ﬁniteenergy signal pulse g in the presence of additive noise w having zero mean and power spectral density Pw .t/ The output is expressed as y.t0 / is maximum.426) Pw . g M : max gM jgu .t0 / is equal to Z C1 (1. f / evaluated in t D t0 .t/ C wu .1.424) The optimum ﬁlter has frequency response GM . . Reference scheme for the matched ﬁlter.425) where K is a constant.t0 /j2 ] G Ł.10.t/ D gu .0/ D 1 x(t)=g(t)+w(t) gM GM (f) = K G* (f) Pw (f) y(t) t0 y (t0 ) = gu (t0 ) + wu (t0 ) e j2π ft0 Figure 1. Proof.e. f / (1. The signal x.t/ wu .t/ (1. f / D K j2³ f t0 (1.t/ D g M Ł w.37.t/ D g. f /jG M . In other words.t/ C w.420) is ﬁltered with a ﬁlter having impulse response g M . 1. while the power of wu .10 Matched ﬁlter Referring to Figure 1.t0 / and the power of the noise component wu . Matched ﬁlter 73 P N where E h D C1 1 jh p j2 is the energy of fh p g.421) (1. The problem is to determine g M so that the ratio between the squared amplitude gu . We indicate with gu and wu respectively the desired signal and the noise component at the output: gu .422) We now suppose that y is observed at a given instant t0 .t0 / coincides with the inverse Fourier transform of G M . gu . f / e Pw . the best ﬁlter selects the frequency components of the desired input signal and weights them with weights that are inversely proportional to the noise level.t/ (1. f /j2 d f rwu .423) (1.t0 /j2 E[jwu .
f / and GŁ. f / it turns out jgu .428) G M . f / D K p Pw .0/ þZ þ þ þ C1 G M . f / Z C1 Pw . i. f /j2 d f (1. f /.0/ Z C1 þ þ 1 j2³ f t0 (1. f /þ w (1.74 Chapter 1.1) to the functions p (1. the ﬁlter has impulse response g M . f / e p Pw . f /G. f / j2³ f t0 þ G M . f / p dfþ e þ Pw . f /jG M .432) from which comes the name of matched ﬁlter (MF). From (1. f / Pw .425) becomes G M . f / e j2³ f t0 þ d f D þ P .433) j2³ f t0 (1.427) p D þZ þ þ þ C1 1 þ2 G.430) Therefore the maximum value is equal to the righthand side of (1. f / Pw .431) the solution (1. f / D Pw is a constant and the optimum solution (1. Matched ﬁlter in the presence of white noise If w is white. f / e G M .431) where K is a constant.430) and is achieved for p G Ł. f / þ d f þ P .f/ þ w C1 þ þ 1 þ2 þ þ pG. f /j2 d f 1 p where the integrand at the numerator was divided and multiplied by Pw . Applying the Schwarz inequality (see Section 1. f /e j2³ f t0 Z 1 C1 1 þ2 þ dfþ þ Pw . The desired signal pulse at the ﬁlter output has the frequency response Gu .t0 /j2 Ä rwu .434) . Elements of signal theory Then jgu .t0 t/ (1. f / 6D 0. f /j2 e j2³ f t0 (1. f / D K G Ł .425) follows immediately. f /jG M .t/ D K g Ł .t0 /j2 D rwu .e. f / D K jG. f /e Correspondingly. matched to the input signal pulse. then Pw . f / j2³ f t0 (1. f / Pw . Implicitly it is assumed that Pw .429) þ2 Z þ þ pG.
t0 ) + wu (t) t0 y(t0 ) gM (t)=Kg*(t0 t) Figure 1.439) Â t T =2 T Ã (1.437) In Figure 1. Example 1. the matched ﬁlter is proportional to g g M .10. using the relation E g D rg .0/ rwu . Note that in this case the matched ﬁlter has also limited duration and it is causal if t0 ½ tg . Z C1 rg .t/ D K rg .438) For t0 D T . Matched ﬁlter for an input pulse in the presence of white noise.a/g Ł .1. Matched ﬁlter 75 x(t)=g(t)+w(t) gM y(t)=Krg (t .1 (MF for a rectangular pulse) Let g.0/ Pw Pw jK j g (1.435) then.436) If E g is the energy of g. gu .a − / da 1 (1.441) .10.38.t t0 / (1. From the deﬁnition of the autocorrelation function of g.0/ the maximum of the functional (1.440) (1.39 the different pulse shapes are illustrated for a signal pulse g with limited duration tg .t/ and the output pulse in the absence of noise is equal to 8 þ þÃ Â þ þ > < K T 1 þ t T þ 0 < t < 2T þ T þ gu .− / D g. as depicted in Figure 1.− / D T 1 Ã − Á j− j rect T 2T (1.t0 /j2 g D D 2 r .t/ D wT .t/ D rect with Â rg .t/ D K wT .0/ Eg jgu .38.t/ D > :0 elsewhere (1.424) becomes jK j2 r2 .
that is the variance of the r. Elements of signal theory g(t) 0 t0 = 0 gM (t) tg t tg t0 = t g 0 gM (t) t 0 r g (t) tg t tg 0 tg t Figure 1.k/] D mx (1.v. mx Á vanishes for K ! 1.k/ K kD0 18 The limit is meant in the meansquare sense. 1. that is we consider the problem of estimating a statistical descriptor of a random process from the observation of a single realization.39. .442) K !1 K kD0 1 P K 1 x. We investigate now the possibility of moving from ensemble averaging to time averaging.11 Ergodic random processes The functions that have been introduced in the previous sections for the analysis of random processes give a valid statistical description of an ensemble of realizations of a random process. If in the limit it holds18 X 1 K 1 lim x.k/ D E[x.76 Chapter 1. Various pulse shapes related to a matched ﬁlter. Let x be a discretetime WSS random process having mean mx .
446) the following limit holds: 2 1 lim E 4 K !1 K Tc þ þ K 1 þ X x. the power spectral density. The property of ergodicity assumes a fundamental importance if we observe that from a single realization it is possible to obtain an estimate of the autocorrelation function and from this.445) and (1.k/ D A. ergodicity is. one could prove that under the hypothesis19 C1 X nD 1 jnj rx .443) x.k n/.n/ D 0 K (1. Analogously to deﬁnition (1.444).k K kD0 n/ D E[x.444) From (1. Let y.k/ mx þ 5 D 0 þ K kD0 þ K !1 or equivalently 1 lim K !1 K K 1 X nD .k n/] D rx . We note that the existence of the limit (1. one obtains the relations among the process itself.442).k/ equal to the sum of sinusoidal signals (see (1.k/x Ł .442) implies the condition 2þ þ2 3 þ1 K 1 þ X þ þ lim E 4þ (1. or x. where A is a random variable.K 1/ Ä 1 ½ jnj cx . Observing (1.k/x Ł . its autocorrelation function and power spectral density shown 19 We note that for random processes with nonzero mean and/or sinusoidal components this property is not veriﬁed.1. f / þ (1.442). Therefore it is usually recommended that the deterministic components of the process be removed before the spectral estimation is performed. we ﬁnd that the ergodicity in correlation of the process x is equivalent to the ergodicity in the mean of the process y.n/ < 1 þ2 3 þ j2³ f kTc þ 5 þ D Px . we say that x is ergodic in correlation if in the limits it holds: lim X 1 K 1 x.k/ D x.444) we see that for a random process to be ergodic in the mean.n/ (1. In other words.447) Then.11. We will not consider particular processes that are not ergodic such as x.k/ e þTc þ kD0 (1.k/x Ł . Alternatively. however.386)).444) for y translates into a condition on the statistical moments of the fourth order for x. In practice. some conditions on the secondorder statistics must be veriﬁed.445) K !1 Also for processes that are ergodic in correlation one could get a condition of ergodicity similar to that expressed by the limit (1. Therefore it is easy to deduce that the condition (1. the timeaverage of samples tends to the statistical mean as the number of samples increases. exploiting the ergodicity of a WSS random process. Ergodic random processes 77 then x is said to be ergodic in the mean. for a process for which the above limit is true. difﬁcult to prove for nonGaussian random processes. . we will assume all stationary processes to be ergodic.
k/ D jx.k/] (1. where Q XTd .k/g.k/x Ł . If we let Q X K Tc . we wish to estimate the mean value of a related process fy.447). while for the estimation of the correlation of x with lag n we set y.k/ K kD0 (1.k n/.40.442) an estimate of the mean value of y is given by the expression my D O X 1 K 1 y. from (1.449) holds also for continuoustime ergodic random processes. makes use of a statistical ensemble of the Fourier transform of process x. in Figure 1. We note how the direct computation of the PSD.1 Mean value estimators Given the random process fx. Relation between ergodic processes and their statistical description. f /j2 ] Td (1.447) becomes Px . Elements of signal theory Figure 1.k/g: for example. Based on a realization of fy.448) where w K is the rectangular window of length K (see (1.449) The relation (1. while the indirect method via ACS makes use of a single realization. given by (1.78 Chapter 1.k/ D x. f / denotes the Fourier transform of the windowed realization of the process.11.k/ w K .k/g. with a rectangular window of duration Td . (1. to estimate the statistical power of x we set y. 1.474)) and Td D K Tc . f / D lim Td !1 Q E[jXTd .k/j2 .450) .40. f / D Tc F[x.
(b) Typical impulse responses: exponential ﬁlter with parameter a D 1 2 5 and rectangular window with K D 33.035 0.k/ D h Ł y.k/g. H.8 0.025 0.01 0.0/ D 1.2 (b) 1 0. Let K be the length of the impulse response with support from k D 0 to k D K 1.6 H(f) 0. Ergodic random processes 79 In fact.0/ D m y O (1.02 h(k) 0.41.5 (c) Figure 1.4 0. in general we can think of extracting the average component of fy. Therefore we assume m y D z.k/ O for k ½ K 1 (1.015 0.2 0 −0.15 0.452) 0. As illustrated in Figure 1.451).41a. Note that for a unit step input signal the transient part of the output signal will last K 1 time instants.45 0.1. the mean value is given by E[m y ] D m y H.3 0.450) attempts to determine the average component of the signal fy. (c) Corresponding frequency responses.4 0. i.35 0. . (1.2 0 0.k/g using an LPF ﬁlter h having unit gain. From (1.25 fT c 0.451) We now compute mean and variance of the estimate. (a) Time average as output of a narrow band lowpass ﬁlter.2 0.05 0.005 0 0 5 10 15 20 k 25 30 35 40 (a) 1.1 0.11.e.03 0. and suitable bandwidth B.
and neglecting a delay factor.80 Chapter 1. for a ﬁxed ". two commonly used ﬁlters are the rectangular window and the exponential ﬁlter. whose impulse responses are shown in Figure 1.n/ (1.n/j D ¦ y C1 X jc y . Using the expression in Table 1. the length K of the ﬁlter impulse response must be larger.459) " 2S (1. from (1.453) Assuming SD C1 X nD 1 2 jc y . to obtain estimates for those processes fy.455) assuming as ﬁlter length K that of the principal lobe of fh.k/ D K :0 k D 0.457) In other words. f / D rect 2B Â Ã jfj < 1 2Tc (1. Elements of signal theory as H.k/g.n/j <1 2 ¦y nD 1 (1.6 of the correlation of a ﬁlter output signal given the input. Because of their simple implementation. : : : . Introducing the criterion that for a good estimate it must be var[m y ] Ä " O with " − jm y j2 .0/. K elsewhere 1 (1.k/g that exhibit larger variance and/or larger correlation among samples. For an ideal lowpass ﬁlter f H. the variance of the estimate is given by 2 var[m y ] D ¦ y D O C1 X nD 1 rh .41.454) and being jrh . it results in E h D 2B and K ' 1=B.0/. the variance in (1.454) and (1.n/j Ä rh .0/ D 1. Rectangular window 8 < 1 h. n/c y .456) (1. from (1.453) is bounded by var[m y ] Ä E h S O where E h D rh .455) it follows BÄ and K ½ 2S " (1.460) . or equivalently the bandwidth B must be smaller.458) (1.459). 1.
k/ D z.k/ (1. Ergodic random processes 81 The frequency response is given by H.k 1/ C y. f /j.1. the ﬁlter time constant is given by K 1 D 2l (1.468) becomes z. f / D 1 1 a ae j2³ f Tc (1.1 a/ y.e. Moreover. i. The 3 dB ﬁlter bandwidth is equal to BD 1 a 1 2³ Tc for a > 0:9 (1.464) with jaj < 1.471) .y.1 0 a/a k k½0 elsewhere (1.k K K/ (1.³ f Tc / (1.k/ z. E h D . adopting as length of h the time constant of the ﬁlter.463) K 1 X nD0 1 y.k/ D z.465) Moreover. from (1.K Tc /.³ f K Tc / sin.k K n/ (1.470) 2 l 1/ C .1 a/=.466).466) where the approximation holds for a ' 1.462) Exponential ﬁlter ( h. The ﬁlter output is given by z.k We note that choosing a as aD1 the expression (1.k 1/ C 2 l .k/ y. the interval it takes for the amplitude of the impulse response to decrease of a factor e.468) (1.k/ D .11. B D 1=.461) For the rectangular window we have E h D 1=K and. The frequency response is given by H.1 C a/ and.k 1// (1.k/ D az. f / D 1 e K j2³ f K 1 Tc 2 Ð sin.467) The ﬁlter output has a simple expression given by the recursive equation z.k/ D that can be expressed as z. adopting as bandwidth the frequency of the ﬁrst zero of jH.469) whose computation requires only two additions and one shift of l bits. K 1D 1 1 ' ln 1=a 1 a (1.
: : : . Hann window 8 > > > < > > > : 0 B 0:50 C 0:50 cos @2³ 0 8 > > > < > > > : þ þ D 1þ þ þk þ þ 2 þ 2þ þ þ D 1 þ þ þ 11 C A k D 0.k/ D 1 0 1 (1. 2. D elsewhere 1 (1. 1.k/ D 1 0 k D 0. be a realization of a random process with K samples. the equations (1.463) and (1. D elsewhere 1 (1. Unbiased estimate rx .k/ window20 (1. 1. a general window can be deﬁned as h. K examine two estimates.474) where D denotes the length of the rectangular window expressed in number of samples. : : : .475) k D D 2 1 w. : : : .k/ D k D 0.2 Correlation estimators 1.k/g of length K .11.472) with fw. K 1 (1.k/ D D 1 > > > : 0 3.n/ D O 1 K n K 1 X kDn x. k D 0. 1. 1.k n/ n D 0.470) give an expression to update the estimates.472) is introduced to normalize the area of h to 1. 1. : : : . We Let fx. The factor A in (1. Rectangular window ( w.477) .k/g. Triangular or Bartlett window k D 0.478) 20 We deﬁne the continuoustime rectangular window with duration Td as Â Ã ( 1 0 < t < Td t Td =2 wTd . Raised cosine or Hamming window 8 0 D 11 > > k > < B 2 C 0:54 C 0:46 cos @2³ A w. : : : .82 Chapter 1.k/ D Aw. We note that.473) (1. 1. D elsewhere w.k/ D w D . 1. Elements of signal theory General window In addition to the two ﬁlters described above.476) 4. : : : . D elsewhere 1 (1.k/x Ł .t/ D rect D Td 0 elsewhere Commonly used discretetime windows are: 1. for random processes with slowly timevarying statistics.
n/] L rx . Biased estimate rx .11.479) If the process is Gaussian.k n/] D rx .n/] C jBIASj2 O (1. one can show that the variance of the estimate is approximately given by var[rx .k/x Ł .n/] D 1 L rx . the biased estimate of the ACS exhibits a meansquare error21 larger than the unbiased. denoted as BIAS : BIAS D E[rx .n/] D L var[rx . but approaches it as K increases.m/ C rx .K n/2 C1 X mD 1 [r2 .m C n/rx .n/ O rx .m x n/] (1.480) (1.487) 21 For example.n/j2 ] D var[rx .0/ ½ jrx .m K K mD 1 x n/] (1.n/ K K !1 (1. Ergodic random processes 83 The unbiased estimate has mean value equal to E[rx .k K kDn Â n/ D 1 Ã jnj rx . for the estimator (1. It should also be noted that the estimate does not necessarily yield sequences that satisfy the properties of autocorrelation functions: for example. Note that the variance of the estimate increases with the correlation lag n. the following property may not be veriﬁed: O rx .m/ C rx .484) For a Gaussian process.481) The above limit holds for n − K .482) The mean value of the biased estimate satisﬁes the following relations: Ã Â jnj E[rx .n/ (1. the mean of the biased estimate is not equal to the autocorrelation function.n/ O K (1.n/ ! rx .486) . the variance of the biased estimate is expressed as Ã Â C1 K jnj 2 1 X var[rx .n/] ' O [r2 . Note that the biased estimate differs from the autocorrelation function by one additive constant.k/x Ł .m C n/rx .n/ (1. especially for large values of n.1.483) Unlike the unbiased estimate.n/j O n 6D 0 (1.478) the meansquare error is deﬁned as E[jrx .n/] O !0 K !1 K .485) In general.n/] D O 1 K n K 1 X kDn E[x.n/] ' O from which it follows var[rx .n/ D L X 1 K 1 x.
491) D Tc W B Ł Px . consequently. E[PPER .k/j2 K kD0 Z 1 2Tc 1 2Tc 1.490) E[rx .K 1/ K 1 X nD .k/g is given by O Mx D X 1 K 1 jx. f / D Tc X .k/g. f / is computed using the samples of rx .11. K an estimate of the statistical power of fx. f /j2 K Tc (1. where X . as PPER . f /j2 d f 1 D K Tc using the properties of the Fourier transform (Parseval theorem). we review some spectral density estimation methods.489) as PPER . f / D Tc and. Moreover. f / is the Fourier transform of fx.n/ even for lags up to K 1.489) rx .n/ D K 0 jnj > K 1 and 1 WB . Based on (1.K 1/ 1 Q jX .3 Power spectral density estimators After examining ACS estimators. (1. f / is the Fourier transform of the Bartlett window ( jnj jnj Ä K 1 1 w B . Periodogram or instantaneous spectrum Q Let X .492) We note the periodogram estimate is affected by BIAS for ﬁnite K .84 Chapter 1. k D 0. .493) (1.³ f K Tc / sin.n/e K j2³ f nTc (1.n/]e L Â 1 j2³ f nTc D Tc Ã jnj rx .³ f Tc / ½2 (1.488).488) Q jX . f /] D Tc K 1 X nD . : : : . Elements of signal theory 1.K 1/ K 1 X nD . whose variance is very large. f / where W B . it also L exhibits a large variance. a PSD estimator called a periodogram is given by PPER .n/ e L j2³ f nTc (1. f / D We can write (1. f / D K Ä sin. f /.
497) where Mw D X 1 D 1 2 w . with the choice S D 0 yielding subsequences with no overlap and therefore with less correlation.D S// k D 0.s/ PPER .499) Mean and variance of the estimate are given by E[PWE .sC1/ . D 1 s D 0. The symbol dae denotes the function ceiling.f/ Ns sD0 PER (1.s/ þ þX .k/ D kD0 (1.s/ P .498) is the normalized energy of the window. compute the Fourier transform Q X . that is the smallest integer larger than or equal to a.447) for ﬁnite K .s/ . .500) w.s/ . In general.s 1/ and with the following one x . for each frequency. : : : . different subsequences of consecutive D samples are extracted.k C s. 0 Ä S Ä D=2.s/ be the sth subsequence. f /] D Tc [jW. : : : . f / D D 1 X kD0 x . As a last step. N s 1 (1. Subsequences may partially overlap. Let x . The number of subsequences Ns is22 ¹ ¼ K D (1.½/j2 Ł Px .s/ .496) þ þ2 1 þ Q . 1. f / þ DTc Mw (1.k/ D w. characterized by S samples in common with the preceding subsequence x . that is the largest integer smaller than or equal to a. f / D Tc and obtain . f / D D 1 X kD0 (1. 1. f / where W.k/e j2³ f kTc (1.11.k/e j2³ f kTc (1.1. average the periodograms: PWE . f / D X 1 Ns 1 .½/]. Given a sequence of K samples.495) For each s.501) 22 The symbol bac denotes the function ﬂoor.k/ x.494) C1 Ns D D S Let w be a window (see footnote 20 on page 82) of D samples: then x . Ergodic random processes 85 Welch periodogram This method is based on applying (1.
504) For a Gaussian process.493) is chosen.502). consider the Fourier w. f /E w D P .n/g. we will use the same symbol in . n D O transform PBT . 1/=2.503) are the same as those introduced in note 20: the only difference is that they are now centered around zero instead of .0/ D 1. has a strong effect on the performance of the estimate.n/rx . f / ½ 0. Then if the Bartlett window (1. f /] D Tc W. The choice of the window type in the frequency domain depends on the compromise between a narrow central lobe (to reduce smearing) and a fast decay of secondary lobes (to reduce leakage).505) Windowing and window closing The operation of windowing time samples in the periodogram.502)). frx .f/ K 3K x (1. f /] / 1 2 P . Blackman and Tukey correlogram For an unbiased estimate of the ACS. 23 The symbol ‘/’ indicates proportional. carried out via the function “rect”. From (1. Smearing yields a lower spectral resolution. that is the capability to distinguish two spectral lines that are close. If K is the number of samples of the realization sequence. if the Bartlett window is chosen. leakage can mask spectral components that are further apart and have different amplitudes. On the other hand. f / (1. : : : . In terms of the mean value of the estimate.D both cases. therefore the application of the Welch method requires many samples. In fact. we get23 var[PWE . f / Ł Px .n/ e O j2³ f nTc (1. 24 The windows used in (1.86 Chapter 1.503) where w is a window24 of length 2L C 1.f/ Ns x (1. D must be large enough so that the generic subsequence represents the process and also Ns must be large to obtain a reliable estimate (see (1. we ﬁnd E[PBT . L. the variance of the estimate is given by var[PBT . with w. any truncation of a sequence is equivalent to a windowing operation.502) Note that the partial overlap introduces correlation between subsequences. Elements of signal theory Assuming the process is Gaussian and the different subsequences are statistically independent. f /] D 1 2 2L 2 Px . and the autocorrelation sequence in the correlogram. In general. we see that the variance of the estimate is reduced by increasing the number of subsequences. f / D Tc L X nD L L . one ﬁnds that PBT . To simplify the notation. we require that L Ä K =5 to reduce the variance of the estimate.
'2 2 U[0.kTc / D 16 1 X h.nTc / is a white random process with zero mean and variance 2 ¦w D 5.508) ³ 1 with T D 4Tc and ² D 0:32.507) (1.1 Consider a realization of K D 10000 samples of the signal: y.11. f 2 D 1:75.1 h. The estimate is then repeated by increasing the number of samples per window. but also characterized by an increasing variance. where '1 .1 C ²/ kTc T T rect Ã # Â 8T C Tc kTc 2 kTc 4² T T 16 X 16 h. where the condition L Ä K =5 must be satisﬁed. An example has already been seen in the correlogram..Ž. For a given observation of K samples. w.kTc / D ²/ kTc T " Ã C 4² Â Ã kTc kTc Ã Â cos ³. In this way we get estimates with a higher resolution. 2 P y . thus decreasing the number of windows. A Dirac impulse is represented by an isosceles triangle having a base equal to twice the desired . f 4 f 2 / C Ž.11.kTc / (1.509) A2 C 2 .nTc /w. f / D ¦w Tc A2 jH.k Ah nD 16 n/Tc / C A1 cos. observing (1. A1 D 1=20. Another example is the Welch periodogram. and Ah D Moreover Â sin ³.1. A2 D 1=40.509) is shown in Figures 1. and therefore a large number of windows (subsequences) over which to average the estimate.2³ f 1 kTc C '1 / (1. f / is the Fourier transform of fh. f 1 D 1:5. f 2 4 Ah f 1 / C Ž. it is initially better to choose a small number of samples over which to perform the DFT. f C f 1 // (1. f C f 2 // where H.506) C A2 cos.264) and (1. The shape of the PSD in (1. Consequently. 2³ /. The procedure is terminated once it is found that the increase in variance is no longer compensated by an increase in the spectral resolution. Tc D 0:2. Ergodic random processes 87 The choice of the window length is based on the compromise between spectral resolution and the variance of the estimate. The aforementioned method is called window closing.2³ f 2 kTc C '2 /.kTc /g.44 as a solid line. Actually y is the sum of two sinusoidal signals and ﬁltered white noise through h.388). Example 1.Ž.42 to 1. f /j2 C 1 .
509). the estimates of Figure 1.496) and (1. f / D jW. Figure 1.4Fq /. f / is the Fourier transform of w. for example. in addition to the analytical PSD (1. frequency resolution Fq . Therefore. in particular we will emphasize the effect on the resolution of the type of window used and the number of samples for each window. Comparison between spectral estimates obtained with Welch periodogram method. Finally.499) are: D D 1000. Consequently.497). fe j2³ f 1 kTc g with fw. using the Hamming or the rectangular window.503) L D 500. in particular. f C f 1 /j2 (1. from (1.88 Chapter 1.43 were obtained using in (1. obtained using the previously described methods.42. Ns D 19 and 50% overlap between windows. Windowing a complex sinusoidal signal f 1 /. Parameters used in (1.43 shows how the Hamming window also improves the estimate carried out with the correlogram. f where W.k/g produces a signal having Fourier transform equal to W. Elements of signal theory Figure 1. In general. We observe that the use of the Hamming window yields an improvement of the estimate due to less leakage. a Dirac impulse. f / centered around f 1 .509). the periodogram of a real sinusoidal signal with amplitude A1 and frequency f 1 is Â Ã2 A1 Tc PPER . We state beforehand the following result.42 shows.510) DMw 2 Figure 1. of area A2 =4 will 1 have a height equal to A2 =. the estimate obtained by the Welch periodogram method using the Hamming or the rectangular windows. f f 1 / C W. Likewise Figure 1. thus maintaining the equivalence in statistical power 1 between different representations. in the frequency domain the spectral line of a sinusoidal signal becomes a signal with shape W. We now compare several spectral estimates. and the analytical PSD given by (1.44 shows how the resolution and .
by varying parameters D ed Ns .509). . Ergodic random processes 89 Figure 1. and the analytical PSD given by (1. Comparison between spectral estimates obtained with the correlogram using the Hamming or the rectangular window. using the Hamming window.1.44. Comparison of spectral estimates obtained with the Welch periodogram method. Figure 1.43.11.
also called observed sequence.k/ D C1 X nD0 h ARMA . .k n/ C q X nD0 bn w. ARMA model of a process x.512) w(k) Tc w(k1) Tc Tc w(kq) b0 b1 bq + x(k)  ap x(kp) Tc x(k2) a2 Tc x(k1) a1 Tc Figure 1. 25 In a simulation of the process.511) Rewriting (1. 1. Elements of signal theory the variance of the estimate obtained by the Welch periodogram vary with the parameters D and Ns . using the Hamming window.k n/ (1. Note that by increasing D. In other words.45.k/ D p X nD1 an x.k/.k n/ (1.n/w.129) we ﬁnd in general x. is the output of an IIR ﬁlter having as input white noise with variance 2 ¦w . the process x.90 Chapter 1.q) model Let us consider the realization of a random process x according to the autoregressive moving average model illustrated in Figure 1.45. and hence decreasing Ns . the ﬁrst samples x.12 Parametric models of random processes ARMA(p.511) should be ignored because they depend on the initial conditions.k/ generated by (1. and is given by the recursive equation25 x. both resolution and variance of the estimate increase. from (1.511) in terms of the input–output relation of the linear system.
z/ A. f / þ A.z/ D 1 A.e j2³ f Tc / (1. Parametric models of random processes 91 which indicates that the ﬁlter used to realize the ARMA model is causal. f / D Tc ¦w þ where A.1.519) .z/ D B.k n/ C w.518) HAR . The output process is described in this case by the recursive equation x. f / D Tc ¦w jB.k/ D N X nD1 2 where w is white noise with variance ¦w .z/ D B. AR(N) model The autoregressive model of order N is shown in Figure 1.46. The equations of the ARMA model therefore are reduced to HMA .k/ (1.z/ and 2 Px . assuming ai D 0 i D 1. f / of a process obtained by the MA model.47. From (1. The transfer function is given by an x. f / D B. f / þ Px .514) MA(q) model If we particularize the ARMA model. f / D A.515) or A.516) (1.z/ where (1.e j2³ f Tc / þ þ 2 þ B. : : : . the power spectral density of the process x is given by: ( þ2 þ B.517) If we represent the function Px .513) n assuming a0 D 1 Using (1.z/ D > an z : nD0 n HARMA . p (1.264).12. we get the moving average model of order q.z/ (1. as illustrated in Figure 1. we see that its behavior is generally characterized by wide “peaks” and narrow “valleys”.z/ D bn z > > < nD0 p > X > > A. 2. f /j2 (1.129) one ﬁnds that the ﬁlter transfer function is given by 8 q X > > B.z/ D 1.
521) For a causal ﬁlter. w(k) +  x(k) aN a2 Tc a1 Tc Tc Figure 1.47. .519) describes a ﬁlter having N poles.z/ D 1 C N X nD1 an z n (1.92 Chapter 1.46. the stability condition is jpi j < 1. with A. i. : : : . Elements of signal theory Figure 1. 2.1 1 /. N .z/ D . Therefore HAR .e.1 pN z 1/ (1. all poles must be inside the unit circle of the z plane.1 p1 z 1 p2 z 1 / Ð Ð Ð . AR model of a process x.520) We observe that (1. i D 1. Power spectral density of a MA process with q D 4.z/ can be expressed as HAR .
given by Px .522) Hence the function Px . Parametric models of random processes 93 In the case of the AR model.1.523) (1. Power spectral density of an AR process with N D 4.z/A Ł Â ÃD 2 ¦w Â Ã 1 A. from Table 1.6 the ztransform of the ACS of x is given by Px . reciprocal to the MA model. .z/ 1 A. f / of an AR process will have narrow “peaks” and wide “valleys” (see Figure 1.12. f / D A. f / D 2 Tc ¦w and 1 j'i e jpi j (1.z/ D Pw .524) jA. f /j2 (1.48.525) Typically.e j2³ f Tc / one obtains the power spectral density of x.z/ has poles of the type jpi je j'i Letting A.z/A Ł Ł z 1 zŁ (1.48). the function Px . Figure 1.
k n/ (1.z/. Every WSS random process y can be decomposed into: y. As stated by the spectral factorization theorem (see page 53) there exists a unique spectral factorization that yields a minimumphase A.z/ is called spectral factorization.k/ (1. is described by the recursive equation s.z/ only the poles of Px .k/ D s.522) can be chosen in 2 N different ways. the output of this latter ﬁlter would be white noise.n/w. If A. The process s. the white process w is also called innovation of the process x.z/.k n/ (1. called predictable process.1 jp1 je j'1 z 1 / Ð Ð Ð . Two examples are illustrated in Figure 1. we have the following decomposition: Px . The selection of the zeros of A.k/ D C1 X nD0 h.z/ is minimum phase.526) For a given Px .527) where s and x are uncorrelated processes.k/ C x.z/ is called whitening ﬁlter (WF). Elements of signal theory Spectral factorization of an AR(N) model Consider the AR process described by (1.94 Chapter 1.50. Whitening ﬁlter We observe an important property illustrated in Figure 1.z/ D 2 ¦w . in the sense that the new information associated with the sample x.529) . Suppose x is modeled as an AR process of order N and has PSD given by (1.49. Wold decomposition.z/ that lie inside the unit circle of the zplane. Observing (1.k/ D C1 X nD1 Þn s. MA and AR models The relations between the three parametric models are expressed through the following propositions. If x is input to a ﬁlter having transfer function A.522).z/.k/ is carried only by w. it is clear that the N zeros of A.522).1 jp N je j' N z 1 / Â Ã Â Ã 1 j'1 1 1 j' N 1 1 ÐÐÐ 1 e z e z jp1 j jp N j (1. In this case the ﬁlter A.528) while x is obtained as ﬁltered white noise: x.k/. Relation between ARMA.z/ in (1.523). which is obtained by associating with A.
Two examples of possible choices of the zeros (ð) of A.z/. .49. Parametric models of random processes 95 Figure 1. among the poles of Px .12.1.z/.
50.k n/.531). m/ C ¦w rŁ .n/ D 2 am rx .k/x Ł .n m/ n>0 (1.k/ is uncorrelated with all past values of x. for n ½ 0 one gets E[x. and observing that w. Whitening ﬁlter for an AR process of order N. Theorem 1. 1.518) by x Ł .96 Chapter 1.1 Autocorrelation of AR processes It is interesting to evaluate the autocorrelation function of a process x obtained by the AR model.k/x Ł . provided that the order is sufﬁciently high. x n/ . Elements of signal theory Figure 1.4 (Kolmogorov theorem) Any ARMA or MA process can be represented by an AR process of inﬁnite order.k n/ (1.k n/ D N X mD1 am x. or AR) can be adopted to approximate the spectrum of a process.530) Taking expectations.532) N X mD1 N X mD1 am rx .k n/] D N X mD1 am E[x.k m/x Ł . MA.533) nD0 n<0 rx .n 2 m/ C ¦w Žn (1.k n/ C w. we ﬁnd x.n/ D In particular we have 8 > > > > > > > > < > > > > > > > > : N X mD1 am rx .531) From (1.k/x Ł .k 2 n/] C ¦w Žn (1. Therefore any one of the three descriptions (ARMA.12. Multiplying both members of (1.k m/ x Ł . it follows rx .
8 ž From (1. for n D 1. In the hypothesis that the matrix R has an inverse. : : : .0/ Ð Ð Ð r.n/ D mD1 . N C 2/ 7 6 a2 7 7 6 76 6 (1.1.540) rx .533). The variance ¦w of white noise at the input can be obtained from (1.n/ D ci pin n>0 (1.n/ !0 n!1 (1.0/ C r H a Observation 1. but only on the correlation coefﬁcients rx .1 and 2.2).537) with obvious deﬁnition of the vectors.1/ r.537) and (1.537) one ﬁnds that a does not depend on rx . allow us to obtain the coefﬁ2 cients of an AR model for a process having autocorrelation function rx . m/ ¦w D rx .N / univocally determines the ACS of an AR.z/.n m/ (1. ž We note that the knowledge of rx . for i D 1.n/ D n D 1.539) mD1 D rx .533) for n D 0.n/ satisﬁes an equation analogous to the (1.0/. This implies that. : : : .1/ r. the set of equations (1. rx . for n > 0. which yields N X 2 am rx .2. r x .534) i D1 Assuming an AR process with j pi j < 1.536) 76 : 7 D 6 : 7 6 : : :: 6 : 7 7 : : 54 : 5 4 : : : ÐÐÐ 4 : 5 : r.539) can be numerically solved by the Levinson–Durbin or by Delsarte–Genin algorithms.0/.0/ r.541) rx . 2.n/ with r.n/ ²x .0/ ž Exploiting the fact that R is Toeplitz and Hermitian. 2. with a computational complexity proportional to N 2 (see Sections 2.535) Simplifying notation rx . N . for n > N .538) Equations (1.0/ r. called Yule–Walker equations.N / process.k/.N 1/ r. : : : . the solution for the coefﬁcients fai g is given by a D R 1r (1. r x can be written as N X rx . Parametric models of random processes 97 We observe that. from (1.538) and (1. we get: rx . with the exception of the component w. : : : . if f pi g are zeros of A.2.N / aN that is Ra D r (1. we get N X am rx .N 2/ Ð Ð Ð r.538).518). rx .n/.2/ 7 7 6 r. N C 1/ 6 6 r.533).0/ C (1. N (1.12. and observing (1.1/. 1/ Ð Ð Ð r. N . one gets a set of equations that in matrix notation are expressed as 3 2 3 2 a1 3 2 r.
538) yields the coefﬁcient vector a.n/ is estimated for jnj Ä N with one of the two methods of Section 1. f / D Tc C1 X nD 1 rx .51. f /.525). we deﬁne as spectral estimate PAR . Figure 1. (1.543) where rx .51. The correlation coefﬁcients were obtained by a biased estimate on 10000 samples. while O for jnj > N the recursive equation (1. Note that the continuous part of the spectrum is estimated only approximately.12. which implies an estimate of the ACS up to lag N is available. because it does not show the effects of ACS truncation. The AR model accurately estimates processes with a spectrum similar to that given in Figure 1. such as PBT .n/ e O j2³ f nTc (1.542) allows a better resolution than estimates obtained by other methods.2 Spectral estimation of an AR(N) process Assuming an AR. a spectral estimate for the process of Example 1. on the other hand. f / D 2 Tc ¦w jA.11.509). In fact the AR model yields PAR .98 Chapter 1. Comparison between the spectral estimate obtained by an AR(12) process model and the analytical PSD given by (1.1 on page 87 obtained by an AR(12) model is depicted in Figure 1. From (1. Elements of signal theory 1.N / model for a process x.542) Usually the estimate (1.2. . f /j2 (1.48.541) is used. the choice of a larger order N would result in an estimate with larger variance. For example.11.
1. From ( rx . AR(1).539) for N D 1 and N D 2.0/ C a1 rx .12.52.1/ .1/ .n/ D from which the spectral density is PAR. the correlation estimation method and a choice of a large N may result in an illconditioned matrix R. Figure 1. and hence the system would be unstable.n 1/ n>0 D rx .n/ D 2 ¦w a1 rx . 1/ 2 ¦w (1. f / D 2 Tc ¦w 1 ja1 j2 .546) The behavior of the spectral density of an AR(1) process is illustrated in Figure 1. . In particular we will focus on the Yule– Walker equations and the relation (1. Some useful relations We will illustrate some examples of AR models. In this case the solution may have poles outside the unit circle.544) we obtain rAR. Spectral density of an AR(1) process.z/ near the unit circle (see page 101). Parametric models of random processes 99 Also note that the presence of spectral lines in the original process leads to zeros of the polynomial A. a1 /jnj (1.52. In practice.545) j1 C a1 e j2³ f Tc j2 (1.
4³ f 0 Tc / C %4 The spectral density is thus given by PAR. where '0 D 2³ f 0 Tc . as illustrated in Figure 1. PAR. as % ! 1.548) #/ (1. f / has a peak that becomes more pronounced. f / D þ þ1 2 Tc ¦w þ2 þ f 0 /Tc þ þ1 j2³.547) a2 D % 2 Letting # D tan we ﬁnd s Ã2 Â Ł2 1 C %2 1 %2 ð 1C tan 1 .2/ .53. . be the two complex roots of A.550) We observe that.549) %e j2³. f C f 0 /Tc þ2 1 Ä 1 %2 tan 1 C %2 1 .2³ f 0 jnjTc rAR. f %e þ (1.100 Chapter 1.z/ D 1 C a1 z 1 C a2 z 2 .k/ tends to exhibit a sinusoidal behavior. Elements of signal theory AR(2).2 D %eš j'0 .2³ f 0 Tc / ½ (1.2/ . Let p1. Spectral density of an AR(2) process.n/ D ¦w 1 %2 cos2 .2³ f 0 Tc / (1. and rx .53.2/ . Figure 1. We consider a real process: ( a1 D 2% cos.2³ f 0 Tc / 2 2 % 1C% 2 1 %jnj cos.
p1 p2 C 1/ 1 # (1.k/ D A cos.2/ rx > a2 D > > r2 .1/ > x x > > 2 : ¦w D rx .2³ f 0 Tc '/ (1.0/. 2/ D x.2 D eš j2³ f 0 Tc (1. 2³ /.557) 2 cos.553) .2/. p2 p1 /.2³ f 0 Tc /z 1 Cz 2 (1.554) with ' 2 U[0. p2 1/ rx .1/ rx .0/rx . p1 p2 C 1/ 2 AR model of sinusoidal processes The general formulation of a sinusoidal process is: x. Consequently the representation of a sinusoidal process via the AR model is not possible.0/ : 1 C a2 In general.2³ f 0 Tc /x.1/rx .1/ and rx .0/ rx .551) Solving the previous set of equations with respect to rx . 1/ D 0.0/ D 1 C a2 > > > 1 a2 . p1 1/ pn p1 /. for n > 0. rx .1.1/rx . Parametric models of random processes 101 Solutions of the Yule–Walker equations are 8 rx .12.556) It is important to verify that these zeros belong to the unit circle of the z plane.2/ >a D > 1 > > r2 .2/ (1.2/ D 1 > x a2 C rx .0/ r2 .0/ pn .1/ C a2 rx .552) 1 C a2 > > ! > > > > a2 > r .z/ D 1 The zeros of A.1/ > x x > < 2 .k 1/ x. We note that the Kronecker impulses determine only the amplitude and phase of x.n/ D rx . p2 2 p2 . we get the homogeneous equation A.k/ D 2 cos.z/ are p1. we have " 2 p1 .1/ D rx .0/ C a1 rx .1 C a2 /2 a2 > 1 > > < a1 rx .0/ (1.554) satisﬁes the following difference equation for k ½ 0: x. We observe that the process described by (1. one obtains 8 2 ¦w > > rx .k 2/ C Žk A cos ' Žk 1 A cos.0/ r2 . In the zdomain.555) with x.2³ f 0 kTc C '/ (1. as the stability .
7. ž An ARMA process of order .2/ . Indepth studies on deterministic systems and signals are found in [3.n/ D A2 cos. 5.9. 14.4³ f 0 Tc / 2 ¦w 2 %!1. in particular [4.102 Chapter 1. Moreover the input (1.38. 15.555) is not white noise. j pi j < 1.9 We can observe the following facts about the order of an AR model approximating a sinusoidal process.2³ f 0 nTc / 5 0 Tc / (1. The two components are then estimated by different methods.558) This autocorrelation function can be approximated by the autocorrelation of an AR(2) 2 process for % ! 1 and ¦w ! 0. 1. the subject of spectral estimation is discussed in detail in [2. we can try to ﬁnd an approximation. on the other hand. Finally.388). is not satisﬁed.z/.559) %2 cos2 . Observing (1. for example by the scheme illustrated in Figure 3.390) one ﬁnds that an AR process of order N is required to model N complex sinusoids. 10.4³ f and impose the condition 2 A2 1 %2 D 2 %2 cos2 . 16].2³ f 0 nTc / 2 (1. In any case.549).13 Guide to the bibliography Many of the topics surveyed in this chapter are treated in general in several texts on digital communications. 12. 11]. 6]. ž From (1.¦w !0 lim 2 (1. one sees that an AR process of order 2N is required to model N real sinusoids. 2N / is required to model N real sinusoids plus white 2 2 2 noise having variance ¦b . for % ' 1 we ﬁnd 2 6 rAR.5. Elements of signal theory condition. . it results ¦w ! ¦b and B. Using the formula (1. 13].n/ ' 6 4 2 2 ¦w 2 1 %2 3 7 7 cos. In the hypothesis of uniform '.560) Observation 1. from Example 1.z/ ! A. from (1. For a statistical analysis of random processes we refer to [1. 9.2N . 8.513). A better estimate is obtained by separating the continuous part from the spectral lines. rx .
Spectral analysis and time series. Signal analysis. NY: Academic Press. N. [15] L. Cariolaro. [18] H. NJ: PrenticeHall. [13] A. Messerschmitt and E. F. Probability. Marple Jr. Papoulis. Englewood Cliffs. Oppenheim and R. A. New York. Erup. [17] L. Multirate digital signal processing. Proakis. A.1. [4] S. [9] A. Englewood Cliffs. Digital communications. 1998.. 1981. New York: John Wiley & Sons. G. and S. E. Priestley. [11] R. Digital communication. Harris. Englewood Cliffs. [5] D. W. Davenport and W. pp. Bibliography 103 Bibliography [1] A. Digital communication receivers. Englewood Cliffs. Discretetime signal processing. Papoulis. Moses. Lee. Kay. 1983. random variables and stochastic processes. Rabiner. NJ: PrenticeHall. 1996. Englewood Cliffs. . [14] S. 998– 1008. New York: McGrawHill. B. Stoica and R. Principles of digital transmission with wireless applications. New York: McGrawHill. 2nd ed. 1999. New York: McGrawHill. [2] M. V. Shiryayev. 41. Vaidyanathan.. Introduction to spectral analysis. The Fourier integral and its applications. Root. B. A. 1987. Multirate systems and ﬁlter banks. IEEE Trans. Englewood Cliffs. 1988. Fechtel. 1993.. [3] A. [8] A. [7] G. on Communications. NJ: PrenticeHall. 1984. NJ: PrenticeHall. 3rd ed. [16] P. M. W. New York: IEEE Press. 1987. Digital spectral analysis with applications. New York: McGrawHill. Meyr. Biglieri. 1997. 1962. Gardner. MA: Kluwer Academic Publishers. G. 3rd ed. 1984. June 1993. P. Probability. [10] P. An introduction to the theory of random signals and noise. NJ: PrenticeHall. “Interpolation in digital modems—Part II: implementation and performance”. 1989. New York: Springer–Verlang. M. L. Boston. and R. New York: Kluwer Academic Publishers. Moeneclaey. Modern spectral estimationtheory and applications. S. 1991. Turin: UTET. R. Crochiere and L. NJ: PrenticeHall. Benedetto and E. [6] J. [12] J. vol. La teoria uniﬁcata dei segnali. 1994. Papoulis. 1995. M. Schafer..
562) (1. between t1 and t2 . with impulse response h. 1. whereas that of the output signal is Tc0 .kTc0 If we assume that h has a ﬁnite support. t 2 <.A.kTc0 nTc /xn D xn 1 h.t/.561) can be written as yk D n2 X nDn 1 h.54. that is kTc0 or equivalently for n> then.kTc0 n 2 Tc / (1.568) xn T c h yk T’ c Figure 1. Elements of signal theory Appendix 1.104 Chapter 1. . letting n1 D ¼ kTc0 t2 Tc ¾ n< kTc0 t1 Tc ³ (1.54.1 Fundamentals We consider the discretetime linear transformation of Figure 1.kTc0 nTc /x. Discretetime linear transformation. 11].kTc0 n 1 Tc / C Ð Ð Ð C xn 2 h.567) (1.kTc0 / h.561) We will use the following simpliﬁed notation: xn D x.A Multirate systems The ﬁrst part of this appendix is a synthesis from [10.563) nTc / 6D 0 for (1. say.nTc / (1.564) kTc0 t2 Tc kTc0 t1 n2 D Tc (1. the sampling period of the input signal is Tc .nTc / yk D y.kTc0 / D C1 X nD 1 h. The input–output relation is given by the equation y.566) ¹ (1.565) nTc < t2 kTc0 nTc > t1 (1.
568) becomes yk D I2 X i DI1 h. t1 .570) it is clear that 1k represents the truncation error of kTc0 =Tc and that 0 Ä 1k < 1.570) ¹ kTc0 C Tc ³ ¼ 0¹ kTc kTc0 C Tc Tc kTc0 Tc ³ ¼ (1. Hence there are only L univocally determined sets of values of h that are used in the computation of fyk g.572) kTc0 Tc ¹ n (1. 1=L . : : : . ž the limits of the summation (1.561) that: ž the values of h that contribute to yk are equally spaced by Tc .k M/mod L L (1.573) From the deﬁnition (1.i C 1k /Tc /xbkTc0 =Tc c i (1. Multirate systems 105 One observes from (1. in particular. In the special case Tc0 M D Tc L with M and L integers. Introducing the change of variable ¼ iD and setting 1k D kTc0 Tc ¾ t1 I1 D Tc ¾ t2 I2 D Tc ¼ ¹ kTc0 Tc ³ ¾ t1 1k D Tc ³ ¾ t2 1k D Tc (1.568) are a complicated function of Tc . while ..1. if L D 1 only one set of coefﬁcients exists.571) (1. and t2 .575) We observe that 1k can assume L values f0.569) (1.A.L 1/=Lg for any value of k. 2=L . Tc0 . we get 1k D k ¼ ¹ M M k L L Â ¼ ¹ Ã 1 M D kM k L L L D 1 .574) (1. .
.55 represents a decimator or downsampler.z/. We will show that Y .580) where W M D e j M is deﬁned in (1.z M W M / M mD0 (1.z/ in terms of X . and the input–output relation is the usual convolution yk D C1 X iD 1 g0.A.55.92). with the output sequence related to the input sequence fxn g by yk D x k M (1.e j! / D 0 X !0 1 M 1 X ej M mD0 yk M 2³ m Á M (1. !0 D 2³ f =Fc .577) C1 X iD 1 gk. the output of a ﬁlter with impulse response h and with different input and output time domains can be expressed as yk D where gk.i xj k M k L i (1. that is for L D M D 1. . Summarizing. Elements of signal theory if M D 1 the sets are L. 1.106 Chapter 1.i D h.578) We will now analyze a few elementary multirate transformations. is an integer number. Equivalently.580) can be written as Y . Decimation or downsampling transformation by a factor M.i x k i (1. the decimation factor. We now obtain an expression for the ztransform of the output Y . in terms of the radian frequency 0 normalized by the sampling frequency.579) where M. (1.z/ D 2³ X 1 1 M 1 m X .2 Decimation Figure 1.581) xn Tc Fc = 1 Tc T’ =MTc c Fc F’c = M Figure 1.576) We note that the system is linear and periodically timevarying. we get 1k D 0. For Tc0 D Tc .i C 1k /Tc / (1.
581) is shown in Figure 1.e j2³ f Tc / Y.e j! / by a factor M. We observe that.e j2³ f M Tc / (1.582) for the signal of Figure 1. as we would expect from a discrete Fourier transform.z/ D C1 X kD 1 yk z k D C1 X kD 1 x Mk z k (1. The ztransform of fyk g can be written as Y .1.585) . 0 ž create M 1 replicas of the expanded version. Decimation by a factor M D 3: a) in the time domain.57.A. f / D X . A graphical interpretation of (1.56: ž expand X . Multirate systems 107 Figure 1. we get Ã X Â m 1 M 1 (1.583) (1. f / D M mD0 M Tc where X . ž sum all the replicas and divide the result by M. f / D Y . after summation. and frequencyshift them uniformly with increments of 2³ for each replica.56 is represented in Figure 1. It is also useful to give the expression of the output sequence in the frequency domain.e j! =M /.582) X f Y. Proof of (1. the result is periodic in !0 with period 2³ . obtaining X . Note that the only difference with respect to the previous representation is that all frequency responses are now functions of the frequency f .56.584) The relation (1.580). and b) in the normalized radian frequency domain.
586) can be expressed as 0 x k D ck x k (1.z 1=M / (1. .586) 0 so that yk D x Mk D x Mk . : : : otherwise (1. With this position we get Y . we note that (1.587) This relation is valid.z/.z/ D X C1 1 M 1 X x k W M km z M mD0 kD 1 k X 1 M 1 W km M mD0 M X C1 Ð 1 M 1 X m x k zW M M mD0 kD 1 (1. It only remains to express X 0 .57.zW M /: hence.580).590) D k (1. because x 0 is nonzero only at multiples of M. š2M. š2M.z/ in terms of X .588) where ck is deﬁned as: ( ck D 1 0 k D 0. šM. : : : otherwise (1. observing (1.589) Note that the (1. Effect of decimation in the frequency domain. We deﬁne the intermediate sequence ( xk 0 xk D 0 k D 0. Elements of signal theory Figure 1. to do this.587) we get (1. šM.108 Chapter 1.591) m The inner summation yields X .z/ D C1 X k0D 1 0 xk 0 M z k0 D C1 X kD 1 0 xk z k=M D X 0 .589) can be written as ck D Hence we obtain X 0 .
Multirate systems 109 1. !0 D 0 2³ f =Fc . We will show that the input–output relation in terms of the ztransforms Y .59: Y . : : : L (1.A. called images. with the input sequence fxn g related to the output sequence by 8 Â Ã > <x k k D 0.595) for the signal of Figure 1. Interpolation or upsampling transformation by a factor L.594) The graphical interpretation of (1. there are L 1 replicas of the compressed spectrum.58.592) yk D > :0 otherwise where L.59 is illustrated in Figure 1. then (1.597) (1.e j2³ f Tc / Tc Á Y. The creation of images implies that a lowpass signal does not remain lowpass after interpolation. xn Tc 1 Fc = Tc L T’ = c yk Tc L F’ =LF c c Figure 1. f / where X . š2L .e j! / is the compressed version by a factor L of X . in terms of radian frequency normalized by the sampling frequency.593) can be expressed as Y .594) is illustrated in Figure 1. we get Y. f / D X .60.e j! L / 0 0 (1.1.596) (1. šL .z/ D X .e j! / D X .A. f / D X .593) Equivalently.z L / (1.3 Interpolation Figure 1. moreover. We note that the only 0 effect of the interpolation is that the signal X must be regarded as periodic with period Fc rather than Fc . It is also useful to give the expression of the output sequence in the frequency domain. .z/ and X . the interpolation factor.595) The (1.z/ is given by Y . f / D Y e j2³ f L (1.e j! /.58 represents an interpolator or upsampler. is an integer number.
110 Chapter 1.A.593). The ﬁlter ensures that the signal vn is bandlimited. Interpolation by a factor L D 3: (a) in the time domain.60. ´ µ Ì½ ¼ ´ µ Ì ½ Ì ¾ Ì ¿ ¹ Ì½ ¼ Ì ½ Ì ¾ Ì ¿ ¹ Figure 1. a downsampler is preceded by a lowpass digital ﬁlter.61. Elements of signal theory ´ µ Ù ¹½ ½ ¼ ÜÒ Ù ¹ Ý Ò ¾ ÉÉ É ¼ ÉÉ É ´ ¼ ÀÀ ¾ ¹ Ù Ù ¹¿ ¹¾ ¹½ µ Ù Ù Ù ¼ ½ ¾ ¿ Ù Ù ¹ ÄÄ ÄÄ ÄÄ ÄÄ ÄÄ ÄÄ ÄÄ ÄÄ Ä¹ Ä ¾ ¼ ¾ ´ µ ¼ Ù ´ µ Figure 1.59.z/ D Observing (1. (b) in the normalized radian frequency domain. to form a decimator ﬁlter as illustrated in Figure 1. .592) we get C1 X kD 1 yk z k D C1 X nD 1 yn L z nL D C1 X nD 1 xn z nL D X . Proof of (1. to avoid aliasing in the downsampling process.4 Decimator ﬁlter In most applications. Y . Effect of interpolation in the frequency domain.598) 1.z L / (1.
600) h i xk M i D C1 X nD 1 hk M n xn (1.1.605) we obtain Y .e j M / (1. Multirate systems 111 ÜÒ Ì ¹ Ý ¹ ¨ À Ì ÅÌ ¼ À ¨ ÜÒ Ì ¹ ÚÒ Ì ¹ Å Ý Ì ¼ ¹ Figure 1. recalling that !0 D 2³ f M Tc . Y .A.z/ D X . From V . equivalently.e M mD0 j !0 2³ m !0 2³ m M /X .e / D j! 0 X 1 M 1 H .z/ it follows that Y .602) C1 X iD 1 C1 X iD 1 (1. Let h n D h.e j! / D 0 j!0 Á 1 X eM M j!0 j Ä ³ (1.62 for M D 4. . i (1. the speciﬁcations of h can be made less stringent. Decimator ﬁlter.z 1=M W M / M mD0 (1.nTc /.61.z 1=M W M /X .i D h i 8k.603) or.577) we get gk.599) h i xn i (1. Then we have yk D vk M and vn D The output can be expressed as yk D Using deﬁnition (1.601) Note that the overall system is not time invariant. The decimator ﬁlter transformations are illustrated in Figure 1.z/ D X 1 M 1 m m H .e j! / D If ( H .604) 1 0 j!j Ä ³ M otherwise (1.606) In this case h is a lowpass ﬁlter that avoids aliasing caused by sampling. unless the delay applied to the input is constrained to be a multiple of M.z/H . if x is bandlimited.
as illustrated in Figure 1.nTc0 /. Elements of signal theory  X (f)  H (f) 0 Fc /2 Fc f  V (f) 0 Fc /2 Fc f  Y (f) 0 Fc /2 Fc f 0 F’ /2 F’ c c Fc /2 Fc f Figure 1. the task of the digital ﬁlter is to suppress images created by upsampling [17].63.608) yk D C1 X rD 1 hk r L xr (1.609) .62.112 Chapter 1.A. 1. Frequency responses related to the transformations in a decimator ﬁlter for M D 4. : : : otherwise (1. Then we have the following input–output relations: yk D C1 X jD 1 hk jwj (1.5 Interpolator ﬁlter An interpolator ﬁlter is given by the cascade of an upsampler and a digital ﬁlter. Let h n D h.607) wk D Therefore 8 < : x 0 Â Ã k L k D 0. šL .
e j! L / 0 0 0 (1.1.k/mod L . 1. however.63.614) H .i xj k k L i (1. The interpolator ﬁlter transformations in the time and frequency domains are illustrated in Figure 1.610) We note that gk. in some applications.z/ D X . Interpolator ﬁlter.z/ D H .i D h i LC.i is periodic in k of period L.64 for L D 3. easier and more convenient to change the sampling frequency by discretetime transformations.A. . using the structure of Figure 1. it is necessary to change the sampling frequency by a rational factor L=M.65.e j! / D H .e j! /X . equivalently. then resampling it at the new frequency.z L / or. From (1.612) (1.A.6 Rate conversion Decimator and interpolator ﬁlters can be employed to vary the sampling frequency of a signal by an integer factor. Y .e j!0 /D X .z L / Y .e j! / 0 0 j!0 j Ä ³ L elsewhere (1.z/X . It is.e / D 0 elsewhere we ﬁnd ( Y .609) we get yk D C1 X iD 1 gk. however.613) where !0 D 2³ f T =L D !=L.z/ D H .611) (1. A possible procedure consists of ﬁrst converting a discretetime signal into a continuoustime signal by a digitaltoanalog converter (DAC). If ( ³ 1 j!0 j Ä j!0 L (1. for example.615) The relation between the input and output signal power for an interpolator ﬁlter is expressed by (1. Let i D bk=Lc r and gk. Multirate systems 113 ÜÒ Ì ¹ Ý ¹ Ì¨ À Ì Ä ¼ À ¨ ÜÒ Ì ¹ Ä Û Ì ¼ ¹ Ý Ì ¼ ¹ Figure 1.z/W . In the ztransform domain we ﬁnd W .419).
Elements of signal theory ÜÒ Ù Ù Ù ´ µ ½ ¼ ½ ¹ Ò Ï ¼ ´ µ ½ ¼ ¾ ¤¤ ´ ¤´ ¤ ¼ ¹ Ù Û Ù ÙÙ ÙÙ Ù ¼½¾¿ ¹ À ¼ ´ µ ¤ ¤¤ ¤ ´´ ¼ ¾ ¤ ¤ ¤¤ ¤ ¤ ¤¤ ´´ ¼ ¹ ¼ ¼ ¾ ¼ ¹ Ý ÙÙ Ù ÙÙ Ù ¼½¾¿ Ù ´ µ ¹ ¼ ¼ ¾ ´´ ¤¤ ¤ ¤ ¹ ¼ Figure 1.64. .114 Chapter 1. ÜÒ Ì ¹ Ý Ì ¼¼ ¹ Å¨ À À ¨ ÄÌ ÜÒ Ì ¹ Ä Ì ¼ ÛÐ ¹ Ì Ä ÚÐ Ì ¼ ¹ Å Ì ¼¼ Ý ¹ ÅÌ ¼ Figure 1. Time and frequency responses related to the transformations in an interpolator ﬁlter for L D 3. Sampling frequency conversion by a rational factor.65.
616) we have 00 (1. In the frequency domain we get X !00 2³l 1 M 1 00 V .e j! / D From (1. .e j M /X . the desired result is obtained by a response 0 H . f / for M (1.622) Example 1.e j! / D > :0 Y.619) (1. as depicted in Figure 1.67. This system can be thought of as the cascade of an interpolator and decimator ﬁlter. L elsewhere Ã Â 1 L j f j Ä min .A.618) V .. Multirate systems 115 Figure 1.621) or (1. L D 5) The inverse transformation of the above example is obtained by a transformation with M D 4 and L D 5.e j! L / X !00 2³l !00 L 2³l 1 M 1 H .e j! / D M lD0 As we obtain Y .A.e j M / M lD0 Ã Â ³M j!00 j Ä min ³.65. Decomposition of the system of Figure 1.k M/mod L /Tc0 / is the timevarying impulse response. L D 4) Transformations for M D 5 and L D 4 are illustrated in Figure 1. as illustrated in Figure 1.i L C .66. where h D h 1 Ł h 2 . 1 j!0 j Ä min 0 L M H .620) 8 j!00 L Á > 1 < X e M 00 M Y .e j! /X .e j! / D H .66. We obtain that ( ³ ³Á . 2Tc 2M Tc 0 0 0 (1.i xj k M k iD 1 L i where gk.617) gk.68. f / D 1 X .e j M / Y .A. Observing the fact 0 ³ ³ that W . Example 1.e j! / D (1.1.i D h.616) 0 elsewhere In the time domain the following relation holds: C1 X yk D (1.1 (M > L: M D 5.e j! / is zero for M Ä !0 Ä 2 ³ L M .2 (M < L: M D 4.e j! / that has the stopband cutoff frequency within this interval.
in fact. As shown in Figure 1. in the case of a very large interpolation factor L.69.623) C t . let fyk g be the sequence that we need to interpolate to produce the signal z. Elements of signal theory X(e j ω) 0 W(e j ω’) 4.t/. one ﬁnds that if L is large the ﬁlter implementation may require nonnegligible complexity.2 π LFc ω" = ω ’ M f Figure 1. Linear interpolation Given two samples yk 1 and yk . linear and quadratic.7 Time interpolation Referring to the interpolator ﬁlter h of Figure 1.67. after a ﬁrst interpolator ﬁlter with a moderate value of the interpolation factor. 1. the number of coefﬁcients required for an FIR ﬁlter implementation can be very large. the samples fyk D y. limited to interval [.k obtained by the linear interpolation z. kTc0 ]. Consequently.63. we describe below two time interpolation methods.2 π 4Fc ω = 2π f T f 0 H(e j ω’) π /L Fc /2 2π 4Fc ω’ = ω L f 0 V(e j ω’) π /M LFc 2M 2π LFc ω’ f 0 Y(e j ω") π /M LFc 2M 2π LFc ω’ f M=5 0 π LFc 2M 5. the signal z. t 2 <.kTc0 /g may be further time interpolated until the desired sampling accuracy is reached [17]. Rate conversion by a rational factor L=M where M > L.t/.A. is (1.k 1/Tc0 .yk Tc0 yk 1/ .t/ D y k 1 1/ Tc0 .116 Chapter 1.
42π LFc ω" = ω ’ M f Figure 1.69. . Linear interpolation in time by a factor P D 4.68.1.2 π 5Fc ω = 2 π f Tc f L=5 0 H(e j ω’) π /5 Fc /2 2π 5Fc ω’ = ω L f 0 V(e j ω’) π /5 Fc /2 2π LFc ω’ f 0 Y(e j ω") π /5 Fc /2 2π LFc ω’ f 0 M π π L MFc Fc 2L 2 2π MFc L . Multirate systems 117 X(e j ω) 0 W(e j ω’) π Fc /2 2π Fc 5. Rate conversion by a rational factor L=M where M < L.A. Ù Ý ½ Ù ¼ ÞÒ Ù ÂÂ Â Ý ÂÂ ÂÂ ´ Þ ´Øµ ÂÂ·½ Ù Ý ´ ¾µÌ ¼ ´ ½µÌ ¹ ¼ Ì ¼ · ½µÌ Ø Figure 1.
or the order of upsampling and ﬁltering as illustrated in Figure 1. it is possible to exchange the order of downsampling and ﬁltering. The proof of the noble identities is simple. hence Y2 . Let yk 1 .70b.627) zn D 2P P P P 2P P with n 0 D 0. .z/ be a rational transfer function. Therefore.y1 y0 / P In fact. yk and ykC1 be the samples to interpolate by a factor P in the interval [. For this purpose the Lagrange interpolation is widely used. regarding y0 and y1 as the two most recent input samples.k 1/P. i. k P 1.70. we need to consider the sampling instants nTc00 D n and the values of z n D z. The case k D 1 is of particular interest: n n D 0.626) z n D y0 C . : : : . P 1 (1.z/ D X 1 M 1 X . Elements of signal theory For an interpolation factor P of yk .z 1=M W M m /G.z/ D M mD0 . 1. The quadratic interpolation yields the values Â Ã ÃÂ Ã Â Ã Â n0 n0 n0 n0 n0 n0 1C yk C 1 yk 1 C 1 C 1 ykC1 (1. a function expressed as the ratio of two polynomials in z or in z 1 .70c is equivalent to that of Figure 1. For the ﬁrst identity.e. instead of connecting two points with a straight line.8 The noble identities We recall some important properties of decimator and interpolator ﬁlters. it is sufﬁcient to note that W M m M D 1.k 1/Tc0 . the system of Figure 1. their linear interpolation originates the sequence of P values given by (1. P 1 and n D .z 1=M W M m / M / M mD0 (1. : : : . : : : .yk yk 1 / P where n D . and the system of Figure 1. In this case we consider a polynomial of degree 2 that passes through 3 points that are determined by the input samples.nTc00 / are given by .626).. 1.k 1/P (1.628) X 1 M 1 X . known as noble identities.70d.624) C n Quadratic interpolation In many applications linear interpolation does not always yield satisfactory results. 1.70a is equivalent to that of Figure 1. Let G.625) .118 Chapter 1.k 1/ P C n 0 . one resorts to a polynomial of degree Q 1 passing through Q points that are determined by the samples of the input sequence. As an example we report here the case of quadratic interpolation.z/ D Y 1 .z 1=M W M m /G. they will be used extensively in the next section on polyphase decomposition. z n D yk 1 Tc0 P (1. in other words.k C 1/Tc0 ].A. .k 1/P C 1..
z/ D E .1/ .z/ D we can write H .k xn L x4.z/ D 1 X mD0 hm M z 1 1 X mD0 mM Cz Letting h m MC1 z mM C ÐÐÐ C z .0/ .z L /X 4 .A.70.631) E .k (d) Figure 1.1/ .k xn G(zM) (b) M y2. we can always decompose H .z/ as H . Separating the coefﬁcients with even and odd time indices.633) h m MCM 1z mM . let M be an integer. Multirate systems 119 xn M G(z) (a) y1. For the second identity it is sufﬁcient to observe that Y4 .0/ .z/ (1.z/ D 1 X mD0 h 2mC1 z m (1.A.`/ em D h m MC` 0Ä`ÄM 1 (1. let us consider a ﬁlter having transfer function P1 H . Noble identities.1.z/ D G.z L /X .z/ D nD0 h n z n . we get H .629) 1.630) h 2m z m E .k xn G(z) (c) L y3.632) To expand this idea. To explain the basic concept.z L / D Y3 .9 The polyphase representation The polyphase representation allows considerable simpliﬁcations in the analysis of transformations via interpolator and decimator ﬁlters.z/ D G. as well as the efﬁcient implementation of such ﬁlters.z/ as H .z 2 / C z 1 1 X mD0 1 X mD0 h 2m z 2m Cz 1 1 X mD0 h 2mC1 z 2m (1.634) .k G(zL ) y4.z/ D Deﬁning E .z 2 / (1.M 1/ 1 X mD0 (1.
71 for M D 3. for M D 3. In the following. The polyphase representation of an impulse response fh n g with 7 coefﬁcients is illustrated in Figure 1. we can express compactly the previous equation as H . 1.635).M Efﬁcient implementations The polyphase representation is the key to obtaining efﬁcient implementation of decimator and interpolator ﬁlters.`/ .635) ei.71.z/ D M 1 X `D0 z . is given by H .M 1 `/ R .`/ . that is R . Polyphase representation of the impulse response fhn g. : : : . M 1. the polyphase components of H . where the components R .635) is called the type 1 polyphase representation (with respect to M).z M / (1.`/ .120 Chapter 1. then we will extend the results to the general case.637) 1 `/ .z/ D E .z M / (1.z/.z/. we will ﬁrst consider the efﬁcient implementations for M D 2 and L D 2. n D 0.`/ .z/.636) The expression (1. : : : .`/ . called type 2 polyphase representation. A variation of (1.z/ D 1 X i D0 M 1 X `D0 z ` E .`/ .z/.z/ D where E .`/ z i (1. .z/ are permutations of E . 6.`/ . where ` D 0. and E . Elements of signal theory ´¼µ Ñ Ù Ù Ù ½ ¾ ¼ Ò ¹Ñ ¹Ñ ¹Ñ Ù ¼ Ù ½ Ù Ù Ù Ù ´½µ Ñ Ù ¼ Ù ¾ ¿ ¹Ò Ñ Ù ´¾µ ½ Ù ¼ ½ Ù Figure 1.
1/ .61. Figure 1. each operating at half the input frequency and having half the number of coefﬁcients as the original ﬁlter.73b.0/ C N .72. where input samples fxn g are alternately presented at the input to the two ﬁlters e .0/ and e . and N . N In this implementation e .1. the ﬁlter representation can be drawn as in Figure 1. as e .72.`/ operates at half the input rate. By (1. we consider a decimator ﬁlter with M D 2.0/ and e . .639) 2 Therefore the complexity is about one half the complexity of the original ﬁlter.1/ . we can represent H .73a.N (1. Note that the system output is now given by the sum of the outputs of two ﬁlters.z/ as illustrated in Figure 1.`/ multiplications and N .`/ requires N . Multirate systems 121 Figure 1. Optimized implementation of a decimator ﬁlter using the type 1 polyphase representation for M D 2.1/ .`/ 1 additions. so that N D N .1/ be the number of coefﬁcients of e . the total cost is still N multiplications and N 1 additions. the computational complexity in terms of multiplications per second (MPS) is N Fc 2 while the number of additions per second (APS) is given by MPS D APS D .73. Referring to Figure 1. let N be the number of coefﬁcients of h. by the noble identities. To formalize the above ideas.A. Decimator ﬁlter.0/ and . but. this latter operation is generally called serial–to–parallel (S/P) conversion. The efﬁcient implementation for the general case is obtained as an extension of the case for M D 2 and is shown in Figure 1.635). The structure can be also drawn as in Figure 1.638) 1/Fc (1. Implementation of a decimator ﬁlter using the type 1 polyphase representation for M D 2. respectively.74.
63.0/ and e .z/ as illustrated in Figure 1.640) . As illustrated in Figure 1. we can represent H . the ﬁlter representation can be drawn as in Figure 1. In the general case. this latter operation is generally called parallel–to–serial (P/S) conversion.76b.1/ . The type 2 polyphase implementations of interpolator ﬁlters are depicted in Figure 1. we consider an interpolator ﬁlter with L D 2.75.74.78. at the receiver of a transmission 0 system it is often useful to interpolate the signal fr. Interpolatordecimator ﬁlter. by the noble identities. ÜÒ ¹ Ì ¾ ¹ ¹ ´¼µ ´ ¾µ Þ ¹ ¹ Þ ¹ ½ ´½µ ´ ¾µ Þ Ý Ì ¼ ¹ Ì ¾ Figure 1. Elements of signal theory Figure 1.75. By (1.q TQ Let rn D r. Implementation of a decimator ﬁlter using the type 1 polyphase representation. With reference to Figure 1.nTQ / 0 xq D x. fx. The structure can be also drawn as in Figure 1. Interpolator ﬁlter.nTQ /g from TQ to TQ to get the signal 0 /g.76a.q TQ / (1. Remarks on the computational complexity are analogous to those of the decimator ﬁlter case.79.122 Chapter 1.77. Implementation of an interpolator ﬁlter using the type 1 polyphase representation for L D 2.635). efﬁcient implementations are easily obtainable as extensions of the case for L D 2 and are shown in Figure 1. where output samples are alternately taken from the output of the two ﬁlters e .
q TQ /g is then downsampled with timing phase t0 . L 1g. Let yk be the output with sampling period Tc . and L0 is a nonnegative integer number.643) where `0 2 f0.642) 0 with L and M positive integer numbers. Optimized implementation of an interpolator ﬁlter using the type 1 polyphase representation for L D 2. Figure 1. yk D x. Moreover.kTc C t0 / To simplify the notation. Implementation of an interpolator ﬁlter using the type 1 polyphase representation. we assume the following relations: LD TQ 0 TQ MD Tc TQ (1. : : : . we assume that t0 is a multiple of TQ . 0 The sequence fx. t0 0 D `0 C L0 L TQ (1. . we refer to [18] (see also Chapter 14).1.A.641) (1. Multirate systems 123 P/S xn E (0) (z) 2 xn Tc z 1 E (0) (z) yk E (1) (z) (b) E (1) (z) 2 (a) yk T’ = Tc c 2 Figure 1.76. For the general 0 case of an interpolatordecimator ﬁlter where t0 and the ratio Tc =TQ are not constrained. 1.77.
that is for Tc D TQ . 1. Figure 1.124 Chapter 1. Interpolatordecimator ﬁlter.646) .`/ . For the special case M D 1.80. Elements of signal theory xn R(0)(z) xn L z1 R(0)(z) P/S k=L1 R(1)(z) L z1 R(1)(z) k=L2 yk R(L1) (z) (a) yk L R(L1) (z) k=0 (b) Figure 1. where yk D vkCL0 (1.645) 0 The interpolator ﬁlter structure from TQ to TQ is illustrated in Figure 1. L 1 (1.77.78.79.nTQ /g with L phases (1. the implementation of the interpolatordecimator ﬁlter is given in Figure 1. Implementation of an interpolator ﬁlter using the type 2 polyphase representation. : : : .644) fE . Based on the above equations we have yk D x k M LC`0 CL0 L 0 We now recall the polyphase representation of fh.z/g ` D 0.
`0 C L0 L/T0 . Polyphase implementation of an interpolatordecimator ﬁlter with timing phase t0 D .1.`0 C L0 L/TQ . Implementation of an interpolatordecimator ﬁlter with timing phase t0 D 0 .81.80. Multirate systems 125 Figure 1. Q Figure 1. .A.
.643) to be considered.80.n L TQ C `TQ / D x. to downsample the signal interpolated 0 at TQ one can still use the polyphase structure of Figure 1.649) As a result. of L0 samples. With 0 reference to Figure 1. as the relation between fvn g and fyk g must take into account a lead.80. the signal is not modiﬁed before the downsampler. z L0 .126 Chapter 1. Elements of signal theory In other words. fyk g coincides with the signal fvn g at the output of branch `0 of the polyphase structure.`0 .81a. 1. ` D 0.z L M / m D 0. is equivalent to that given in Figure 1. In fact. and n integer.80.`0 / .81b.647) We now consider the general case M 6D 1. once t0 is chosen. L 1. the output fxq g at instants that are multiples of TQ is given by the outputs of the various polyphase branches in sequence. Using now the representation of E . let q D ` C n L. : : : . In particular we have r 0p D r p N0 and x 0p D x p N0 (1. M 1 (1. we determine a positive integer N0 so that L0 C N0 is a multiple of M. considering only branch `0 .650) an efﬁcient implementation of the interpolatordecimator ﬁlter is given in Figure 1.648) The structure of Figure 1.z L / in M phases: E . In any case. the branch is identiﬁed (say `0 ) and its output must be downsampled by a factor M L. 1. Notice that there is the timing lead L0 L in (1. In practice we need to ignore the ﬁrst L0 samples of fvn g. Given L0 . in which we have introduced a lag of N0 samples on the sequence frn g and a further lead of N0 samples before the downsampler. First.m/ . that is L0 C N0 D M0 M (1.nTQ C `TQ / (1. : : : . we have 0 0 0 x`Cn L D x.
as the real and imaginary N components.B.652) 0 a<0 Observing (1.1. each with zero mean and variance equal to 0. N w D A e j' N (1. .s in the interval [0. Generation of Gaussian noise 127 Appendix 1..651).v.654) In terms of real components.652) and (1. if u 1 and u 2 are two uniform r.v. 2³ /.v.v. have a circularly symmetric Gaussian joint probability density function.653) AD and ' D 2³ u 2 (1.v.5. being statistically independent with equal variance.v.B Generation of Gaussian noise Let w D w I C j w Q be a complex Gaussian r. In polar notation.v.s. then p ln. with zero mean and unit variance. in [0. with probability distribution ( 2 1 e a a>0 P[A Ä a] D (1.651) It can be shown that ' is a uniform r.1 u 1 / (1. note N N N N N N that w I D Re [w] and w Q D Im [w]. w is also called circularly symmetric Gaussian r. it results that w I D A cos ' N and w Q D A sin ' N (1. The r.655) are two statistically independent Gaussian r. 1/. and A is a Rayleigh r.
.
3) min J (2.k/. to cause the ﬁlter output y.k/ to replicate as closely as possible d. the output y. we deﬁne the estimation error as e. 2] that will be presented in this chapter is fundamental to the comprehension of several important applications.k/. The FIR ﬁlter in Figure 2.k/ D d.1. N 1.k/. 2. if the ﬁlter input is x. An approximation of the Wiener ﬁlter can be obtained by least squares methods.k/ y.2) In the Wiener theory. The development of this theory assumes the knowledge of the correlation functions of the relevant processes. the N coefﬁcients of the ﬁlter. as the output is formed by summing the products of the delayed input samples by suitable coefﬁcients. 1.k/ is the desired sample at the ﬁlter output at instant k.1 is called transversal ﬁlter.k/ (2.1. let x and d be two individually and jointly wide sense stationary random processes with zero mean.1 The Wiener ﬁlter With reference to Figure 2.N 1 (2.k/j2 ] and the coefﬁcients of the optimum ﬁlter are those that minimize J : fcn g. we have y. The Wiener theory provides the means to design the required ﬁlter. the problem is to determine the FIR ﬁlter so that.Chapter 2 The Wiener ﬁlter and linear prediction The theory of the Wiener ﬁlter [1. Therefore the cost function is deﬁned as J D E[je. n D 0. If we indicate with fcn g.:::. through realizations of the processes involved.nD0.k/ replicates as closely as possible d. the coefﬁcients of the ﬁlter are determined using the minimum meansquare error (MMSE) criterion. : : : .1) If d.k/ D N 1 X nD0 cn x.4) .k n/ (2.
x.k/.k The ﬁlter output at instant k is expressed as y. A brief introduction to estimation theory is given in Appendix 2.k/. The Wiener ﬁlter with N coefﬁcients.k/ (2. .7) 1/. c1 .130 Chapter 2.k N C 1/] (2. in the second half of the Appendix.k/ cT x. the formulation of the Wiener theory is further extended to the case of vector signals. of which reading should be deferred until the end of this section.6) 1] (2. : : : .k/ D xT .A. We deﬁne:1 1. x. : : : . The Wiener ﬁlter problem can be formulated as the problem of estimating d. Filter input vector at instant k xT .k/ D [x.k N C 1/. Coefﬁcient vector cT D [c0 .5) 1 The components of an N dimensional vector are usually identiﬁed by an index varying either from 1 to N or from 0 to N 1.k/ by a linear combination of x. x.8) (2.k/ D cT x. Matrix formulation The problem introduced in the previous section is now formulated using matrix notation.k/ D d.1.k/ c and the estimation error as e. c N 2. : : : . The Wiener ﬁlter and linear prediction Figure 2.
k/d. it follows J D E[d Ł .k/cŁ eŁ .9) We express now the cost function J as a function of the vector c.13) 6 7 : : 6 7 : 4 5 E[d.k/ D x H . Then. N ð N correlation matrix of the ﬁlter input vector.k/xT .k/xŁ . it holds p H D E[d Ł .k/d Ł . We will then seek that particular vector copt that minimizes J . as deﬁned in (1.k/xT . for the particular cases N D 1 and N D 2.k/eŁ . 1. R D E[xŁ .12) 2. : : : .k/xT .d Ł .k/ c H xŁ . .14) (2. Plots of J are shown in Figure 2.n/ n D 0.11) xT . we introduce the following quantities: 1.k/x Ł .17) The cost function J .k/c/.k/ (2.k 1/] 7 6 7 7Dp rdx D E[d.2.k/x Ł .k/ D c H xŁ .k/ and computing the products. The Wiener ﬁlter 131 Moreover. Correlation between the desired output at instant k and the ﬁlter input vector at the same instant 2 3 E[d. is a quadratic function.k/ D d Ł .k/] 6 7 6 E[d.k/] c H E[xŁ .k/] (2.k/x Ł .10) E[d Ł .k/ c H xŁ .k/]c Assuming that x and d are individually and jointly WSS. N 1 (2. J admits one and only one minimum value.k/] D E[. y Ł .k/xT .16) cH p p H c C c H Rc (2.k/] D 6 (2. if R is positive deﬁnite. Recalling the deﬁnition J D E[e.k/d.k/] Then 2 J D ¦d n/] D rdx .k//] (2.k/]c C c H E[xŁ .1.d. considered as a function of c.346).k N C 1/] The components of p are given by [p]n D E[d. Variance of the desired signal 2 ¦d D E[d.k/] 3.k/x Ł .k Moreover.k/] C (2.15) (2.2.
19) In general. we also write p D p I C jp Q (2. The Wiener ﬁlter and linear prediction Figure 2. the real independent variables are 2N .2.Q and also rc J D rc I J C jrc Q J (2.20) .I @c0.I @c1.17) with respect to c. because the vector p and the autocorrelation matrix R are complex. to accomplish this task we deﬁne the derivative of J with respect to c as the gradient vector 2 3 @J @J Cj 6 7 @c0.Q (2.I @c N 1.18) D6 rc J D 7 6 7 @c : : 6 7 : 6 7 4 @J 5 @J Cj @c N 1. as c D c I C jc Q . Plot of J for the cases N D 1 and N D 2.132 Chapter 2. Determination of the optimum ﬁlter coefﬁcients It is now a matter of ﬁnding the minimum of (2.Q 6 7 6 7 @J @J 6 7 Cj @J 6 7 @c1. Recognizing that.
: : : .34) .n i/ D rdx .22) (2. 1. hence we get Rcopt D p Then (2. Observation 2.32) is called the Wiener–Hopf equation (WH).31) For the optimum coefﬁcient vector copt the components of rc J are all equal to zero.c I C jc Q / D jp H rc I c H p D rc I .i rx .c H Rc/ D 2R I c Q C 2R Q c I From the above equations we obtain rc p H c D 0 rc c p D 2p rc c H Rc D 2Rc Substituting the above results into (2. the solution of (2. In scalar form.32) copt. it turns out rc J D 2p C 2Rc D 2. we ﬁnd rc I p H c D rc I p H .27) (2.17).19).26) (2.c H Rc/ D 2R I c I jcT /p D p Q jcT /p D Q 2R Q c Q jp (2.23) (2.33) If R 1 exists.30) (2.c I C jc Q / D p H rc Q p H c D rc Q p H .17) using (2.29) (2.n/ n D 0. The Wiener ﬁlter 133 and R D R I C jR Q If now we take the derivative of the terms of (2.cT I rc Q c H p D rc Q .28) (2.1 The computation of the optimum coefﬁcients copt requires the knowledge only of the input correlation matrix R and of the crosscorrelation vector p between the desired output and the input vector.2.25) (2.cT I rc I . N 1 (2.32) is copt D R 1 p (2.1.24) (2.21) rc Q .Rc p/ H (2. the Wiener–Hopf equation is a system of N equations in N unknowns: N 1 X i D0 (2.
1. : : : . Orthogonality of signals for an optimum ﬁlter. Theorem 2.38) d(k) e(k) y(k) Figure 2. e.1 For c D copt . the notion of orthogonality between random variables is used.36) xT .k/y Ł . N 1.3 depicts the relation between the three signals d. that is E[e.k/] D E[e.k/] D cH 0 D0 For an optimum ﬁlter. E[e.37) (2.134 Chapter 2. (2. : : : . n D 0.k/] D 0 Formally the following is true.k/.k/] D 0 In fact.k/.k/] D c H E[e.k/ and y.d. The Wiener ﬁlter and linear prediction The principle of orthogonality It is interesting to observe the relation E[e.39) for c D copt (2. 1. the ﬁlter is optimum if e.k/ are orthogonal.k/xŁ .k/xŁ . . In other words. 2 Note that orthogonality holds only if e and x are considered at the same instant. using the orthogonality principle.k/y Ł .k/ which for c D copt yields E[e.k/.k/x Ł .1 (Principle of orthogonality) The condition of optimality for c is satisﬁed if e. e.k/.k n/g. and y. N 1 (2.k/c H xŁ . Figure 2.k n/] D 0 n D 0.k/c/] D p Rc (2.2 In scalar form.3.k/xŁ .k/ are orthogonal: E[e.k/ is orthogonal to fx.k/] D E[xŁ .k/ and x.35) Corollary 2.
369) we get J D Jmin C .c copt / is nonnegative and in particular it vanishes for c D copt . using the decomposition (1.45) 3 ¹1 6 : 7 ν D 4 : 5 D U H .k/ D e. Then J assumes the form: J D Jmin C ν H ν D Jmin C N X i D1 N X i D1 ½i j¹i j2 (2.c copt / H R.c copt / H R.44) Recalling that the autocorrelation matrix is positive semideﬁnite.c copt /. Substituting the expression (2.40) we can ﬁnd an alternative expression to (2.41) (2.2.c . The Wiener ﬁlter 135 Expression of the minimum meansquare error We now determine the value of the cost function J in correspondence of copt . we get 2 Jmin D ¦d 2 D ¦d H copt p H p H copt C copt p p H copt (2.42) whereby it follows 2 Jmin D ¦d 2 ¦y (2.c : ¹N copt / (2.k/ are orthogonal for c D copt .47) copt /j2 D Jmin C ½i juiH . In fact.2): d.k/ As e.17) for the cost function J : J D Jmin C . The vector ν may be interpreted as a translation and a rotation of the vector c.43) Using (2.k/ and y.c copt / (2.1.46) where ¹i D uiH .32) of copt in (2.c Let us now deﬁne 2 copt / H U U H .17).c copt / (2.k/ C y. it follows that the quantity .40) Another useful expression of Jmin is obtained from (2.44) allows further observations on J . then 2 2 ¦d D Jmin C ¦ y (2. Characterization of the cost function surface The result expressed by (2.
z/ Px .3. not necessarily causal.1 Let d.5.4.i rx . it follows that rc J is largest along u½max .50) (2.k/. This is also observed in Figure 2. respectively.z/ D Pdx .4.1. Pdx .n/ 8n (2. the equation (2. The result (2. Loci of points with constant J (contour plots).51) . The above observation allows us to deduce that J increases more rapidly in the direction of the eigenvector corresponding to the maximum eigenvalue ½max .33) of the optimum ﬁlter becomes C1 X iD 1 copt.z/ D Pdx .136 Chapter 2.z/ (2. Example 2.48) Taking the ztransform of both members yields Copt . In this case. equation (2. as shown in Figure 2.50) is employed to analyze the system in the general case of an IIR ﬁlter.z/ (2. The Wiener ﬁlter and linear prediction Figure 2. The Wiener ﬁlter in the zdomain For a ﬁlter with an inﬁnite number of coefﬁcients.k/ D h Ł x. Let u½max and u½min denote the eigenvalues of R in correspondence of ½max and ½min .z/Px .49) We note that while (2.z/ Then the transfer function of the optimum ﬁlter is given by Copt . from Table 1.z/H .34) is useful in evaluating the coefﬁcients of the optimum FIR ﬁlter. where sets (loci) of points c for which a constant value of J is obtained are graphically represented. Note that each component is proportional to the corresponding eigenvalue.z/ D Px . Likewise the increase is slower in the direction of the eigenvector corresponding to the minimum eigenvalue ½min .n i/ D rdx .47) expresses the excess meansquare error J Jmin as the sum of N components in the direction of each eigenvector of R. In the 2dimensional case they trace ellipses with axes that are parallel to the direction of the eigenvectors and ratio of axes that is related to the value of the eigenvalues.
f / D 2 ¦d Z 1 2Tc 1 2Tc jPdx . f / Pdx . f /C opt . f /j2 df Px . An application of the Wiener ﬁlter theory.56) where D is a known delay. f / df Px . 2³ /. The autocorrelation function of x and the .e j2³ f Tc / d f 2 D ¦d Z 1 2Tc 1 2Tc Using (2. applying Fourier transform properties.e.1.k/ D Ae j .55) !0 D 2³ f 0 Tc is the tone radian frequency normalized to the sampling period.e j2³ f Tc /j2 df Px .i/ dx (2.!0 kC'/ C w.5. and w is white noise with 2 zero mean and variance ¦w . i. in radians.i rŁ .e j2³ f Tc / (2.54) D 2 ¦d Z Tc 1 2Tc 1 2Tc Example 2.2 We want to ﬁlter the noise from a signal given by one complex sinusoid (tone) plus noise.z/ From (2.k/ D B e j[!0 .z/ D H . The Wiener ﬁlter 137 h d(k) + c x(k) y(k) e(k) Figure 2.50): Jmin D 2 ¦d Z 1 2Tc 1 2Tc Ł Pdx . We assume the desired signal is given by d.k/ (2.40) in scalar notation. The optimum ﬁlter is given by Copt . we get 2 Jmin D ¦d N 1 X i D0 (2. uncorrelated with '. x.55) In (2.1.k D/C'] (2. f / jPdx .2.0.52) copt.53) Ł Pdx . We also assume that ' 2 U.
61) (2.!0 / R 1D 2 I 2 ¦w ¦w C N A 2 Hence.!0 /E H .59) A2 e j!0 : : : A2 e j!0 .n/ D AB e j!0 .40) the minimum value of the cost function J is given by Jmin D B 2 ABe j!0 D E H .!0 / E.!/ D N . the inverse of R is given by # " 1 A2 E.64) (2.N 1 e j!0 : : : e j!0 .58) rdx .63) p D ABe j!0 D E.!0 /E H .n/ D A2 e j!0 n C ¦w Žn (2.N :: : ::: 2/ A2 e j!0 .N we can express R and p as 2 R D ¦w I C A2 E.!0 / Observing that E H .57) (2. : : : . The Wiener ﬁlter and linear prediction crosscorrelation between d and x are given by 2 rx .138 Chapter 2. From (2.!0 / D 2 A 1 C N3 ¦w C N A 2 (2.34): copt D ABe j!0 D B 3e j!0 D E.!/ D [1.!0 / 1/ ] (2.N ::: 2 A 2 C ¦w 3 7 7 7 AB e 7 5 j!0 D (2.60) Deﬁning ET . using (2.n D/ For a Wiener ﬁlter with N coefﬁcients.N 1/ 1/ 2 A 2 C ¦w : : : : : : A2 e j!0 .!0 /E.!/E.!0 / ABe j!0 D B2 D 2 1 C N3 ¦w C N A 2 (2. e j! . e j!.62) (2.N 1/ 2/ 3 7 7 7 7 5 (2.65) 2 where 3 D A2 =¦w is the signaltonoise ratio. the autocorrelation matrix R and the vector p have the following structure: 2 6 6 RD6 6 4 2 6 6 pD6 6 4 2 A 2 C ¦w A2 e j!0 : : : A2 e j!0 .66) .
A Figure 2.! !0 /N j . The Wiener ﬁlter 139 Deﬁning ! D 2³ fTc .e j! / D N 1 X i D0 copt.e j2³ fTc / given by (2. 3 D 30 dB. copt D B e AN j!0 D E. 1.2. Magnitude of Copt . B D A.e j!0 /j D B . the optimum ﬁlter frequency response is given by Copt .68) for f0 Tc D 1=2.! !0 /i (2.i e j!i D E H .67) 8 j!0 D > B N 3e > < A 1 C N3 Copt .! !0 / (2.!/copt D that is. 2. for 3 × 1. 3.1. X B 3e j!0 D N 1 e A 1 C N 3 i D0 j . and N D 35. .!0 /.6. jCopt . Jmin becomes negligible.68) ! 6D !0 We observe that.e j! / D > B 3e j!0 D 1 e > : A 1 C N3 1 e ! D !0 j .
Forward linear predictor The estimate of x.k/.7. In this case. : : : .2 Linear prediction The Wiener theory considered in the previous section has an important application to the solution of the following problem. Jmin D B 2 . 2. given the values of x.7.e. 3. There exists also the problem of predicting x. when the power of the useful signal is negligible with respect to the power of the additive noise. The plot of jCopt . x. attempts to estimate the value of x.k N /] (2. as the signaltonoise ratio vanishes the best choice is to set the output y to zero. The Wiener ﬁlter and linear prediction Conversely.k i/ (2.140 Chapter 2.k N /.k O 1// D N X i D1 ci x. Indeed. Let x be a discretetime WSS random process with zero mean. jCopt . x(k) Tc x(k1) Tc x(k2) Tc c N1 x(kN) c1 c2 cN ^ x(k x (k1) ) Figure 2. prediction consists in estimating a “future” value of the process starting from a set of known “past” values. copt D 0.e j2³ f Tc /j is given in Figure 2. the system is called the onestep backward predictor of order N .k/. let us deﬁne the vector xT . .k N C 1/.k j x.k 2/.k 1/ D [x. x.69) The onestep forward predictor of order N . for 3 ! 0. : : : .k 1/. i.k 1/. In particular.e j!0 /j D 0. given xT . it results in 1. x. Linear predictor of order N.6.70) The block diagram of the linear predictor is represented in Figure 2.k/ is expressed as a linear combination of the preceding N samples: x. 2.
N / with R N N ð N correlation matrix.78) (2.0/ H r N copt (2.72)).76) E[x . .k (2.73) 1/ (2. 1.72) to determine the predictor coefﬁcients.k j x.k/j2 ] (2.k T 1/] D R N 2 6 6 1/] D 6 6 4 3 rx .2.75) (2. the optimum coefﬁcients satisfy the equation R N copt D r N Moreover.77) Applying (2.k/] D ¦x D rx .1/ rx . and p D E[d.k/ 2.k 1/] D E[x.80) 0N copt rN RN where 0 N is the column vector of N zeros.71) i/ ci x. Cost function J (given by (2. Jmin D J N D rx . we can use the optimization results according to Wiener.k/ D x.2/ 7 7 7 : 7 D rN : 5 : rx .79) We can combine the latter two equations to get an augmented form of the WH equation for the linear predictor: " #" # " # H 1 JN rx . Desired signal d.k Optimum predictor coefﬁcients If we adopt the criterion of minimizing the meansquare prediction error.k/ x.k 3.40) we get the minimum value of the cost function J .0/ r N D (2.k O N X i D1 1// (2.k/xŁ .k Ł 1/x .0/ (2.k/xŁ .k/x Ł .74) (2.69)) xT .32). from (2. Linear prediction 141 This estimate will be subject to a forward prediction error given by f N . Filter input vector (deﬁned by (2.k/ D x.k/ D x. Then it turns out: 2 2 ¦d D E[x. J D E[j f N .2. We recall the following deﬁnitions.
k/ D N X i D0 0 ai.N ] are directly obtained by substituting 0.8.N f (k) N Figure 2. The Wiener ﬁlter and linear prediction Forward “prediction error ﬁlter” We determine the ﬁlter that gives the forward linear prediction error f N . : : : .k/ D x.82) in (2.81) and taking care to extend the equation also to i D 0.N (2.i Ä D i D0 i D 1.k/ We introduce the vector ( 0 ai.83) in (2.83) where a D copt . 2. : : : .i x.N a’ 1.82) which can be rewritten as a0N 1 a (2. a0N .k i/ (2. Substituting (2.N a’ N1. a0 .142 Chapter 2.84) as shown in Figure 2.N N X i D1 copt. 0T The coefﬁcients a N D [a0 .81) D 1 copt. f N . .N a’ N.N a’ 2. we obtain f N .N x. N ½ (2.N 1. Forward prediction error ﬁlter.8.85) R N C1 a N D 0N x(k) Tc x(k1) Tc x(k2) Tc x(kN) a’ 0.k i/ (2.80): ½ Ä JN 0 (2. For an optimum predictor.
N/ processes. comparing (2.78) with the Yule–Walker equation (1.537) allows us to state what follows: given an AR process x of order N . producing at the output only the uncorrelated or “white” component.81) with (1. the optimum prediction coefﬁcients copt coincide 2 with the parameters a of the process and. can be used. the prediction error f N coincides with white noise having statistical power J N . that is. for copt D a. Linear prediction 143 With a similar procedure. copt ].k/. the parameters copt and J N can be determined. Using the predictor then we determine the prediction error f f N .k i C 1/ (2. an allpole ﬁlter with 0T 2 coefﬁcients a N D [1. we can derive the ﬁlter that gives the backward linear prediction error.9.k/g of power ¦w D J N as input. In general.9. b N .k N/ N X i D1 gi x.k/ D w. while prediction can be interpreted as the analysis of an AR process.2. by estimating the autocorrelation sequence over a suitable observation window. we can observe that this ﬁlter has whitening properties. the AR model may be regarded as the synthesis of the process. Moreover. As illustrated in Figure 2. . in that it is capable of removing the correlated signal component that is present at the input. Analysis and synthesis of AR . moreover. if the order of the prediction error ﬁlter is large enough.k/ D f N . Actually. Figure 2.2.87) where B is the backward operator that orders the elements of a vector backward.k/g. for a process x.k/g. J N D ¦w .86) It can be shown that the optimum coefﬁcients are given by BŁ gopt D copt (2. from the last to the ﬁrst (see page 27). Relation between linear prediction and AR models The similarity of (2.k/ D x.518) we ﬁnd f N . having white noise fw. given a realization of the process fx.k/g. To reproduce fx.
1/j2 (2.1/ >c : opt.0/ > > 0.0/ þ As a0 D 1.1/j2 x ) 8 ².0/ ) 8 > copt.1 J1 rx .1/j2 > > ².0/ rx .0/ rŁ .2/ j².2/ ² 2 . These results extend to the complex case the formulae obtained in Section 1.1 x < 1r > > 0 >a D : 1.1/ rx .1 D > < 1 ² Ł .1/j2 (2.1 where þ þ þ r .1 a0 1.144 Chapter 2.2/ C j².1/j2 x rx .0/ 2j².2.12.1/ 1r (2.1/j2 j².0/ jrx .540).1/ þ þ þ x x 1r D þ þ D r2 .0/ > > 0 >a : ž N D 2. introduced in (1. ž N D 1.0/ #" a0 0.1/ >c > opt.2/ 1 j².1/ x r2 .1/j2 C ² Ł2 . From " rx . . 8 > a0 D > 1.91) j².1/ x rx .92) We note that in the above equations ².2/ x r2 .n/ is the correlation coefﬁcient of x.2 and J2 1 D rx .90) D rx .0/ rŁ .0/rx .1/ rx .0/ rŁ .2 > < > > 0 >a D : 2.2 D 1 j². The Wiener ﬁlter and linear prediction First and second order solutions We give below formulae to compute the predictor ﬁlter coefﬁcients and prediction error ﬁlter coefﬁcients for orders N D 1 and N D 2. it turns out 0.2/ r2 .1 # D " J1 0 # (2.1/².1/j2 ² Ł .1/ 1.1/rx .1 (2.0/ x þ rx .93) 1.1/ rx .89) jrx .0/ jrx .1 D > < > > : a0 D ².1/rx .88) it results 8 > a0 D J1 r .1/j2 J1 D1 rx .1/².1 8 > J D 1r > 1 > < rx .2/j2 (2.
given in Section 2. We set: J0 D rx .k/j2 ] It results in 0 Ä Jn Ä Jn with J0 D rx .97) corresponds to the scalar equations: a0 D a0 k.2.1/ 2. R N C1 is symmetric and the computational complexity of the Delsarte–Genin algorithm (DGA).97) 1 Then (2.2.103) .85). B 0 1n D .95) 1n Jn 1 1 (2. and Toeplitz.100) Jn D Jn 1 . N .1 jCn j2 / (2. in which R N C1 is positive deﬁnite. Jn represents the statistical power of the forward prediction error at the nth iteration: Jn D E[j f n .0/ and J N D J0 N Y nD1 1 (2. we report a stepbystep description of the LDA: 1.0/ 10 D rx .104) . We calculate Cn D " 0 an D (2. Moreover.n k. Linear prediction 145 2. n D 1.rnC1 /T an (2.1 jCn j2 / We now interpret the physical meaning of the parameters in the algorithm. with a computational complexity proportional to N 2 . is halved with respect to that of LDA. 1. Here. In the case of real signals. : : : . Hermitian. nth iteration. instead of N 3 as happens with algorithms that make use of the inverse matrix.1 The Levinson–Durbin algorithm The Levinson–Durbin algorithm (LDA) yields the solution of matrix equations like (2.101) n½1 (2. Initialization.n 1 1 C Cn an 0Ł k. : : : .2.n with a0 0.98) D 0.n 1 0 D 1 and an.96) # C Cn " 0 an 0 BŁ 0 an 1 0 # (2. n (2.94) (2.2.2. 2.99) (2.n 1 k D 0.102) (2.
k 1/ # (2.n n 0.k/j2 ] 1/] 1 .105) In other words.k/ C Ð Ð Ð C a0 Ł x.k/ 1 (2. Its analysis permits us to implement the prediction error ﬁlter via a modular structure.97) we obtain " f n .k/ D a0 Ł x.k/ D D fn 0 an 1 0 #T xnC1 . Deﬁning xnC1 . Lattice ﬁlters We have just described the Levinson–Durbin algorithm.k/ C Cn " 0 an 0 BŁ #T xnC1 .k/ x. : : : .101) and (2.k we can write: " xnC1 .n From (2.k n/ # D " x. from (2. x.k n/ D anT xnC1 .k 1/ By a similar procedure we also ﬁnd bn .k/j ] E[ f n Ł 1 .k 1/] (2.96).110) n/]T (2.k/j ] D E[jbn jCn j Ä 1 The coefﬁcients fCn g are called reﬂection coefﬁcients or partial correlation coefﬁcients (PARCOR).n (2.k/ 0.n (2.k/bn 1 . along with (2. Cn satisﬁes the following property: 0 Cn D an.n x. noting that E[j f n 2 1 .107) 1/j2 ].k/ D bn 1 .108) D E[jbn 2 1 .105).109) We recall the relation for forward and backward linear prediction error ﬁlters of order n: 8 0 0 < f n .k/ C Ð Ð Ð C an.107) we have (2.111) : b . 1n can be interpreted as the crosscorrelation between the forward linear prediction error and the backward linear prediction error delayed by one sample.k (2.113) .k/ D xn .146 Chapter 2.k/ (2. The Wiener ﬁlter and linear prediction The following relation holds for 1n : 1n 1 D E[ f n Ł 1 .k/ D [x. by substitution.k E[j f n 1 .k/ n nC1 n.106) Finally.k/ C Cn bn 1 .k/bn 1 .k/.k/ xn .k n/ D a0 B H x . we get Cn D and.k/ D a0 x.112) 1 . from (2.k Ł 1/ C Cn f n 1 .
The optimum coefﬁcients Cn . also known as the split Levinson algorithm [3].116) 3 Faster algorithms. . n D 1. 1. Observation 2. If the conditions jCn j Ä 1.1/ C rx . the DGA. the ﬁlter is minimum phase. 1.1/ D rx . We set v0 D 1 v1 D [1. in which the output is given by f N . 3. 1]T þ0 D rx .115) (2.2 The Delsarte–Genin algorithm In the case of real signals.3 Here is the stepbystep description. with a complexity proportional to N . are veriﬁed.k/ D b0 .0 the block diagram of Figure 2. Initialization.2/ (2.2.2. N . therefore one can change N without having to recalculate all the coefﬁcients.108). Linear prediction 147 f0 (k) f1 (k) f m1 (k) fm (k) f N1 (k) fN (k) CN C1 x(k) Cm C* 1 b0 (k) * Cm * CN Tc b1 (k) bm1 (k) Tc bm (k) bN1 (k) Tc bN (k) Figure 2. : : : . : : : .0/ þ1 D rx . n D 1.1/ 0 1 D rx .k/ and a0 D 1 0. f 0 . N . This property is useful if the ﬁlter length is unknown and must be estimated. 2. at least for N ½ 10. taking into account the initial conditions.10 is obtained.114) 2. we ﬁnd that all predictor error ﬁlters are minimum phase. We list the following fundamental properties.2 From the above property 2 and (2. (2. have been proposed by Kumar [4]. are independent of the order of the ﬁlter.10. Finally.2.0/ C rx . The lattice ﬁlters are quite insensitive to coefﬁcient quantization.log N /2 . Lattice ﬁlter.k/ D x. further reduces the number of operations with respect to the LDA.
1).2/ C rx .3 The least squares (LS) method The Wiener ﬁlter will prove to be a powerful analytical tool in various applications.k/g and fd.þn Ä vn D n 1 2 1 n 1/ n 2/ (2. The Wiener ﬁlter and linear prediction 2. We reconsider the problem of Section 2.þn .k/ is given by (2. : : : . in which from the estimate of rx we construct R as a Toeplitz correlation matrix.148 Chapter 2.rx .n C 1// C [vn ]2 .N 1 min E (2.127) . N . 2.121) Ä 0 vn n 1 1 0 an D vn ½ (2. from a practical point of view.n// C Ð Ð Ð (2.1/ C rx .:::.k/g k D 0. Based on the observation of the sequences fx.130).125) where y.nD0. n D 2. in particular it is [vn ]1 D [vn ]nC1 D 1. introducing a new cost function.k/ (2.119) T D rnC1 vn D .k/ y. often only realizations of the processes fx.k/g are available. : : : . Two possible methods are: 1) the autocorrelation method. and various alternatives emerge.124) Jn D þn Cn D 1 We note that (2.rx .122) (2. Therefore to get the solution it is necessary to determine estimates of rx and rdx . 2]. in which we estimate each element of R by (2. K 1 and fd.118) (2.120) ½n D þn þn 1 ½n ½n ½n (2. K 1 (2. : : : .126) k D 0.117) 2 þn D 2þn Þn þn ½ C Ä vn 1 0 0 vn 1 ½ 0 Þn 4 vn 2 5 0 2 3 (2.k/g and of the error e.k/ D d.1. In this case the matrix is only Hermitian and the solution that we are going to illustrate is of the LS type [1. according to the least squares method the optimum ﬁlter coefﬁcients yield the minimum of the sum of the squared errors: fcn g.1.123) (2. We compute Þn D . and 2) the covariance method. one of which is indeed prediction. nth iteration. However.120) exploits the symmetry of the vector vn .
N / 7 6 x.3. n/ depend on both indices .K 1/ x. 1. deﬁned by (2.K {z N/ In (2.N / x.k/ x Ł .K 1/  x.n/ D K 1 X kDN 1 d.1/ : : : 1.N 2/ y.0/ x. n/ and not only upon their difference.K 1. n D 0.0/ to x.n/ O n/ (2.1). can be 32 76 76 76 54 } c0 c1 : : : cN 1 3 7 7 7 5 (2.k/j2 (2. N 1 (2. is LðN where L D K N C1. The least squares (LS) method 149 where ED K 1 X kDN 1 je. given by (2.n/ D .132) (2.k/g Ed D K 1 X kDN 1 jd.128) Note that in the LS method a time average is substituted for the expectation (2.K N C 1/rx . n/ D .133) in which the values of 8.N 1/ x.134) .N 1/ 6 y.K #.130) #. K ::: ::: :: : x.i. : : : .k i/ x.k/g.i. k D N expressed as 3 2 2 x.i O N C 1/rdx .478) for an unbiased estimate of the correlation.i.k n/ i.129).k n/ n D 0. Data windowing In matrix notation.K 1/. 1. Matrix formulation We deﬁne 8. Energy of fd.i.129) data matrix T 2/ : : : x. Other choices are possible for the input data window. the output fy.N 1/ 7 6 6 7D6 6 : : : : : : 5 4 4 : : : y. N 1 (2. : : : . We give some deﬁnitions: 1.129) we note that the input data sequence actually used goes from x. which gives the MSE. The case examined is called the covariance method and the data matrix T.3).k/j2 (2.131) Using (1.2. especially if K is not very large. n/ D K 1 X kDN 1 x Ł . the following identities hold: 8. : : : .
2. The Wiener ﬁlter and linear prediction 2.17).0/. D K 1 X kDN 1 xŁ . N 8. 3. the gradient of (2.e.140) Then the vector of optimum coefﬁcients based on the LS method.141) . 0/ 6 D6 : : 4 : 8. satisﬁes the normal equation cls D ϑ (2.0. #. Input autocorrelation matrix 2 8. c ϑ/ (2.1/.1. 1/ : : : ::: ::: :: : 1.136) Then the cost function can be written as E D Ed cH ϑ ϑ H c C cH c (2.k/. 1/ 8.k/ (2.N 8. Crosscorrelation vector between d and x ϑ T D [#.0.k/xT .137) is given by rc E D 2.N 3.1.N 1/] (2. can be written as D TH T (2. We note that the matrix T is Toeplitz. N 1/ 1/ 1/ 3 7 7 7 5 (2. is Hermitian.0. i.150 Chapter 2.139) with T input data matrix deﬁned by (2. 0/ 6 8.135) 1. Determination of the optimum ﬁlter coefﬁcients By analogy of (2. N : : : 1. are real and nonnegative.N 8. 1/ : : : 8.129). cls . Eigenvalues of 4.137) with (2. : : : . #. is positive semideﬁnite.k/xT . 0/ 8.138) Properties of 1.1. .137) Correlation matrix is the time average of xŁ .
the solution to (2.k D 2 K 1 X kDN 1 n/e.147) C j .k n/ D 0 n D 0.n/ n D 0. N 1 (2. then the optimum coefﬁcients must satisfy the conditions K 1 X kDN 1 emin .k j x.I E C jrcn .k//] (2. that is cls K !1 ! copt (2. cls .k/x Ł .k/ If we denote with femin . 1.145) In other words. we have rcn E D rcn .k n/eŁ .2.148) . i/cls. (2. for K ! 1 the covariance method gives the same solution as the autocorrelation method.k/g the estimation error found with the optimum coefﬁcient values. : : : .128). 1. : : : .141) becomes a system of N equations in N unknowns: N 1 X i D0 8.k/ x. taking the gradient with respect to cn .n.Q E D K 1 X kDN 1 [ x Ł .142) In the solution of the LS problem.i D #.k/ n/eŁ .141) corresponds to the Wiener–Hopf equation (2. The least squares (LS) method 151 If 1 exists.141) is given by cls D 1 ϑ (2.144) We ﬁnd that the LS solution tends toward the Wiener solution for sufﬁciently large K .1 The principle of orthogonality From (2.32).3.143) K (2. the equation (2. In scalar notation.k n/e.k n/e. As for an ergodic process (2.146) 2.132) yields: K and 1 ϑ N C1 !p K !1 1 N C1 !R K !1 (2. j x Ł . N 1 (2.k/ x Ł .3.
Note that for c D cls we have d.N 1/ x Ł . because of the orthogonality (2.N / 7 76 7 76 7 : : 54 5 : N/ d.N /.153) Ey (2.N 1/ 2/ 7 6 d.152) (2.1/ 7 6 x Ł .150) An alternative expression to Emin uses the energy of the output sequence: Ey D K 1 X kDN 1 jy.130).K 1/] (2.k/ C emin .151) observing (2.k/ D 0 (2. N 1.156) .149) expresses the fundamental result: the optimum ﬁlter output sequence is orthogonal to the minimum estimation error sequence. d.k/ a linear combination of fx. (2.k n/g. the minimum cost function can be written as Emin D Ed ϑ H cls (2.k/y Ł .149) Equation (2.2/ x x #.K : : : x Ł . it follows that Ed D E y C Emin from which. : : : .141) in (2.K 1/ (2.K : :: : : : Ł .151). n D 0. we have K 1 X kDN 1 emin .N 2/ x Ł . d.152 Chapter 2.36). substituting (2.k/ then.N 1/ : : : x Ł .k/j2 D c H c (2.N / 6 #.n/ we get 2 3 2 Ł #.131) of #.0/ x .k/ D y. being y.N 1/. Expressions of the minimum cost function Substituting (2.155) from the deﬁnition (2.141) in (2. : : : .N 1/ 6 7 6 6 7D6 : : : : : : 4 5 4 : : : Ł .137).K ::: x 32 3 1/ d. Moreover.149) between y and emin . we get Emin D Ed H where E y D cls ϑ . The Wiener ﬁlter and linear prediction which represent the timeaverage version of the statistical orthogonality principle (2.154) The normal equation using the T matrix Deﬁning the vector of desired samples dT D [d. 1.0/ Ł .
from (2. The least squares (LS) method 153 that is ϑ D TH d Thus.164) Correspondingly.T H T/ 1 TH d (2.161) depend only on the desired signal samples and input samples.129) the vector of ﬁlter output samples yT D [y.157) Associated with system (2.166) .160) we get y D Tcls D T.158).2. Tc D d From (2.162) can be related to the input data matrix T as y D Tc This relation will still be valid for c D cls .157). the solution c is unique only if the columns of T are linearly independent. Moreover. that is the case of nonsingular T H T.N 1/. it is useful to introduce the system of equations for the minimization of E. Let I be the identity matrix: the difference O? D I ODI T.139) and (2.T H T/ 1 (2. the estimation vector error is given by emin D d y (2.161) We note how both formulae (2. y.T H T/ 1 T H can be thought of as a projection operator deﬁned on the space generated by the columns of T. if .T H T/ 1 TH d (2. the solution is cls D .158) (2.K 1/] (2. using the (2.160) and (2.150) becomes Emin D d H d d H T.N /.T H T/ 1 TH (2.159) must be overdetermined with more equations than unknowns.3. that is the system of equations (2. This requires at least K N C 1 > N . Geometric interpretation: the projection operator In general.165) The matrix O D T.T H T/ 1 (2.159) exists. : : : . y.158). the normal equation (2. and from (2.160) and correspondingly (2.141) becomes T H Tcls D T H d (2.163) TH d (2.
11.170) can be found by the successive substitutions method with O. In fact. This is what we will do in this section after taking a closer look at the associated system of equations (2.165) emin D d y D O ?d (2.169) In Figure 2. a solution to the system (2.154 Chapter 2. y. if T is nonsingular. Moreover. 2. The Wiener ﬁlter and linear prediction d emin y Figure 2.168) (2. Factorization of T T D LU (2. In general. (2. let us consider the solutions to a linear system of equations Tc D d with T N ð N square matrix. is the complementary projection operator.N 2 / operations.T H T/ does not exist.159). 2.161) can be written as H Emin D emin emin D d H emin D d H O ? d (2. Relations among vectors in the LS minimization. orthogonal to O . which involves three steps: a. from (2. If T is triangular and nonsingular. one can use the Gauss method.160) must be reexamined.3. the solution c D T is unique and can be 1.149)).164) y D Od and from (2. .11 an example illustrating the relation among d.170) 1d exists.167) where emin ? y (see (2.2 Solutions to the LS problem If the inverse of . In general.171) with L lower triangular having all ones along the diagonal and U upper triangular. and emin is given. the solution of the LS problem (2. If T obtained in various ways [5]: 1 (2.
u2 .176) (2.N 3 / operations. one can use the generalized Shur algorithm with a complexity of O. the factorization (2.170) [5]: in particular we will consider the method of the pseudoinverse. two unitary matrices V and U exist. This method requires O.369) how the N ð N Hermitian matrix R can be decomposed in terms of a matrix U of eigenvectors and a diagonal matrix of eigenvalues.177) (2. The least squares (LS) method 155 b. v N ] N ðN . : : : . First. We also recall the Kumar fast algorithm [4]..g. If T is Toeplitz and nonsingular. it is necessary to use alternative methods to solve the system (2. Solution of the system in c Uc D z through the successive substitutions method. Singular value decomposition (SVD) of T We have seen in (1. Solution of the system in z Lz D d through the successive substitutions method.N 3 / operations and O. ¦2 . ¦ R / U D [u1 . if T 1 does not exist. 3. about half as many as the Gauss method.171) becomes the Cholesky decomposition: T D LL H (2. : : : . 4. This method requires O. c.2.¦1 . u L ] LðL V D [v1 . v2 .3.172) with L lower triangular having nonzero elements on the diagonal. Given an L ð N matrix T of rank R. because T is not a square matrix. If T is Hermitian and nonsingular. Now we extend this concept to an arbitrary complex matrix T.N 2 /: generally it is applicable to all T structured matrices [6]. 0 LðN ¦1 > ¦ 2 > Ð Ð Ð > ¦ R > 0 UU H D I LðL VV H D I N ðN (2.175) D D diag. so that T D U VH with 8 >D D> : 0 9 0 > > .174) (2. However.N 2 / memory locations.179) (2.178) (2. we will state the following result.173) (2. : : : . e.
180) UH D R X i D1 ¦i 1 vi uiH (2.183) In this case T# d coincides with the solution of system (2.T H T/ 1 TH (2.12. N /. is given by the matrix T# D V where # # (2.12. Again.1 The pseudoinverse of T.177) the f¦i g.181) Ä D D 1 0 0 0 ½ D 1 D diag ¦1 1 . ¦ R 1 Á (2.TT H / 1 (2. hence there are inﬁnite solutions to the system (2. ¦2 1 . of rank R. it follows U H TV D as illustrated in Figure 2. The Wiener ﬁlter and linear prediction Figure 2. L ð N .170). Singular value decomposition of matrix T. Note that in this case there are fewer equations than unknowns.T/. .L . : : : .170) has more equations than unknowns.184) 4 We will denote the rank of T by rank. Case of an underdetermined system (L < N ) and R D L.141). it can be shown that T# D T H .156 Chapter 2. : : : .182) We ﬁnd an expression of T# for the two cases in which T has full rank. R. Note that in this case the system (2. In (2. Using the above relations it can be shown that T# D . are singular values of T. Case of an overdetermined system (L > N ) and R D N . i D 1. Deﬁnition 2. Being U and V unitary.4 that is R D min.
185) cls D 3.190) Only solutions (2.3. ¦i2 vi (2. rank. rank.T/ D L: .187) coincide with the solution (2.128). E D jjejj2 D jjy djj2 D jjTc djj2 . and simultaneously minimizes the norm of the solution.189) and cls is the minimum norm solution of an underdetermined system of equations. ¦i2 T H ui (2.142). cls D R X uH d i i D1 1 R X vH TH d i i D1 1 1 (2.186) TH (2.T H T/ b. rank. in other words it solves the problem of ﬁnding the vector c that minimizes the squared error (2.186) and (2. T is nonsingular. If L > N and a. or in the form (2.187) and cls is the LS solution of an overdetermined system of equations (2.T/ D N .T/ D R (also < L).T/ D N . The computation of the pseudoinverse T# directly from SVD and the expansion of c in terms of fui g. T# D T 2.170).189). We list the different cases: 1. from (2.185) in the cases (2.TT H / b. then T# D T H . If L < N and a.2. If L D N and rank. The least squares (LS) method 157 Minimum norm solution Deﬁnition 2.2 The solution of a least squares problem is given by the vector cls D T# d where T# is the pseudoinverse of T. rank. jjcjj2 . for L > N and rank. i.T/ D R (also < N ).T/ D L.e. fvi g and f¦i2 g have two advantages with respect to the direct computation of T# in the form (2.188) (2.185). The constraint on jjcjj2 is needed in those cases in which there is more than one vector that minimizes jjTc djj2 .T/ D N .185) (2. By applying (2. for L < N and rank. the pseudoinverse matrix T# gives the LS solution of minimum norm. then T# D .187).
2. New York: Cambridge University Press. 254–267. [5] G. Kay. 2462– 2473. IEEE Trans. Genin. [6] N. There are two algorithms to determine the SVD of T: the Jacobi algorithm and the Householder transformation [7]. NJ: PrenticeHall.TT H / 1 . P. [2] M. Marple Jr. Adaptive ﬁlters: structures. IEEE Trans. M. A. vol.. 1993. pp. T. The SVD also gives the rank of T through the number of nonzero singular values. Honig and D. vol. Ciofﬁ. 1989. [9] S. vol. 1988. MA: Kluwer Academic Publishers. 1988. V. Numerical Recipes. Englewood Cliffs. 34. Boston. Matrix computations. The required accuracy in computing T# via SVD is almost halved with respect to the computation of . Englewood Cliffs. 1996. The Wiener ﬁlter and linear prediction 1. 9]. [4] R. H. algorithms and applications. 1985. AlDhahir and J. [7] S. Feb. L. 470–478. Press. Baltimore and London: The Johns Hopkins University Press. Englewood Cliffs. Digital spectral analysis with applications. 33. NJ: PrenticeHall. 3rd ed.. “Fast computation of channelestimate based equalizers in packet data transmission”. on Signal Processing.158 Chapter 2. Speech and Signal Processing. T. Haykin. 1987. “A fast algorithm for solving a Toeplitz system of equations”. 1995. H. Vetterling. 2nd ed. B. on Acoustics. Fundamentals of statistical signal processing: estimation theory. [10] S. Englewood Cliffs. G. Speech and Signal Processing. Kay. NJ: PrenticeHall. June 1986. W.. Delsarte and Y.. which report examples of realizations of the algorithms described in this section. IEEE Trans. F. Bibliography [1] S. pp. We conclude citing two texts [8. “The split Levinson algorithm”. . Modern spectral estimationtheory and applications. on Acoustics. 3rd ed. M. S. [3] P. Adaptive ﬁlter theory. [8] L. Flannery and W. Kumar. pp. Messerschmitt. M. Golub and C.T H T/ 1 or . Nov. NJ: PrenticeHall. van Loan. 43. 1984.
Þ j þ/ dÞ D 0 8þ (2.d/. þ/ D px . the solution would be trivial. The integral (2.2 The estimator h. respectively.193) [Þ h.Þ. Theorem 2. that is x D þ. we often know only the joint probability density function of the two r.191) MMSE estimation Let pd .þ/ that minimizes J is given by the expected value of d given x D þ. somehow related via the function f . þ/ dÞ dþ J D E[e ] D 1 1 Z D C1 1 px .196) 2 1 .þ/] pdjx . however. that is x D f .v. using as estimate of d the function O d D h. and pdjx . In any case. if f were known and the inverse function f 1 existed.x/ the estimation error is given by eDd O d (2. that is Z C1 Z C1 2 [Þ h. We wish to determine the function h that minimizes the meansquare error. Obviously. The estimation problem 159 Appendix 2.Þ.Þ j þ/ is used. h.þ/ 6D 0.s.Þ. let the value of x equal to þ. pdx .194) (2. Using the variational method (see Appendix 8. we ﬁnd that this occurs if Z C1 [Þ h. On the basis of an observation. þ/.A.A The estimation problem The estimation problem for random variables Let d and x be two r.2.þ/] pdjx .Þ j þ/ dÞ dþ 2 1 where the relation pdx .þ/ Z C1 (2.195) is minimized for every value of þ.193) is minimum when the function Z C1 [Þ h.192) (2. moreover let px . The estimation problem is to determine what the corresponding value of d is.Þ j þ/ the conditional probability density function of d given x D þ.Þ j þ/ dÞ 1 (2.þ/ be the probability density functions of d and x.þ/]2 pdx .þ/ pdjx .þ/ D E[d j x D þ] Proof. 8þ.A).þ/]2 pdjx .Þ/ and px .v.s.
A.201) Example 2.A.2 Let x D d C w.s. the MAP criterion becomes the maximum likelihood (ML) criterion.þ/ D 1 C1 Þ pdjx . Example 2. it can be shown that h.þ/ (2.þ/ (2. respectively.160 Chapter 2.s with mean values md and mx . and covariance c D E[.þ/ D md C c .0. x N D þ N (2.200) The corresponding meansquare error is equal to Jmin D 2 ¦d Â c ¦x Ã2 (2. The Wiener ﬁlter and linear prediction that is for Z h. x 1 D þ1 . which yields O d D arg max pdjx .þ/ D tanh. : : : .199) Examples of both MAP and ML criteria are given in Chapters 6 and 14.197) from which the (2.Þ j þ/ Þ (2.d md /. .v.203) the estimation of d is obtained by applying the following theorem.1 Let d and x be two jointly Gaussian r. þ/ dÞ px . 1g with P[d D 1] D P[d D 1] D 1=2. it can be shown that [10] h. whose proof is similar to the case of a single observation. where O d D arg max pxjd .v.21).þ j Þ/ Þ (2. If the distribution of d is uniform. 1/ and d 2 f 1.202) Extension to multiple observations In the case of several observations. After several steps.Þ j þ/ dÞ D Z C1 1 Þ pdx . For w 2 N .þ 2 ¦x mx / (2. where d and w are two statistically independent r. O An alternative to the MMSE criterion for determining d is given by the maximum a posteriori probability (MAP) criterion.194) follows.Þ.198) where the notation arg max is deﬁned in (6.x mx /].
using the deﬁnitions (2. þ1 .4 Given the vector of observations x.206) β (2. it is often convenient to consider a linear function h. þ N / In the following. : : : .A.s.x1 .2. and O (2. þ N / D E[d j x 1 D þ1 .s with zero mean and the following second order description: ž Correlation matrix of observations R D E[x xT ] ž Crosscorrelation vector p D E[dx] For x D β. Letting c D [c1 .209) d D cT x C b where b is a constant. : : : . Theorem 2. d D h.β/ D pT R and 2 Jmin D ¦d 1 (2.3 Let d.Þ.d O d/2 ] is given by h. : : : .þ1 . In the case of realvalued r. : : : .þ1 .x1 :::x N .v. : : : .208) MMSE linear estimation For a low complexity of implementation.Þ j x 1 D þ1 . it can be shown that h.207) pT R 1 p (2.205) (2. in the case of multiple observations the estimate is a linear combination of observations. x N D þ N ] Z C1 D Þ pdjx1 :::x N . x D [x1 . The estimation problem 161 Theorem 2. be realvalued jointly Gaussian r. to simplify the formulation we will refer to r. the MMSE linear estimator of d has the following expression O (2.A.v. : : : . x N /.204) Z D Þ pd.s with zero mean.3 O The estimator of d. that minimizes J D E[. Example 2.210) d D pT R 1 x . x N ]T .205) and (2. : : : . : : : . þ N / dÞ px1 :::x N .206) it is easy to prove the following theorem (see page 130). x N D þ N / dÞ 1 (2. c N ]T .v.
coincides with the linear function of the observations (2.v.s are assumed to have zero mean. let d be the desired vector.212) (2.s. respectively. MMSE linear estimation for random vectors We extend the results of the previous section to the case of complexvalued r.215) (2.210) and (2.214) (2.216) (2.v.219) Deﬁnition 2. we note that.207) and (2.s. rd2 x .s. consisting of the N ð M matrix C.3 The linear minimum meansquare error (LMMSE) estimator.v. x2 .218) Rx D E[xŁ xT ] Rd D E[d d ] The problem is to determine a linear transformation of x. rd M x ] Ł T H Rdx D E[dŁ xT ] D Rxd (2. d2 . and for a desired vector signal.219) that minimizes the cost function J D E[jjd O djj2 ] D M X mD1 E[jdm O dm j2 ] (2. and of the M ð 1 vector b. Observation 2. xT D [x1 .213) (2. : : : .211) with (2. Let x be an observation.v. modeled as a vector of N r.217) (2.b (2.220) In other words.221) . copt D R and the corresponding meansquare error is 2 Jmin D ¦d 1 p 1 pT R p (2. the optimum coefﬁcients C and b are the solution of the following problem: min J C.v.162 Chapter 2. dT D [d1 . d M ] We introduce the following correlation matrices: rdi x D E[di xŁ ] Rxd D E[x d ] D [rd1 x . linear estimation coincides with optimum MMSE estimation. modeled as a vector of M r. x N ] Moreover.211) Note that the r.3 Comparing (2. Ł T (2.s.208). in the case of jointly Gaussian r. : : : . : : : . given by O d D CT x C b O such that d is a close replica of d in the meansquare error sense. The Wiener ﬁlter and linear prediction In other words.
respectively.k d D [d.226) hence. Scalar case.A. d2 . and T O O d D d1 D c1 x D xT c1 (2. Therefore the columns of the matrix C. We determine now the expression of C and b in terms of the correlation matrices introduced above.224) Q Q CT xjj2 ] C jjbjj2 Q being E[d] D E[x] D 0. cm . The (2. : : : . In fact J D E[jjd D E[jjd D E[jjd Q CT x Q CT xjj2 ] Q bjj2 ] 2RefE[. Rx cm D rdm x m D 1.221) leads to M O O O scalar problems.221) leads again to the Wiener ﬁlter. we observe that if d and x have zero mean. satisfy equations of the type (2. The estimation problem 163 We note that in the formulation of Section 2. : : : . Q Q Q since the choice of b D b 6D 0 implies an estimator CT x C b with a larger value of the cost function.229) (2. it results in C D Rx 1 Rxd Thus.d Q Q Q CT x/ H b]g C jjbjj2 (2.224) implies that the choice b D 0 yields the minimum value of J . the solution is given by Rx c1 D rd1 x where rd1 x is deﬁned by (2. d M . : : : . we will assume that both x and d are zero mean random vectors.Rx 1 Rxd /T x (2.220) operates on single components. the optimum estimator in the LMMSE sense is given by O d D . d and d are Mdimensional vectors.1 we have xT D [x. d D d1 .223) that is M D 1.k/] T N C 1/] (2. then b D 0. For M D 1. Nevertheless.225) with c1 column vector with N coefﬁcients. First of all. the vector problem (2. Without loss of generality. x. O Vector case.k/. since the function (2. For M > 1. In this case the problem (2.228) .215).2. each with input x and output d1 . M (2.226). based on the deﬁnition (2.222) (2.227) (2.214). and the matrix C becomes a column vector.
232).231) O d (2. J D tr[Re ] Substituting (2. The Wiener ﬁlter and linear prediction Value of the cost function. yields Jmin D tr[Rd Rdx Rx 1 Rxd ] (2.228) in (2.230) the cost function (2.233) (2.220) is given by the trace of Re .164 Chapter 2.232) . On the basis of the estimation error eDd with correlation matrix Re D E[eŁ eT ] D Rd Rdx C C H Rxd C C H Rx C (2.
k/. we want to determine the coefﬁcients of a FIR ﬁlter having input x. Depending on the application.15)). c1 .4) 1 2 Two estimation methods are presented in Section 1.k/ (3.1.k The output signal is given by y. .k/ (3.1) 1/. In general the coefﬁcients may vary with time.1. : : : .2) ci . Adopting.k/ y. we develop iterative algorithms with low computational complexity to obtain an approximation of the Wiener solution. we obtain the estimation error3 e. for example. The ﬁlter structure at instant k is illustrated in Figure 3. the optimum solution requires solving a system of equations with a computational complexity that is at least proportional to the square of the number of ﬁlter coefﬁcients. For the analysis of IIR adaptive ﬁlters we refer the reader to [1. : : : . Input vector at instant k: xT .k N C 1/] (3.k/. it is required that the autocorrelation matrix R of the ﬁlter input vector. We deﬁne: 1. x.k i/ D xT .k/x. Given two random processes x and d. and the crosscorrelation p between the desired output and the input vector be known (see (2. Coefﬁcient vector at instant k: cT . c N 2. x.1 Moreover.k/ with the desired response d.Chapter 3 Adaptive transversal ﬁlters We reconsider the Wiener ﬁlter introduced in Section 2.3) Comparing y. the meansquare error criterion.k/ D d. Estimating these correlations is usually difﬁcult.k/ D [c0 .k/] (3.k/.k/.k/ D N 1 X i D0 1 . Some caution is therefore necessary in using the equations of an adaptive ﬁlter.2. 3 In this chapter the deﬁnition of the estimation error is given as the difference between the desired signal and the ﬁlter output. the estimation error may be deﬁned using the opposite sign.k/ D [x.k/c. 2]. so that the ﬁlter output y is a replica as accurate as possible of the process d.11. In this chapter. We will consider transversal FIR ﬁlters2 with N coefﬁcients.
3. Depending on the cost function associated with fe.15).k/ D copt . Adaptive transversal ﬁlters x(k) Tc x(k1) Tc x(k2) Tc x(kN+1) c0 (k) c1 (k) c 2(k) c N1 (k) + y(k) e(k) + d(k) Figure 3.1. however.k/j2 ] (3.6) where R and p are deﬁned respectively in (2.5) Assuming that x and d are individually and jointly WSS. 3. least squares (LS). In the following sections we will present iterative algorithms for each of the two classes. J . analogously to (2. Structure of an adaptive transversal ﬁlter at instant k.1 Adaptive transversal ﬁlter: MSE criterion The cost function.17).k/g. 2.166 Chapter 3.k/ D ¦d c H .k/p p H c. it requires that R and p be known. given by (2. The optimum Wiener–Hopf solution is c.1 Steepest descent or gradient algorithm Our ﬁrst step is to realize a deterministic iterative procedure to compute copt . .k/Rc.1. to minimize is J .k/ can be written as 2 J . in Chapter 2 two classes of algorithms have been developed: 1.k/ C c H . or functional. meansquare error (MSE).40).k/ (3. We will see that this method avoids the computation of the inverse R 1 . The corresponding minimum value of J .34).16) and (2.k/ D E[je.k/ is Jmin . where copt is given by (2.
k/ is orthogonal to the locus of points with constant J that includes c.Rc.k/ D 2rx .k C 1/ D c.Rc.c0 .0/.k/ 1 2 ¼rc.11) copt.9) p/ (3.k/ p/ (3.c0 . from (2.k/ ¼.2. Behavior of J and sign of the gradient vector rc in the scalar case (N D 1).k C 1/ D c.18)).k C 1/ D c0 .0 / (3.8) In the scalar case (N D 1).10) The iterative algorithm is given by: c0 .0 c0(1) c0(0) Figure 3. J Jmin 0 c opt.1.3.k/ J .k/ is a quadratic function of the vector of coefﬁcients.k/ and rc0 J .k/ J .k/ copt.31) we ﬁnd rc.0 /2 (3. and k is the iteration index.k/ with respect to c (see (2.k/ ¼rx . Adaptive transversal ﬁlter: MSE criterion 167 The steepest descent or gradient algorithm is deﬁned as: c. a realvalued positive constant.k/ @c0 copt.k/ is illustrated in Figure 3.k/ D 2.3.0 / (3.0/. ¼ is the adaptation gain. In the twodimensional case (N D 2). 0<∆ 0=∆ c0 0>∆ .0/.k/ denotes the gradient of J .2. As J .12) The behavior of J and the sign of rc0 J .k/ (3. in general not necessarily coinciding with time instants.k/ as a function of c0 is illustrated in Figure 3. for realvalued signals the above relations become: J .c0 .7) where rc.k/ D @ J .k/ hence c.k/ J . We recall that in general the gradient vector for c D c.k/. the trajectory of rc J .k/ D Jmin C rx .
369). where U is the unitary matrix formed of eigenvectors of R.k C 1/ to 0.k C 1/ D [I ¼ ]ν.k/ equation (3.0/ D c. Using the decomposition (1.k/ Rcopt ] (3. Adaptive transversal ﬁlters Figure 3. we determine now the conditions for the convergence of c. and setting (see (2.46)) ν.k/ (3. N .14) ¼[Rc.16) . Stability of the steepest descent algorithm Substituting for p the expression given by (2.k/ from (3.32). and is the diagonal matrix of eigenvalues f½i g.0/.k/ to copt or.15) copt (3. equivalently. i D 1.168 Chapter 3. the iterative algorithm (3.13) we obtain 1c.k/ D c.0/ copt .3.13) ¼R]c.k C 1/ D [I D [I ¼R]c. for the convergence of 1c.k/ D [I Deﬁning the coefﬁcient error vector as 1c. Loci of points with constant J and trajectory of rc J in the twodimensional case (N D 2).17) (3. : : : .k/ D U H 1c. that is with 1c.15) becomes ν. R D U U H .k/ C [¼R ¼R]1c.k C 1/ D c.k/ C ¼Rcopt Starting at k D 0 with arbitrary c.9) can be written as c.k/ I]copt (3.
N (3. ¹i .k/ i D 1. 1<1 ¼½i < 1 (3.k/ as a function of k is given by (see Figure 3. : : : .17) satisﬁes the difference equation: ¹i .360)).1. 2. the ith component of the vector ν.k/ converges. : : : . 2. equivalently.23) If ½max (½min ) is the largest (smallest) eigenvalue of R. In the case ¼½i > 1 and j1 ¼½i j < 1.21) !0 k!1 8¹i . Plot of ¹i .4): ¹i . observing (3.18) Hence.1 ¼½i /¹i . but it assumes alternating positive and negative values.22) ¼½i j < 1 (3.23) the convergence condition can be expressed as 0<¼< ν (k) i 2 ½max (3.k/ in (3.k/ D .3.k/ as a function of k for ¼½i < 1 and j1 ¼½i j < 1. that is ¹i . . Adaptive transversal ﬁlter: MSE criterion 169 Conditions for convergence As is diagonal. we have that the algorithm converges if and only if 0<¼< 2 ½i i D 1. ¹i .24) 0 1 2 3 4 5 6 k Figure 3.k C 1/ D .20) As ½i is positive (see (1. N (3.0/ (3.1 ¼½i /k ¹i .19) The ith component of the vector ν.4.k/ is still decreasing in magnitude.k/ if and only if j1 or.0/ (3.
170 Chapter 3.1 ¼½max / (3.16) and (3.27) where the approximation is valid if ¼½i − 1.1 ¼½i /k ¹i . N (3. observing (3. u2 . In (3. for each coefﬁcient it results in cn .25) D copt C ui .k/ D copt C Uν.23) is satisﬁed.28) then we need to determine min ¾. depends on the choice of ¼. each with the time constant4 −i D 1 ln j1 ¼½i j ' 1 ¼½i i D 1.n . n D 0.¼/ D max j1 i ¼½i j (3. Choice of the adaptation gain for fastest convergence The speed of convergence.n .k/ D copt C [u1 . : : : . converges to the optimum solution as a weighted sum of N exponentials.k/ (3. : : : .0/ n D 0. given by c. . If we let ¾.k/ D copt. : : : .26) where u i. N 1 (3. 2. each coefﬁcient cn .29) As illustrated in Figure 3.1 ¼½i /k ¹i .19) we obtain the expression of the vector of coefﬁcients.k/ D copt C N X i D1 N X i D1 ui ¹i .1 ¼½i /k ¹i .0/ Therefore.n C N X i D1 u i.¼/ ¼ (3. N 1. We deﬁne as ¼opt the value of ¼ that minimizes the time constant of the slowest mode.5.0/ characterizes the ith mode of convergence. which is related to the inverse of the convergence time. Adaptive transversal ﬁlters Correspondingly.26) the term u i. u N ]ν. the solution is obtained for 1 ¼½min D .30) 4 The time constant is the number of iterations needed to reduce the signal associated with the ith mode of a factor e.n is the nth component of ui . if the convergence condition (3. : : : . Note that.
6).k/j2 (3.k/ ! Jmin as a weighted sum of exponentials.k/ D Jmin C Now using (3. Plot of ¾ and j1 ¼½i j as a function of ¼ for different values of ½i : ½min < ½i < ½max . for k ! 1.34) Consequently.i D 1 2 ln j1 ¼½i j ' 1 2¼½i (3. if the condition for convergence is veriﬁed.19) we have J .k/ D Jmin C N X i D1 N X i D1 ½i j¹i .¼opt / D 1 2 ½max ½min ½min D D ½max C ½min ½max C ½min . and consequently the larger the eigenvalue spread the slower the convergence. We note that other values of ¼ (associated with ½max or ½min ) cause a slower mode.33) ½i .1. The ith mode will have a time constant given by −MSE.¼opt / is a monotonic function of the eigenvalue spread (see Figure 3.35) .µλ i  1 ξ(µ) 0 0 1 µopt 1 λmax λi 1 λmin µ Figure 3.0/ (3.3. Adaptive transversal ﬁlter: MSE criterion 171 1.R/ D ½max =½min is the eigenvalue spread (1.47) the general relation holds J . J . from which we get ¼opt D and ¾. Transient behavior of the MSE From (2.31) where .5. We emphasize that ¾.R/ C 1 (3.1 ¼½i /2k ¹i2 .32) ½max 2 C ½min (3.376).R/ 1 .
4). let us examine the twodimensional case (N D 2). We note that (3. Choosing c. Case 1 for ½1 − ½2 . choosing ¼ D ¼opt the algorithm exhibits monotonic convergence along u½min .34) is different from (3. modes associated with small eigenvalues tend to weigh less in the convergence of J .¼opt / as a function of the eigenvalue spread . If no further information is given regarding the initial condition c. Recalling the observation that J .R/ D ½max =½min .2 0 0 1 2 3 4 5 χ(R) 6 7 8 9 10 Figure 3.6 0. assuming ¼½i − 1.0/ on the ¹2 axis (in correspondence of ½max ) we have the following situations: 8 >< 1 > > > ½max > > > > < 1 if ¼ D > ½max > > > > > > >> 1 : ½max the iterative algorithm has a nonoscillatory behavior the iterative algorithm converges in one iteration the iterative algorithm has a trajectory that oscillates around the minimum (3. and an oscillatory behavior around the minimum along u½max .26) because of the presence of ½i as weight of the iith mode: consequently. In particular.172 Chapter 3. .k/ increases more rapidly (slowly) in the direction of the eigenvector corresponding to ½ D ½max (½ D ½min ) (see Figure 2.k/.4 0. we have the following two cases.8 ξ(µopt) 0.0/.36) Let u½min and u½max be the eigenvectors corresponding to ½min and ½max . respectively. Adaptive transversal ﬁlters 1 0. ¾.6.
k/ D d.k/xŁ .k/ J .k/xT .39) (3.k/e.k/ 2xŁ .k/ D xŁ .k/] (3.k/xŁ .k/ J .k C 1/ D c.1.k/ and O p.k/ C ¼e.41) 1 2 ¼rc.k C 1/ D c.k/xŁ .40) Observation 3.8).k/[d. but in general it also exhibits a large variance.1.k/c. Actually.k/ J .1 The same equation is obtained for a cost function equal to je.8) is thus estimated to be5 rc.k/ that is c. . becomes c. 5 We note that (3. where k now denotes a given instant.k/ C 2xŁ .k/ The gradient vector (3.38) (3. Adaptive transversal ﬁlter: MSE criterion 173 Case 2 for ½2 D ½1 .3.k/ (3.0/.k/xŁ . whose gradient is given by rc. with reference to the following parameters and equations. independently of the initial condition c. Choosing ¼ D 1=½max the algorithm converges in one iteration.42) Implementation The block diagram for the implementation of the LMS algorithm is shown in Figure 3.k/ (3.k/xT .k/ (3.2 The least meansquare (LMS) algorithm The LMS or stochastic gradient algorithm is an algorithm with low computational complexity that provides an approximation to the optimum Wiener–Hopf solution without requiring knowledge of R and p. the following instantaneous estimates are used: O R.k/ D D D 2d.k/ xT .k/j2 .k/ 2xŁ .k/ D 2e. 3.k/c.7.37) The equation for the adaptation of the ﬁlter coefﬁcients.39) represents an unbiased estimate of the gradient (3.
k/.k/xŁ .45) y.7 are used to memorize coefﬁcients. Parameters. Block diagram of an adaptive transversal ﬁlter adapted according to the LMS algorithm.46) (3. Required parameters are: 1.k/ C ¼e.7. 2.k/c. 0 < ¼ < Filter. we set c. number of coefﬁcients of the ﬁlter.k/ (3.k/ D d. 2 . 1.k C 1/ D c. which are updated by the current value of ¼e. Adaptive transversal ﬁlters x(k) Tc * x(k1) Tc * x(k2) Tc * c N1 (k) ACC x(kN+1) * c0 (k) ACC c1 (k) ACC c2 (k) ACC e(k) µ d(k) y(k) + Figure 3.43) Adaptation.0/ D 0 (3. Estimation error e.k/xŁ .k/ D xT .k/ (3.k/ Initialization. N .174 Chapter 3.44) The accumulators (ACC) in Figure 3. If no a priori information is available. Coefﬁcient vector adaptation c.k/ 2. statistical power of input vector The ﬁlter output is given by y. .
k/ d.k/x Q .k/x I .k/ c Q .49) (3.k/ C jd Q .k/c I .k/ D e I .k/ Estimation error e. we derive the new equations: y I .k/c Q .k/ C j y Q .k/ C jx Q .k/ Q (3. E[c.k/ ¼[e I .k/c Q . Canonical model The LMS algorithm operates with complexvalued signals and coefﬁcients. Adaptive transversal ﬁlter: MSE criterion 175 Computational complexity For every iteration we have 2N C 1 complex multiplication (N due to ﬁltering and N C 1 to adaptation) and 2N complex additions.k/ C xT .k/ I Q c I .1.k/ D d I .k/ I e I .k/ D xT .58) (3.k/ e Q .59) !0 .k/ C e Q .50) (3.47) (3.k/ C ¼[e I . Conditions for convergence Recalling that the objective of the LMS algorithm is to approximate the Wiener–Hopf solution.k C 1/ D c I .k/ D x I .56) (3. Convergence of the mean.k/ (3.k/ e Q .55) (3. Input vector Desired signal x.52) (3.k/ D d Q .k/ D d I .k/ C jc Q .k/ Output ﬁlter y.k/] E[e.k/] k!1 k!1 ! copt (3.48) (3.51) Coefﬁcient vector c. We can rewrite complexvalued quantities as follows. This scheme is adopted in practice if only processors that use real arithmetic are available.54) (3.k/ D c I .k/ C je Q .k/c I .k/ Using the above deﬁnitions and considering separately real and imaginary terms.k C 1/ D c Q .k/x Q .k/ D xT .k/] xT .N /.k/ D y I .k/ y Q .k/] Therefore a complexvalued LMS algorithm is equivalent to a set of two realvalued LMS algorithms with crosscoupling.57) y Q .3.k/ y I . we introduce two criteria for convergence.53) (3.k/x I . Therefore the LMS algorithm has a complexity of O.
61) . e(k)2 (dB) J(k)=E[ e(k)2 ] −2 −4 −6 −8 −10 −12 J 0 min 50 100 150 k Figure 3.5 −1 0 E[c(k)] −a 50 k 100 150 2 0 J(k) . and ¦x D 1 (i. For c. For a ﬁrstorder predictor we adapt the coefﬁcient.k/x. we compute 1. Adaptive transversal ﬁlters 2 x(k) 0 −2 0 50 k 100 150 0 c(k) −0.8 we illustrate the results of a simple experiment for an input x given by a realvalued AR(1) random process: Á 2 (3.k/ 3.k/g.k/ D d.62) 1/ (3.k/ w.k 1/ C w.k/j2 g for a onecoefﬁcient predictor (N D 1).e. Predictor output y. in Figure 3.k 2. coefﬁcient fc.63) y.176 Chapter 3. Prediction error e. ¦w D 0:097).k/g and squared error fje.k/ 2 N 0. To show the weakness of this criterion.8. adapted according to the LMS algorithm.k/x. Coefﬁcient update c. ¦w 2 2 where a D 0:95.k C 1/ D c.0/ D 0 and k ½ 0. according to the LMS algorithm with ¼ D 0:1.k/.k/ D a x.k/ C ¼e. c.k 1/ (3. Realizations of input fx.k/ D x. it is required that the mean of the iterative solution converges to the Wiener– Hopf solution and the mean of the estimation error approaches zero.k/ y. In other words.k/ D c.60) x.k/ (3.
By itself.k/copt (3. copt and Jmin represent the Wiener–Hopf solution. Optimum ﬁlter output error emin . because the iterative solution c may exhibit very large oscillations around the optimum solution.1/ In other words. estimated by averaging over 500 realizations. Choosing a small ¼ the adaptation will be slow and the effect of the gradient noise on the coefﬁcients will be strongly attenuated. Convergence in the meansquare sense. for which they may be considered uncorrelated. xT .65) J . 1.64) constant (3.k/xŁ .k/ copt jj2 ] ! constant k!1 k!1 (3.k/g and fje. A constraint on the amplitude of the oscillations must be introduced.k/ 2.1.k/j2 ].k/j2 ] ! J .1/=Jmin can be made small by choosing a small adaptation gain ¼. c follows mean statistical parameters associated with the process x and not the process itself. statistically independent.3. From the plots in Figure 3. fx.k/.1.66) . It is interesting to observe that this hypothesis corresponds to assuming the ﬁlter input vectors.k/j2 g are illustrated. Actually. such as the steepestdescent algorithm.3 Convergence analysis of the LMS algorithm We recall the following deﬁnitions. We note that the coefﬁcients are obtained by averaging in time the quantity ¼e. E[jjc. The random processes x and c exhibit a completely different behavior. In any case. however. realizations of the processes fx. 2. both the mean of the coefﬁcient error vector norm and the output meansquare error must be ﬁnite. Adaptive transversal ﬁlter: MSE criterion 177 In Figure 3.k/g.67) copt (3.8 we observe two facts: 1. for small values of ¼. Convergence of the mean is an easily reachable objective. we will see that the ratio Jex .k/ We also make the following assumptions. it does not yield the desired results.1/ is the MSE in excess and represents the price paid for using a random adaptation algorithm for the coefﬁcients rather than a deterministic one. fc. Coefﬁcient error vector 1c. at convergence.k/ D c.1/ Jmin D Jex .k/ D d.k/ D E[je. as well as mean values E[c.k/ D E[je. The quantity J .k/g.k/] and J . 3.8.
k/ D [¹1 . Fourthorder moments can be expressed as products of secondorder moments (see (3.33).k 2/. c.70) ¼xŁ . N (3.c (3.k/j2 ] x.178 Chapter 3.16)). The adaptation equation of the LMS algorithm (3.k/ (see (3.k/]]E[1c.70) and exploiting the statistical independence between x.k/ D E [je.k/ is statistically independent of x.368)) and 1c.k/] (3.k/ D [I xT .k/1c.71) can be written as J .k C 1/ D 1c.k/xŁ .k/ xT . x.k/] (3.k/xT .k/ (see (1. we get E[1c. ν. Adaptive transversal ﬁlters 1. n D 1.k/xT .41) can thus be written as 1c.k/j2 ] (3.c E denotes the expectation with respect to x and c. with the change of variables (3. : : : .k/] D U H 1c.k/]1c.k/xŁ . observing (3. .73) As E[xŁ .k/xT .k/ C ¼emin .k/ (see (3.k C 1/] D [I ¼R]E[1c.74) 6 x.68) This assumption is justiﬁed by the observation that the linear transformation that orthogonalizes both x.k/c. the cost function6 J . 2.k/] C ¼E[emin .36) of the optimum ﬁlter.k/ C ¼xŁ . : : : Moreover.16). The components of the coefﬁcient error vector.97)). : : : .k/] D R and the second term on the righthand side of (3.72) Convergence of the mean Taking the expectation of (3.k/[d. transformed according to U H .k/ D Jmin C N X i D1 ½i E[j¹i .k/ depends only on the terms x. 3.k/ C ¼xŁ .k/ We note that 1c. we obtain the same equation as in the case of the steepest descent algorithm: E[1c.k/] D 0 i 6D n i.k/[emin .16)) in the gradient algorithm is given by U H .k/] (3.k/] (3.k/.k/ and 1c.k/copt to the terms within parentheses we obtain 1c.69) Adding and subtracting xT .k/.73) vanishes for the orthogonality property (2.k/.k C 1/] D [I ¼E[xŁ . ¹ N .k 1/. are orthogonal: Ł E[¹i .k C 1/ D 1c.k/¹n .
k/ (3.k/]Jmin (3.9.k/]E[1c2 .32). the vector E[1c.k/ allows us to deduce by a simple analysis important properties of the convergence in the meansquare sense.k/ and 1c.1. we get E[1c2 . The conclusions of the analysis x are similar.76) because x.0//E[1c2 .k/] becomes rapidly negligible with respect to the meansquare error during the process of convergence.25) and (3. 0<¼< Convergence in the meansquare sense (real scalar case) The assumption of a ﬁlter with realvalued input and only one coefﬁcient c.1/] D D ' ¼2 rx .81) 2 rx . E[x 4 . Then for the convergence of the difference equation (3.77) becomes x E[1c2 .k/ are statistically independent.k/x.3. Consequently ¼ must satisfy the condition 0<¼< Moreover.1 ¼x 2 .78) it must be j j < 1.0// .1 C ¼2 r2 .2 ¼ Jmin ¼rx .k/ are assumed to be statistically independent and x.k C 1/] D E[1 C ¼2 x 4 . We can therefore assume that E[1c.0/.80) 2¼rx .k/] D 3r2 . and choosing the value of ¼ D 2=.0/ − 1.0/ (3. we get E[1c2 .½max C ½min /. From 1c.k/ is orthogonal to emin .0/Jmin (3. as 1c.79) whose behavior as a function of ¼ is given in Figure 3.k/¼x.k//1c.k/ has zero mean and is statistically independent of all other terms.k C 1/ D .k/] where the last term vanishes.k/x 2 . Assuming7 moreover.k/] is reduced at each iteration at least by the factor .78) ¼ Jmin 2 7 In other texts the Gaussian assumption is made.77) ¼x 2 .k/ C 2E[. whereby E[x 4 .0/ (3.0/.1 2¼x 2 .k/] C ¼2 rx .75) ½max Observing (3.k/] D E[x 2 . and assuming furthermore that x.0/ Jmin ¼rx .k/] D r2 .k C 1/] D . (3.k//1c.k/. for the LMS algorithm the convergence of the mean is obtained if 2 (3.0/ x Let D 1 C ¼2 r2 .k/] C ¼2 E[x 2 .½max C ½min /.k/emin .k/ and emin .2 ¼rx .½max ½min / = .k/ C ¼emin .0/.0/ x 2¼rx . assuming ¼rx . .0// (3. Adaptive transversal ﬁlter: MSE criterion 179 Consequently.
Likewise.k/U H xŁ . or misadjustment.4 0.k/ (3.70) becomes ν. is: MSD D J .83) that is J .0/E[1c2 .9. (3. Adaptive transversal ﬁlters 1 0.k/ 1c.1/ ¼ D D rx . from e.k/xT .180 Chapter 3.84) Convergence in the meansquare sense (general case) The convergence theory given here follows the method developed in [3].k/ D d. we have J .k/ (3.85) (3.k/c.87) .1/ ' Jmin C rx .k/ D emin .0/ The relative MSE deviation.k/ it turns out 2 E[e2 .1/ Jmin Jex . With the change of variables (3.k/] x. Plot of as a function of ¼.86) ¼ Jmin 2 (3.82) (3.k/] C E[x 2 .k/]E[1c2 .16).k/x.0/ Jmin Jmin 2 (3.k C 1/ D [I ¼U H xŁ .k/U]ν.k/ C ¼emin .6 γ 0.k/] D E[emin .8 0.k/] In particular.k/ D Jmin C rx .2 0 0 1/ r (0) x µ 2/ r (0) x Figure 3. for k ! 1.
k/]T D UT x.88) Recalling Assumption 1 of the convergence analysis. n/ D xiŁ .k/Q Ł . n D 1.89) (3.k/j2 ]] (3. i/ of the matrix expressed by (3.k/ independent Q of x.k/ x (3.k/.91) i.1.368) we get E[ ] D hence the components fxi .k C 1/ν T .k/[.I ν ¼ T T /] (3.k/ C ¼2 E[jxi . the elements with indices .90).k/xT .I ¼ T /] D E [.k/g are mutually orthogonal.91). n/ .I ¼ Ł /ν Ł .i.k C 1/] D E[.k/xn .k/ν T .3.90) (3. i/Án .k/j2 ].k C 1/ D [I ¼ ]ν.k/Q T .93) Observing (3.k/.I ¼ T /]E[ν . Adaptive transversal ﬁlter: MSE criterion 181 Let us deﬁne Q x.92) (3.k/ Q Q and Q D xŁ .i.94) are given by " # N X Ł E Ái .k/ Q Q 2¼½i Ái .I x ¼ ¼ Ł /E ]ν Ł .k/ 2¼½i Ái .87) becomes Q ν. with elements on the main diagonal given by the vector η T . we ﬁnd that the matrix E[ν Ł . E[j¹ N .k/ν T .i.k/ (3.k/ Q Q From (1.k/.k/.k/] Ł Recalling Assumption 2 of the convergence analysis.k/ and x.k/ and x. considering that ν. x N .96) D Ái .k/ν .k/U x N ð N matrix with elements .k/ ν T . and assuming emin . Then (3.i. Q Moreover.95) Observing (3. : : : .93) is equal to ¼2 Jmin . the correlation matrix of ν Ł at the instant k C 1 can be expressed as E[ν Ł .k/] D [E[j¹1 .k/ D [x1 .k/.I ¼ T /] C ¼2 Jmin E[ Ł ] (3.k/ is statistically independent of x.k/ C ¼2 .I Ł /.k/ν T . i/Ái . and consequently emin . the second term on the righthand side of (3.k/ C ¼2 N X nD1 ½i ½n Án . : : : .n.k/ nD1 N X nD1 D Ái . N (3.k/ C ¼emin . the ﬁrst term can be written as x. : : : .k/ are not only orthogonal but also statistically independent.k/j2 ]Án .k/ D U H xŁ . Á N .I ¼ Ł /ν Ł . : : : .k/] is diagonal.94) D E[.k/ .k/j2 jxn .k/ D [Á1 .ν E [.k/.k/ 2¼ .
0/ 2 ¼N rx .0/ according to (3.104) 1 Ki D viH η. and .103) the constants fKi g are determined by the initial conditions Ã Â ¼Jmin i D 1.95) and (3. and V is the unitary matrix formed by the eigenvectors fvi g of B.k/j2 ] D ½i Q (3.182 Chapter 3.96). we obtain the relation η.0/ k½0 (3.106) .98) N ð N symmetric positive deﬁnite matrix with positive elements.102) the solution of the vector difference equation (3. the cost function J .98). : : : .175) becomes B D V diag.k/ given by (3.93) and (3.13).k/ D Jmin C λT η.k/ (3. : : : .0/ D tr[R] D N X i D1 ½i (3.1 ¼½1 /2 ¼2 ½1 ½2 : : : ¼2 ½1 ½ N 6 ¼2 ½2 ½1 . : : : .95) and (3.16): H Án .1 ¼½2 /2 : : : ¼2 ½2 ½ N 6 BD6 : : : :: : : : 4 : : : : 2½ ½ 2½ ½ ¼ N 1 ¼ N 2 : : : . Adaptive transversal ﬁlters where. the general decomposition (2.1 ¼½ N /2 2 3 7 7 7 5 (3.k/ D N X i D1 Ki ¦ik vi C 2 ¼Jmin 1 ¼N rx .k C 1/ D Bη. ½ N ] be the vector of eigenvalues of R. After simple steps.97) Let λT D [½1 .25) from (3.0/ D E[j¹n .0/ where the components of η. N (3. 1]T . 1.k/j2 ]E[jxi . In (3.103) where 1 D [1.100) is given by: η.0/j2 ] D E[jun 1c.72) becomes J .k/j4 ] D E[jxi .101) (3. recalling Assumption 3 of the convergence analysis.99) (3. From (3. N (3. ¦ N /V H (3. : : : .k/ C ¼2 Jmin λ Using the properties of B.¦1 . 2 Q Q E[jxi .0/ depend on the choice of c. similar to those applied to get (3.100) where f¦i g denote the eigenvalues of B. and using the relation N rx . : : : .0/j2 ] n D 1.105) Using (3.
0/ D N 1 X i D0 (3.k/ D where Ci D Ki λT vi (3. . 1.110) becomes (3. from (3. Basic results From the above convergence analysis.112) i/j ] 2 E[jx. we will obtain some fundamental properties of the LMS algorithm.109) In fact the adaptive system is stable and J converges to a constant steadystate value under the conditions j¦i j < 1. 2.k D statistical power of input vector the equation (3.106).99) the sum of the elements of the ith row of B satisﬁes N X nD1 [B]i.109).1. The transient behavior of J does not exhibit oscillations.n D 1 ¼½i . This happens if 0<¼< 2 N rx .110).0/ (3.2 ¼N rx .0/ (3. whereas the second term gives the steadystate value.111) A matrix with these properties and whose elements are all positive has eigenvalues with absolute value less than one. i D 1.107) The ﬁrst term on the righthand side of (3. The LMS algorithm converges if the adaptation gain ¼ satisﬁes the condition 0<¼< 2 statistical power of input vector (3. : : : . N . being N X i D1 ½i D tr[R] D N rx . In particular.103) in (3. this result is obtained by observing the properties of the eigenvalues of B. if ¼ satisﬁes (3.107) describes the convergence behavior of the meansquare error. we ﬁnd J . Adaptive transversal ﬁlter: MSE criterion 183 Substituting the result (3.108) N X i D1 Ci ¦ik C 2 2 Jmin ¼N rx . Therefore further investigation of the properties of the matrix B will allow us to characterize the transient behavior of J .110) Conversely.0// < 1 (3.3.
184 Chapter 3. 3. rather than on the eigenvalue distribution of the matrix R.0//. : : : .0/ ¼N rx . for convergence of the mean.117) Observations 1.N rx . As λT vi D 0.i D 1.0/ 2 (3.108) we get Ci D 0.1/ D 2 2 Jmin ¼N rx . 0/.0/ ' N rx .0. 0. from (3. [B]i.1/ Jmin D (3.110) can be intuitively explained noting that.0/ Jmin 2 ¼N rx . vi. convergence in the meansquare imposes a tighter bound to allowable values of the adaptation gain ¼ than that imposed by convergence of the mean (3.0.107) reveals a simple relation between the adaptation gain ¼ and the value J . a large time constant of one of the modes implies a low probability that the corresponding term contributes signiﬁcantly to the meansquare error. The new bound depends on the number of coefﬁcients. 0. it must be 0<¼< but since N X i D1 2 ½max (3.113). However. a small eigenvalue of the matrix R (½i ! 0) determines a large time constant for one of the convergence modes of J . If ½i D 0. As shown below. an increase in the number of coefﬁcients causes an increase in the excess meansquare error due to ﬂuctuations of the coefﬁcients around the mean value. . In other words. as ¦i ! 1. for a given value of ¼.i D 1.116) and the misadjustment has the expression MSD D for ¼ − 2=. 0. 0/. : : : . Jex . Increasing the number of coefﬁcients without reducing the value of ¼ leads to instability of the adaptive system. : : : .113) ½i > ½max (3.0/ D Jmin 2 ¼N rx . : : : . Consequently ¦i D 1 and viT D . Proof. Adaptive transversal ﬁlters We recall that. The relation (3. For ¼ ! 0 all eigenvalues of B tend toward 1.1/ ¼ ¼N rx .0/ (3. the ith row of B becomes .115) from which the excess MSE is given by Jex .114) the condition for convergence in the meansquare implies convergence of the mean.1/ D J . 0. Equation (3.k/ in the steady state (k ! 1): J . 2.
k/ × Jmin is normally veriﬁed at the beginning of the iteration process. the Perron–Frobenius theorem afﬁrms that the maximum eigenvalue of a positive matrix B is a positive real number and that the elements of the corresponding eigenvector are positive real numbers [4]. If all eigenvalues of the matrix R are equal. for example. we obtain J . the other eigenvectors of B are orthogonal to λ.1.k/ (3.0/.0/ (3.8 ½i D rx .0//]J .122) 1 N rx . ½i D rx .118) The remaining eigenvalues of B do not inﬂuence the transient behavior of J . i 6D i max . Moreover.3. 1] is the corresponding eigenvector. mitigates this effect. : : : .0/ J . if the input x is white noise.k/ Jmin 1 N rx .118) and considering a time varying adaptation gain ¼. Therefore the convergence of J is less inﬂuenced by the eigenvalue spread of R than would be the convergence of 1c.121) 8 This occurs.2 ¼N rx . Adaptive transversal ﬁlter: MSE criterion 185 It is generally correct to state that a large eigenvalue spread of R determines a slow convergence of J to the steady state.0// (3.k C 1/ ' 1 1 N Ã J . combining (3.k/ D J . 3.0/.k/ C 2¼. the fact that modes with a large time constant usually contribute to J less than the modes that converge more rapidly. : : : .k/rx .0/. i 6D i max . Hence Ci D 0. 4.k/.119) The maximum convergence rate of J is obtained for the adaptation gain ¼opt .0/Jmin (3. It is easily veriﬁed that ¦imax is an eigenvalue of B and viT D N 1=2 max [1. it follows that ¦imax is the maximum eigenvalue of B. : : : . N . N .2 ¼. Proof. it results ¼opt .120) As the condition J .k/ ' and Â J . If all eigenvalues of the matrix R are equal.k C 1/ ' [1 ¼. i D 1. 1.k/rx . Moreover. However.k/.107) with the (3. the maximum eigenvalue of the matrix B is given by ¦imax D 1 ¼rx . Since all elements of vimax are positive.0/.k/N rx . i D 1. since Ci D 0.k/ (3. because vimax is parallel to λT . .
however.1/. In this case. 6. Therefore. We now give a brief introduction to other versions that can be used for various applications. Jex . instead. for time varying systems ¼ must be chosen as a compromise between Jex and J` and cannot be arbitrarily small [5.124) 3.1/ is determined by the large eigenvalues of R. Thus (3. whereas the speed of convergence of E[c. and J` depends on the ability of the LMS algorithm to track the system variations and expresses the lag error in the estimate of the coefﬁcients.123) where Jmin .1/.4 Other versions of the LMS algorithm In the previous section. The relatively slow convergence is inﬂuenced by ¼.0/ statistical power of input vector (3.k/ corresponds to the Wiener–Hopf solution. In particular it must be 0<¼< 2 2 D N rx .k/ C Jex . Jex . recalling Assumption 2 of the convergence analysis. 7]. . The LMS algorithm is easy to implement.103) indicates that at steady state all elements of η become equal. 2. 6.k/ is less sensitive to this parameter. Consequently. In case the LMS algorithm is used to estimate the coefﬁcients of a system that slowly changes in time. the adaptation is fast at the expense of a large Jex . If the eigenvalue spread of R increases. the adaptation gain ¼ has a lower bound larger than 0. that the convergence behavior depends on the initial condition c. The mean corresponds to the optimum vector copt . the number of coefﬁcients and the eigenvalues of R. the convergence of E[c. Adaptive transversal ﬁlters We note that the number of iterations required to reduce the value of J . the value of J “in steady state” varies with time and is given by the sum of three terms: Jtot . Final remarks 1.k/ by one order of magnitude is approximately 2:3N .k/] is imposed by ½min . in steady state the ﬁlter coefﬁcients are uncorrelated random variables with equal variance.k/ D Jmin . It turns out that J` is inversely proportional to ¼.1/ depends instead on the LMS algorithm and is directly proportional to ¼.0/ [8].186 Chapter 3.1. and in a small excess MSE at convergence Jex . 5. Choosing a small ¼ results in a slow adaptation.1/ C J` (3. 4. on the other hand the convergence of J . the basic version of the LMS algorithm was presented. 3.k/] becomes slower. Note. For a large ¼.
as usual.129) .x.R C ÞI/ 1 p (3. Therefore the leaky LMS algorithm is used in cases where the Wiener problem is illconditioned. : : : .k// D [sgn.x I . It is usually sufﬁcient to choose Þ two or three orders of magnitude smaller than rx .125) with 0 < Þ − rx .k// sgn.k N C1//] (3. for the same J .k//xŁ .0/.126) In other words.128) It is interesting to give another interpretation to what has been stated.x Q . Both approaches are useful to make irreversible an illconditioned matrix R.k/ sgn.3.k//. or to accelerate the convergence of the LMS algorithm. at the expense of a lower speed of convergence.130) > : sgn.1 ¼Þ/c.k/x.k/ D E[je.k/ D E[je.k C 1/ D c. In steady state we get k!1 lim E[c.131) 9 The sign of a vector of complexvalued elements is deﬁned as follows: sgn.k C 1/ D .k// Note that the ﬁrst version has as objective the minimization of the cost function J .k/j] (3.k/j2 ] C Þ E[jjc.k// (3.x I .x Q .k/ C ¼e. Three versions are:9 8 > sgn. e.128). the cost function includes an additional term proportional to the norm of the vector of coefﬁcients.xŁ .k/jj2 ] where.127) (3. Observing (3.k/ D d. the application of the leaky LMS algorithm results in the addition of a small constant Þ to the terms on the main diagonal of the correlation matrix of the input process.k/] D .k/ C ¼ e.e. one obtains the same result by summing white noise with statistical power Þ to the input process.k N C1// C j sgn.1/.k/ (3. This equation corresponds to the following cost function: J . sgn. in order not to modify substantially the original Wiener–Hopf solution.k// C j sgn.xŁ .k/xŁ .e. Sign algorithm There are adaptation equations that are simpler to implement. and multiple solutions exist.k/ (3.k/ < c.0/.1. Adaptive transversal ﬁlter: MSE criterion 187 Leaky LMS The leaky LMS algorithm is a variant of the LMS algorithm that uses the following adaptation equation: c.k/ cT .
8 > > < per a < per A (3. Sigmoidal function for various values of the parameter þ.k//xŁ .10) [9]: '.8 −1 −1 −0.133) where þ is a positive parameter.8 −0.6 0.k/ C ¼ e.a/ D tanh Â þa 2 Ã D 1 e 1Ce þa þa (3.2 0.2 β =6 β =12 β =24 β =48 ϕ (a) 0 −0.4 −0.k// '.6 0.134) AÄaÄ A per a > A 1 0.a/ is the sigmoidal function (see Figure 3.8 0.4 0.2 −0. .a/ D > A > : 1 where A is a positive parameter.6 −0.k/'.x where '.8 1 Figure 3.128).2 0 a 0.10. There also exists a piecewise linear version of the sigmoidal function deﬁned as 1 a '.k/ < c.xŁ .6 −0.188 Chapter 3.e.k//'.k// (3.k C 1/ D c.4 −0. Adaptive transversal ﬁlters Sigmoidal algorithm As extensions of the algorithms given in (3. the following adaptation equations may be considered: 8 > '.132) > : Ł .e.4 0.
Two values of ¼.137) N 1 X i D0 ¼ Q O p C Mx . In (3.1 a/jx. alternatively. if some x. Initially a large value of ¼ is chosen for fast convergence.138) where 0 < a < 1.k/ (3.k 1/ C .0//.136) O where Mx .135).k/.k i/j2 (3.k/j2 (3.k/ D jjxjj2 D or. 1.k/ D a Mx .k/ D N Mx .3.k/ (3.135) jx. b.139) for a ' 1. the adaptation algorithm is affected by strong noise in the gradient. A simple estimate is obtained by the iterative equation (see (1.k/ is the estimate of the statistical power of x.468)): O O Mx . O O Mx . To be able to apply the normalized algorithm.N rx . This problem can be overcome by choosing an adaptation gain ¼ of the type: ¼D where 0 < ¼ < 2.1/.k/ assume large values. with time constant given by −D 1 1 ' ln a 1 a (3. . and Q O Mx . in order to assign the values of Mx and p so that the adaptation process does not become unstable. Adaptive transversal ﬁlter: MSE criterion 189 Normalized LMS In the LMS algorithm. ¼ D 1=. for example. a. Variable adaptation gain In the following variants of the LMS algorithms the coefﬁcient ¼ varies with time. for uncorrelated as well as correlated input signals [10].140) The normalized LMS algorithm has a speed of convergence that is potentially higher than the standard algorithm. however. p is a positive parameter that is introduced to avoid the denominator becoming too small.1. Subsequently ¼ is reduced to achieve a smaller J . typically p' 1 Mx 10 (3. some knowledge of the input process is necessary.
Decreasing ¼. ¼ N 1 ].k/ C Þ2 je. The following expression of ¼ is used: ¼.130) is given by ¼. : : : . Let µT D [¼0 . µ1 J(k) µ2 = µ1 / 2 Jmin 0 K 1 k Figure 3. 4. two approaches are possible.k/. Vector of values of ¼. Adaptive transversal ﬁlters For a choice of ¼ of the type ( ¼D ¼1 ¼2 per 0 Ä k Ä K 1 per k ½ K 1 (3.142) 3.190 Chapter 3. For a timeinvariant system.k/ obtained by using two values of ¼. Typical values are Þ1 ' 1 and Þ2 − 1.11. .141) the behavior of J will be illustrated in Figure 3.k/j2 (3.k C 1/ D Þ1 ¼. ¼ proportional to e. the adaptation gain usually selected for application with the sign algorithm (3. ¼max ].143) with ¼ limited to the range [¼min .11. 2. a. Initially larger values ¼i are chosen in correspondence of those coefﬁcients ci that have larger amplitude.k/ D ¼1 ¼2 C k k½0 (3. Behavior of J.
z/ are given by %e š'0 .3.k/ D a1 x.k/Þ never changed sign in the last m 1 iterations (3. c2 ]. 3.147) Â a1 2% Ã D 2:28 rad (3.145) with w additive white Gaussian noise (AWGN). as illustrated in Figure 3. ¼i changes with time following the rule 8 if the ith component of the gradient has al> ¼i . however.83) we expect to ﬁnd in steady state c' a (3.k C 1/ D > > if the ith component of the gradient has : ¼i . is not as simple as for transversal ﬁlters.12.148) 2 Being rx . although they were popular in the past when fast hardware implementations were rather costly.552) we ﬁnd that the statistical power of w is given by 2 ¦w D 1 a2 [.0/ D ¦x D 1.1. For the study of the LMS algorithm for lattice ﬁlters we refer the reader to [11.1.149) We construct a predictor for x of order N D 2 with coefﬁcients cT D [c1 .547).144) with ¼ limited to the range [¼min . Adaptive transversal ﬁlter: MSE criterion 191 b. Typical values are m 0 . For this reason such ﬁlters are now rarely used.k/ (3. LMS for lattice ﬁlters We saw in Section 2.k/ 1 > < ways changed sign in the last m 0 iterations Þ ¼i . 12]. the roots of A. From (2.146) From (1. and a1 D 1:3 a2 D 0:995 (3.150) .k 1/ a2 x. from the (1. ¼max ]. described by the equation x. where p % D a2 D 0:997 and '0 D cos 1 (3. 3g and Þ D 2.1 C a2 /2 1 C a2 a2 ] D 0:0057 D 1 22:4 dB (3. m 1 2 f1.5 Example of application: the predictor We consider a real AR(2) process of unit power.k 2/ C w. The application of the LMS algorithm for lattice ﬁlters.1 that ﬁlters with a lattice structure have some interesting properties.2. using the LMS algorithm and some of its variants [13].
k/ (3. that is c1 ' a1 .k/ D x. Example 3.k/ C ¼e.k/ D cT .2 (Leaky LMS) The equation for updating the coefﬁcient vector is c. Adaptive transversal ﬁlters Figure 3. c2 ' 2 2 a2 .12.k/ x. thus giving a more accurate solution. Example 3. Predictor of order N D 2.15 for a single realization and for the mean (estimated over 500 realizations) of the coefﬁcients and of the squared prediction error. but the convergence time increases. In any case. the predictor output is given by y.k 1/ (3.k with prediction error 1/ 1/ C c2 . and ¦e ' ¦w .k C 1/ D .192 Chapter 3.k C 1/ D c.153) Convergence curves are plotted in Figure 3.k 1/ (3. the excess error decreases. In Figure 3.152) For the predictor of Figure 3. for ¼ D 0:04.k 2/ (3.k/ y.k/x. by decreasing ¼.1. We observe that.1.12 we now consider various versions of the adaptive LMS algorithm and their relative performance.154) Convergence curves are plotted in Figure 3.k/x. .1 ¼Þ/c.14 a comparison is made between the curves of convergence of the mean for three values of ¼.k/x.k D c1 .1 (Standard LMS) The equation for updating the coefﬁcient vector is c.k/x.13 for a single realization and for the mean (estimated over 500 realizations) of the coefﬁcients and of the squared prediction error.151) e.k/ C ¼e.
2 1.3.04 µ =00. Comparison among curves of convergence of the mean obtained by the standard LMS algorithm for three values of ¼. −a 0. Convergence curves for the predictor of order N D 2. .2 0 −0.04 µ =0.01 J(k) (dB) −10 −15 σ2 −20 w −25 0 100 200 300 400 500 600 700 800 900 1000 k Figure 3.13. Adaptive transversal ﬁlter: MSE criterion 193 Figure 3.2 1 0 1 0.01 µ =0.4 −0.6 −0.8 µ =0.1 µ =0.1 c1(k) 0. obtained by the standard LMS algorithm.1.1 µ =0.04 µ =0.8 −a −1 2 200 400 600 800 1000 0 200 400 600 800 1000 k k 0 −5 µ =0.2 µ =0.14.6 0.2 0 c2(k) −0.4 0.1 −0.
k/ D a ¦x .194 Chapter 3.155) ¦x .159) (3. Convergence curves for the predictor of order N D 2.1 a/jx.15. 1/ D 1 [jx.k The adaptation gain ¼ is of the type ¼.157) (3. for ¼ D 0:04 and Þ D 0:01.k/ O2 (3. Example 3. We note that the steadystate values are worse than in the previous case. obtained by the leaky LMS.k/x. Adaptive transversal ﬁlters Figure 3.k O2 O2 1/ C .1.3 (Normalized LMS ) The equation for updating the coefﬁcient vector is c.k/e.160) .156) 1/ (3. 1/j2 C jx. 2/j2 ] O2 2 with aD1 and pD 1 E[jjxjj2 ] D 0:2 10 2 5 D 0:97 (3.158) ¼ Q p C N ¦x .k/ C ¼.k/ D where ¦x .k/j2 k½0 (3.k C 1/ D c.
We note that.k/ sgn. A direct comparison of the convergence curves obtained in the previous examples is given in Figure 3. the convergence is Q considerably faster.k C 1/ D c. Convergence curves for the predictor of order N D 2.1. the value of ¼ could be further lowered. yields fastest convergence. To decrease the prediction error in steady state for versions (1) and (3).3.1.k/ C ¼ sgn. Version (3).k/ C ¼ sgn.e.x. Adaptive transversal ﬁlter: MSE criterion 195 Figure 3.16 for a single realization and for the mean (estimated over 500 realizations) of the coefﬁcients and of the squared prediction error. obtained by the normalized LMS algorithm.e. for ¼ D 0:04.k (2) c.4 (Sign LMS algorithm) We consider the three versions of the sign LMS algorithm: (1) c.k//x. 1//. . 1//. yields the best performance in steady state.k C 1/ D c.17.k/ C ¼e.k// sgn.18 for the three versions of the sign LMS algorithm. It turns out that version (2). for ¼ D 0:08.k 1/.k A comparison of convergence curves is given in Figure 3.16. (3) c. at the expense of reducing the speed of convergence. with respect to the standard LMS algorithm. Convergence curves are plotted in Figure 3. where the estimation error in the adaptation equation is not quantized. Example 3. however.x.k C 1/ D c.
6 −0. obtained by three versions of the LMS algorithm.2 0.3 ver.2 0 c2(k) −0. −a1 1.18.4 −0.2 0 1 ver.2 ver.2 0.3 −5 J(k) (dB) ver.1 −10 ver. Comparison of convergence curves for the predictor of order N D 2. .1 ver.2 −15 σ2 −20 w −25 0 100 200 300 400 500 600 700 800 900 1000 k Figure 3.196 Chapter 3. Comparison of convergence curves obtained by three versions of the sign LMS algorithm.2 −0.6 0.3 c1(k) 0.1 ver.2 0 −0.8 ver.4 0.8 −a −1 2 200 400 600 800 1000 0 200 400 600 800 1000 k k 0 ver. Adaptive transversal ﬁlters Figure 3.17.
i/ D [x.2.k/.19.161) (3. obtained at the expense of a larger computational complexity. named recursive least squares (RLS) algorithm. c N 1 . The recursive least squares (RLS) algorithm 197 Observation 3. : : : .i/ D cT . obtained for the vector of coefﬁcients c. Filter output signal at instant i. The RLS algorithm is characterized by a speed of convergence that can be one order of magnitude faster than the LMS algorithm. With reference to the system illustrated in Figure 3. In this case it is necessary to adopt a method that ensures the stability of the error prediction ﬁlter. if the order of the predictor is greater than the required minimum.k/ x(i) Tc x(i1) Tc x(i2) Tc x(iN+1) (3. Reference system for a RLS adaptive algorithm. 3.i/ D xT . Input vector at instant i xT .3. : : : .k/x. the correlation matrix result is illconditioned with a large eigenvalue spread. for an AR process x.19.2 As observed on page 97.k/ D [c0 . x.k/. Coefﬁcient vector at instant k cT .k/] 1/.k/ y. we introduce the following quantities: 1.i/. Thus the convergence of the LMS prediction algorithm can be extremely slow and can lead to a solution quite different from the Yule–Walker solution. c1 .162) 3. x.2 The recursive least squares (RLS) algorithm We now consider a recursive algorithm to estimate the vector of coefﬁcients c by an LS method.i 2.163) c0 (k) c1 (k) c2 (k) cN1 (k) + y(i) e(i) + d(i) Figure 3. . such as the leaky LMS.i/c.i N C 1/] (3.
k/ (3. that enables proper ﬁltering operations even with nonstationary signals or slowly timevarying systems.i/c.k/ D From (3. Adaptive transversal ﬁlters 4.k/ k X i D1 ½k i d.k/ satisﬁes the normal equation . the optimum value of c.i/ (3. ž ½ is a forgetting factor. applied to a sequence of prewindowed samples with the exponential weighting factor ½k . 2.k/ D ϑ .k/c. Deﬁning E. the solution is given by c.i/ (3. based on the observation of the sequences fx.k/.k/ is the minimum sum of squared errors up to instant k.169) ϑ .k/ k X i D1 ½k i je.k/ D 1 .168).i/ At instant k.164) the criterion for the optimization of the vector of coefﬁcients c.198 Chapter 3.i/ D d.i/j2 (3. k (3.i/g i D 1.k/ D we want to ﬁnd min E.171) .k/ where . The memory of the algorithm is approximately 1=. Normal equation Using the gradient method.166) (3.167) where the error signal is e.165) (3. if 1 .i/xT . Desired output at instant i d.i/ Two observations arise: xT .i/xŁ . ž This problem is the classical LS problem (2. : : : .k/ D k X i D1 (3.k/ϑ .k/ c.1 ½/.170) exists.168) ½k i xŁ .i/g fd.128).
Then A For AD .k/½ 1 1/xŁ . Therefore we seek a recursive algorithm for k D 1.k 1/ C d.k/ (3.175) where A.k/ can be written recursively.k 1 1 .k/ B 1 1 DB BC.k and similarly ϑ .k/ D ½ 1 P.k/ DD1 (3.177) the equation (3.k/ D it follows that .k 1/ ½ 1 1 .k/ 1/xŁ . B and D are positive deﬁnite matrices.k/xT . : : : .k 1 xT .k/ D and kŁ .k/ (3.173) k 1 X i D1 ½k i xŁ .k 1/ (3.i/ C xŁ .k/ D ½ 1C½ 1 1 .k/½ 1 .k/ D ½ .k/ may be too hard.176) D ½ .k/xT .k/ 1 .k/xT . Both expressions of .174) 1/ C xŁ .i/xT . especially if N is large.k 1 .3.k 1/xŁ .k/ and ϑ .k/P.k 1/ C D xŁ .178) we have the recursive relation P.172) We now recall the following identity known as matrix inversion lemma [12]. The recursive least squares (RLS) algorithm 199 Derivation of the RLS algorithm To solve the normal equation by the inversion of .2.178) We introduce two quantities: P.k/ (3.k/ D ½ϑ . From .D C C H BC/ 1 CH B (3. Let ADB 1 C CD 1 CH (3.179) 1/xŁ .k 1/ ½ 1 Ł k .k/ D ½ 1 1 .k/xT .k/xŁ .180) also called the Kalman vector gain.k/ (3. From (3.k 1/ 1 C xT .k/ (3.181) . 2.176) becomes 1 .k/ (3.
182) P.k P.184) has been used.k/c.k/ Using the (3. In any case the relation holds c.k/[1 C ½ from which we get kŁ .k/ P. the RLS algorithm consists of four equations: kŁ .k/ xT .k 1/]ϑ .k/xT . that is computed before updating c. In other words.k 1/ C kŁ .186) kŁ .k/ c.k/ D c.k 1/ ½ 1/ 1 Ł k .k 1/xŁ .k/kŁ .k/ in the ﬁrst term.k/] D ½ 1 1 .190) (3.k/ ½ C xT .k/ž.k/ D P. Adaptive transversal ﬁlters We derive now a simpler expression for kŁ .183) 1 k .k/d.k/ D P. it follows kŁ .k/P.k/ xT .k/ D ½ 1 1/ C ž.184) .k/P.k xT .193) ž.k/ϑ . Deﬁning the a priori estimation error. we get P.200 Chapter 3.k 1/] 1/ C P.k/ is an approximated value of e.k/ (3.k/c.k 1 Substituting the recursive expression for P.k/ (3.k 1/]xŁ .181).k/[d.k/x .k/xT .k/ D d.k/ D 1 1 1 T x .k/ (3.k 1/ϑ .k/ϑ .k/c.k/ Ł D P.k 1/ (3.k/ (3.k/d.k 1/xŁ .k/ D ½P.k/ xT .k 1/ϑ .k 1/ (3.k/c.k/xŁ .187) we note that xT .k/ (3.k/xT .k 1/xŁ .180) we obtain kŁ .k/ 1/ C P.k/ where in the last step (3.k/xT .k/ D d.k/P.k D c.k/ (3.k/ 1/ ½ ½ 1 Ł 1 Ł k .k/P.k/c.k 1/xŁ . the recursive equation to update the estimate of c is given by c.k/ D c.k/ϑ .k/.k/ D ½[½ D P.k 1/ ½ 1 Ł k . From (3.k P.k 1/ C kŁ .k/xT .k/ (3.189) In summary.k/P.k/xŁ .k/ 1 .k c.k/d.k .k 1/xŁ .192) 1/ (3.k/xŁ .185) 1/ C P.k/ D ½ D [½ Using (3.k/ D d.k/P.174). from the a posteriori estimation error e.188) we could say that ž.k 1/xŁ .191) (3. ž.k/.k 1/ is the ﬁlter output at instant k obtained by using the old coefﬁcient estimate.
Consequently P.k 1/ C ž.k/ P. Initialization For k D 1.k/] H D π T .k/ in (3.k 1/xŁ .k/ (inverse of the Hermitian matrix . : : : c.k 1/xŁ .k/ ½Cx kŁ .3. hence xT .0/ is the statistical power of the input signal.k/ D d.k// .i/xT .k/c. In Table 3.2. The term xT .194) ½k i xŁ . The recursive least squares (RLS) algorithm 201 In (3. k.k/ D P. Initialization of the RLS algorithm We need to assign a value to P.k/ (3.k/ ž. We modify the deﬁnition of .k/ D c.k/π T .k/.1 we give a version of the RLS algorithm that exploits the fact that P.k/ D so that .k/ D ½ 1 .k/P.k/) is Hermitian. N C 1/ D .0/ D Ž 1 I 1/xŁ .k/ may be interpreted as the energy of the ﬁltered input.k 1/xŁ .k π Ł .0/ D 0 P.k/ 1 r.0/.k 1/ c.1 RLS algorithm.k/π Ł .k/ is the input vector ﬁltered by P.195) k X i D1 .k/kŁ .197) where rx .k 1/ kŁ .0/ (3.0/ D ŽI (3.i/ C Ž½k I with Ž − 1 This is equivalent to having for k Ä 0 an all zero input with the exception of x.k/ D T .k/ P.k/ D r.k/ xT .P. 2.k 1/ and normalized by the ½ C xT .k/π Ł .k 1/ D [P.198) Table 3.k/P.0/ (3.190).½ N C1 Ž/1=2 .196) D 100 rx .0/ D Ž Typically Ž 1 1 I Ž − rx .
k/ D x H .k/xT .171) it follows that ϑ H .k/e.184) and (3. the recursive relation is given by Emin .k/ is a real scalar value.k/ c H .179).k ½ϑ H .k/ž.202 Chapter 3.k/ž.k/j2 (3.k//Ł ] (3.k/ ϑ H .k/kŁ .k 1/ C d.k/ D[ 1 .k/ D k X i D1 ½k i jd.k/ .174) and (3.k/[d Ł .k/ .204) (3.184) we obtain 1 Using the expression (3.k/ž.203) Finally. and recalling that ϑ H .k/ 1/ C ž.k/c.202) .k d Ł .k/c. (3.k/ D ½Emin .k 1/ 1/ ϑ H .k/ D Ed .k/xŁ .k/ Moreover from (3.k D ½Emin .k/] (3.k/xŁ .k/d Ł .k/ observing (3.xT .k/ (3.k/kŁ .k/d Ł .k/ž.k/kŁ .k/ D c H .k 1/ C jd. we get ž.k/eŁ .k/ (3.k/kŁ .i/j2 D ½Ed . as Emin .150).199) From the general LS expression (2.k/ that is ž.k 1/ C d Ł .k/cŁ .k/ D ½Emin .k/c.205) .k 1/c.200) [½ϑ H .k 1/ C d Ł .206) 1/ C ž.201) becomes Emin .k/ D ϑ H .k/ D ž Ł .k/ϑ .k/ Then (3.k/ D ½Ed .k/kŁ . Emin .k/ is real.k/ is Hermitian.192) we get Emin .k/ D ½Emin . Adaptive transversal ﬁlters Recursive form of E min We set Ed .k We note that.k/eŁ .k/] H xŁ .k/j2 1/ C xT .k/xŁ .k/][c.k/eŁ .k/ž.201) ϑ H . from (3.k D ½Ed .k/ (3.k/ 1/ C ž.k 1/ C jd.
the algorithm is capable of “tracking” the changes. ž In any case. the computational complexity of the RLS algorithm.20 for a single realization and for the mean (estimated over 500 realizations) of the coefﬁcients and of the squared estimation error.k/. in fact the RLS algorithm converges in a number of iterations of the order of N . For ½ < 1 and 1=. ž For k ! 1 there is no excess error and the misadjustment MSD is zero.K (3. is given by CCRLS D 2N 2 C 4N For a number of . when ½ < 1 the “memory” of the algorithm is approximately 1=. expressed as the number of complex multiplications per output sample. convergence curves for the RLS algorithm are plotted in Figure 3. This is true for ½ D 1. the direct method is more convenient. Example of application: the predictor With reference to the AR(2) process considered in Section 3. ž The RLS algorithm converges in the meansquare sense in about 2N iterations.208) N C 1/ output samples.3.2. It provides an estimate of the coefﬁcients at each step and not only at the end of the data sequence. . We note that a different scale is used for the abscissa as compared to the LMS method. In any case the RLS solution has other advantages: 1. It can be numerically more stable than the direct method. independently of the eigenvalue spread of R. ž On the other hand the RLS algorithm for ½ < 1 can be used for tracking slowly timevarying systems. for ½ D 1. 3. if K × N . The recursive least squares (RLS) algorithm 203 Convergence of the RLS algorithm We make some remarks on the convergence of the RLS algorithm. 2. the direct method (3.171) requires instead CCDIR D N 2 C N C K N3 N C1 (3.1.209) We note that.1 ½/ much less than the time interval it takes for the input samples to change statistics.5. Computational complexity of the RLS algorithm Exploiting the symmetry of P.207) MSD D 1C½ ž From the above observation it follows that the RLS algorithm for ½ < 1 gives origin to noisy estimates.1 ½/ and 1 ½ N (3.
The fast Kalman algorithm has the same speed of convergence as the RLS. Therefore we will list a few fast algorithms. Adaptive transversal ﬁlters Figure 3. strong and weak points similar to those already discussed in the case of the LMS algorithm for lattice structures [12. Falconer and Ljung [14] have shown that the recursive equation (3. As a consequence the fast algorithms may become numerically unstable. with their fast transversal ﬁlter (FTF). obtained by the RLS algorithm. 16]. 3.193) requires only 10. Convergence curves for the predictor of order N D 2.3 Fast recursive algorithms As observed in the previous section. Algorithms for lattice ﬁlters.k/. whose computational complexity increases linearly with N .20. the RLS algorithm has the disadvantage of requiring . There are versions of the RLS algorithm for lattice structures that in the literature are called recursive least squares lattice (LSL) that have. in addition to a lower computational complexity than the standard RLS form.2N C 1/ multiplications. Ciofﬁ and Kailath [15]. Algorithms for transversal ﬁlters. but with a computational complexity comparable to that of the LMS algorithm.204 Chapter 3. 2. have further reduced the number of multiplications to 7. 1. . The implementation of these algorithms still remains relatively simple. their weak point resides in the sensitivity of the operations to round off errors in the various coefﬁcients and signals. Exploiting some properties of the correlation matrix .2N 2 C 4N / multiplications per iteration.2N C 1/. the number of dimensions of the coefﬁcient vector c.
For further study on the subject we refer the reader to [17. The name comes from the use of an orthogonal triangularization process. 23]. Regarding the computational complexity per output sample. Block adaptive algorithms in the frequency domain 205 Table 3. for example from the time to the frequency domain.1 Comparison of the various algorithms In practice the choice of an algorithm must be made bearing in mind some fundamental aspects: ž computational complexity. A particular structure is the QR decompositionbased LSL. ž robustness. which does not require the a priori knowledge of the ﬁlter order and is suitable for implementation in very largescale integration (VLSI) technology. this approach may exhibit: a) lower computational complexity. 20.2 Comparison of three adaptive algorithms in terms of computational complexity. Algorithms for ﬁlters based on systolic structures. that is good performance achieved in the presence of a large eigenvalue spread and ﬁniteprecision arithmetic [5.3. ž a very efﬁcient and modular structure.2. RLS and FTF is given in Table 3. therefore it is rarely used. and tracking capabilities under nonstationary conditions. before adaptive ﬁltering.4 Block adaptive algorithms in the frequency domain In this section some algorithms are examined that transform the input signal. error in steady state. a brief comparison among LMS.4.2N C 1/ 3. usually known as QR decomposition. cost function algorithm multiplications MSE LS LMS RLS FTF 2N C 1 2N 2 C 7N C 5 7. Although the FTF method exhibits a lower computational complexity than the RLS method. 22].2N C 1/ divisions 0 N2 C 4N C 3 4 additions subtractions 2N 2N 2 C 6N C 4 6. ž performance in terms of speed of convergence. that leads to a systolictype structure with the following characteristics: ž high speed of convergence. its implementation is rather laborious. 19. 3.3. With respect to the LMS algorithm. owing to the QR decomposition and lattice structure. 3. ž numerical stability. 18. 21. or b) improved .
N 1. i D 0. Each input block is transformed using the DFT (see Section 1.n N / i D 0. As for real data the complexity of an N point FFT in Figure 3.210) In the following. the LMS adaptation algorithm is expressed as: Ci .n N /g and fYi . lower case letters will be used to indicate sequences in the time domain. : : : .n N / D Di . 1. : : : . Deﬁning E i . the method operates over blocks of N samples. where n is an integer number.n N / Yi .1 Block LMS algorithm in the frequency domain: the basic scheme The basic scheme includes a ﬁlter that performs the equivalent operation of a circular convolution in the frequency domain. The algorithm requires three N point FFTs and 2N complex multiplications to update fCi g and compute fYi g. : : : . the DFT of the desired output and of the adaptive ﬁlter output.n N /g. N 1. Computational complexity of the block LMS algorithm via FFT We consider the computational complexity of the scheme of Figure 3.n N / C ¼E i .4).206 Chapter 3. Adaptive transversal ﬁlters convergence properties of the adaptive process.21.n C 1/N / D Ci . Adaptive transversal ﬁlter in the frequency domain. We will ﬁrst consider some adaptive algorithms in the frequency domain that offer some advantages from the standpoint of computational complexity [24.n N /X iŁ ..21 for N sample real input vectors. We indicate with fDi . 1. 1.4. N 1 (3. The instant at which a block is processed is k D n N . 3.n N /g.n N /. i D 0. . As illustrated in Figure 3. 25. while upper case letters will denote sequences in the frequency domain.21. 26. respectively. The samples of the transformed sequence are denoted by fX i . 27].
However.214) A comparison between the computational complexity of the LMS algorithm via FFT and the standard LMS algorithm is given in Table 3.3.213) CCLMS f D 4 4 2 We note that the complexity in terms of real multiplications per output sample of the standard LMS algorithm is CCLMSt D 2N C 1 ' 2N (3.2 Block LMS algorithm in the frequency domain: the FLMS algorithm We consider a block LMS adaptive algorithm in the time domain.15 0.212) log2 CCLMS f D 3 4 2 using the fact that fYi g and fCi g. the direct application of the scheme of Figure 3.41 0.015 terms of complex multiplications is given by N point FFT of N real samples D N N point FFT + 2 2 Â Ã N N N N D C log2 4 2 2 2 (3. 3. Let us deﬁne: . i D 0. Block adaptive algorithms in the frequency domain 207 Table 3. As each complex multiplication requires four real multiplications. 1.4. : : : . N 16 64 1024 CCLMS f =CCLMSt 0. N 1 are Hermitian sequences. We note that the advantage of the LMS algorithm via FFT is non negligible even for small values of N . the complexity in terms of real multiplications per output sample becomes Ã Â N 3 log2 C1 (3.211) then the algorithm requires a number of complex multiplications per output sample equal to Ã Â N 1 C1 (3.4.3 Comparison between the computational complexity of the LMS algorithm via FFT and the standard LMS for various values of the ﬁlter length N. as the product between DFTs of two time sequences is equivalent to a circular convolution.21 is appropriate only if the relation between y and x is a circular convolution rather than a linear convolution.3. for blocks of N input samples.
n N C i/ D cT .n N C N }  {z block n and Y0 . we deﬁne10 C0 T .n N /. 1. : : : .n N C i/ i D 0.n N / D X0 . The above equations can be efﬁciently implemented in the frequency domain by the overlapsave technique (see (1.n N /. where for example L D 2N .n N / D DFT[cT .n N C i/ 4. : : : . . n N C N 3 7 7 7 D last N elements of DFT 5 1. : : : .218) (3. 0]  {z } N zeros (3.216) The equation for updating the coefﬁcients according to the block LMS algorithm is given by c. Assuming Lpoint blocks.n N C N D n N .n N /.112)).219) As in the case of the standard LMS algorithm. : : : .n N C i/xŁ . x. x. c1 . ﬁlter output signal at instant n N C i y.n N / D diag DFT[x.n N / D [c0 . 0.k N C 1/] (3.n N /x. is given by (3.215) (3.n N /] (3.n N /C0 . ∇.221) ² X0 .n N C i/ D d. x.220) ¦ 1/] } (3.208 Chapter 3. coefﬁcient vector at instant n N cT .k 2.n N / D 6 : : 4 : y. input vector at instant k xT . the updating term is the estimate of the gradient at instant n N .222) 1 [Y0 .n N C i/ y..n N C 1/ 6 y.n N C i/ (3. x. c N 3. : : : .n N /.217) 1 .n C 1/N / D c.n N / C ¼ N 1 X i D0 e.n N / 6 y. x. : : : .n N / then the ﬁlter output at instants k 2 y.n N /] 1/. error at instant n N C i e.k/ D [x. Adaptive transversal ﬁlters 1. : : : N 1 (3.223) 1/ 10 The superscript 0 denotes a vector of 2N elements.n N /.n N  N /. n N C 1.k/.n N {z block n 1 1/.
n N /C0 .n N /D[0T . [∇. : : : .n N / D DFT[0.n N /D N ðN N ðN F 1 [X0 .231) The implementation of the FLMS algorithm is illustrated in Figure 3. N 1 (3. In summary.n N / C ¼F N ðN N ðN F 0 N ðN 0 N ðN E0 .n N /. k/.n N C N {z 1/ y. : : : . the adaptation equation (3. and F the 2N ð 2N DFT matrix. 0.224) This component is given by the correlation between the error sequence fe. then the following equations deﬁne the fast LMS (FLMS): d0 T . .n N /] (3.k/g and input fx.n N /.n N /] 0 N ðN I N ðN y0 . Block adaptive algorithms in the frequency domain 209 We give now the equations to update the coefﬁcients in the frequency domain.229) (3. d. d. the complexity in terms of real multiplications per output sample is given by CCFLMS D 10 log2 N C 8 (3.226) In the frequency domain.22. Let us consider the mth component of the gradient.3.n N / C ¼DFT 0 (3.n N / D ﬁrst N elements of DFT 1 [X0 Ł . if 0 N ðN is the N ð N all zero matrix. Let E0 T . d. : : : .n N /E0 ..230) 1 [X0 Ł . For real input samples. Computational complexity of the FLMS algorithm For N output samples we have to evaluate ﬁve 2N point FFTs and 4N complex multiplications.n N C N 1/] Ä ½ 0 0 y0 .n N /DF[d0 . : : : .232) A comparison between the computational complexity of the FLMS algorithm and the standard LMS is given in Table 3.n N C N errors in block n 1/] } (3.k/ and x Ł . d.228) (3. referring to the scheme in Figure 3.22.k/g.n N /E0 .n N /] ½ Ä I 0 C0 . I N ðN the N ð N identity matrix.n N /] (3. 1.227) where 0 is the null vector with N elements..n N / (3.225) then ∇.n N /]m D N 1 X i D0 e.4. which is also equal to the convolution between e.4.n N / C0 .n C1/N /DC0 .219) becomes Ä ½ ∇.n C 1/N / D C0 .n N /  {z }  N zeros y.n N C i m/ m D 0.n N C i/x Ł .
Adaptive transversal ﬁlters Figure 3.22. .210 Chapter 3. Implementation of the FLMS algorithm.
.n C 1/N /] D E[c.5 LMS algorithm in a transformed domain We consider now some adaptive algorithms in the frequency domain that offer some advantages in terms of speed of convergence [28]. as usual. where ½max is the maximum eigenvalue of R. N 16 32 64 1024 CCFLMS =CCLMS 1.234) for 0 < ¼ < 2=. The FLMS algorithm converges in the mean to the same solution of the LMS.k/xŁ .N ½max /..05 Convergence in the mean of the coefﬁcients for the FLMS algorithm Observing (3.k/xT . R D E[xŁ .k/].235) samples 1 D ¼½i equal to that of the LMS algorithm. and taking the expectation of both members of the adaptation equation (3.0/ 2 2 3.1..k/] and p D rdx D E[d. we get E[c.n N /]/ (3. Recalling the analysis of the convergence of the steepestdescent algorithm of Section 3. For ¼ − 2=N ½max .218).n C 1/N /] D R 1 p (3.233) ¼N R/E[c.n N /] C ¼N p where.53 0.p D . LMS algorithm in a transformed domain 211 Table 3. 2. however.n N /] C ¼N .236) MSD D tr[R] D N rx .3.I R E[c.217) and (3. ¼ must be smaller by a factor N in order to guarantee stability.85 0. 3. From these equations we can conclude: 1.5. The time constant for the convergence of the ith mode (for ¼ − 1) is −i D 1 ¼½i N blocks (3.4 Computational complexity comparison between FLMS and LMS. it can be seen that the misadjustment is equal to that of the LMS algorithm: ¼ ¼ (3. we have n!1 lim E[c.1.5 0.219).
Adaptive transversal ﬁlters 3. . x. c1 .238) (3.k/. we deﬁne the following quantities.k/ D [z 0 .k/] 1 1 .241) Figure 3.k/] (3.23. General scheme for a LMS algorithm in a transformed domain.239) D GH (3.1 Basic scheme Referring to Figure 3.k/.k/ D [c0 . Input vector at instant k xT .5.k/ where G is a unitary matrix of rank N : G 3.237) with correlation matrix Rx D E[xŁ .k/ D Gx.212 Chapter 3. : : : . 1. z. : : : . x.k N C 1/] (3.23. : : : . c N 1 .k/ D [x. Coefﬁcient vector at instant k cT .k/.k 1/.240) (3.k/xT .k/.k/]. 2. z 1 . z N In general.k/. Transformed vector zT .
k/ 5. for a suitable choice of ¼.242) 6.k/ (3.k C 1/ D ci .k/ D d. Let N D diagfE[jz 0 .250) GŁ rdx 1 Rx 1 GŁ GŁ rdx D G H Rx 1 rdx D G H .g.k C 1/ D c. N 1 (3.k C 1/ D copt D Rz 1 rdz (3.245) i D 0. : : : ..k/ y.k/ Q We ﬁnd that.k/GT ] D GŁ Rx GT and rdz D E[d. Estimation error e. : : : .k/j2 ].244) can be written in vector notation as c.k/ D zT .k/c.k/ where ¼i D ¼ Q E[jz i . by considering a small window of input samples or recursively.k/] D E[GŁ xŁ .5.Rx 1 rdx / where Rx 1 rdx is the optimum Wiener solution without transformation.243) (3. E[jz N 2 1 . 1. e.k/ C ¼i e.GŁ Rx G/ DG 1 1 (3.k/ (3.k/ C ¼e.244) We note that each component of the adaptation gain vector has been normalized using the statistical power of the corresponding component of the transformed input vector. (3. Equation for updating the coefﬁcients.k/xŁ .k/j2 ].k/j ]g (3. LMS algorithm in a transformed domain 213 4.k/xT . E[jz 1 .k/ D cT .k/z iŁ . Q k!1 1 Ł N z .251) .246) Then (3.249) (3.k/zŁ .k/zT .k/j2 ] (3. LMS type: ci .248) where Rz D E[zŁ . The various powers can be estimated.247) lim c.k/z. Filter output signal y.3.k/] D GŁ rdx Then copt D .k/] D GŁ E[d.
3. 3. is used for the adaptation process. They reduce the number of computations to evaluate z.n N / are indeed uncorrelated. these two transformations.24.N log2 N /. recalling the deﬁnition (1.5. whiten the signal x by operating on the different subbands. If Rz is diagonal.3 LMS algorithm in the frequency domain In this case GDF N ð N DFT matrix. Consequently the adaptation gain ¼ is adjusted to the various modes. N 1 (3. used in lattice ﬁlters. 1. however. .k/ in (3. Consequently. Observation 3.214 Chapter 3. In both cases the computational complexity to evaluate the output sample y. even if more costly from the point of view of the computational complexity. in a simpler recursive form.k/ is O. z i .k/ D N 1 X mD0 (3. then the eigenvalue spread of N 1 Rz is equal to one. requires that the components of X0 . a transformation with these characteristics exhibits the best convergence properties.253) or.5.k m/e mi j2³ N i D 0.n N /] in (3.k Â 1/ exp j2³ i N Ã C x.k/ D z i .k N/ (3. implemented by either 1) FFT with parallel input or 2) recursively with serial input to implement equations (3.3 ž A ﬁlter bank can be more effective in separating the various subchannels in frequency. In this case the adaptation algorithm reduces to N independent scalar adaptation algorithms in the transformed domain.239) from O.N log2 N /. DFT and discrete cosine transform (DCT). Adaptive transversal ﬁlters On the speed of convergence The speed of convergence depends on the eigenvalue spread of the matrix Rz . and consequently is e difﬁcult to evaluate in real time. 2. Then z i . Lower triangular matrix transformation. with reduced spectral variations.254) The ﬁlters are of passband comb type.2 Normalized FLMS algorithm The convergence of the FLMS algorithm can be improved by dividing each component of the vector [X0 Ł .254).n N /E0 . The KLT depends on Rx . and the N modes of convergence do not inﬂuence each other. This procedure. 3. Moreover.231) by the power of the respective component of X0 . the resulting signal.k/ x. as illustrated in Figure 3. Common choices for G are the following: 1.376) of the eigenvalue spread.N 2 / to O. for the normalization N 1 . : : : . KarhunenLo` ve transform (KLT).252) x.n N /.
1. N 2 1 (3. ž There are versions of the algorithm where each output z i .24.4 LMS algorithm in the DCT domain The LMS algorithm in the DCT domain is obtained by ﬁltering the input by the ﬁlter bank of Figure 3.k/] D . respectively.k/ is decimated. with the aim of reducing the number of operations. the ﬁlter coefﬁcients satisfy the Hermitian property: ci . where the ith ﬁlter has impulse response and transfer function given by.5.k/ D cŁ N 1 i .k/ i D 0.k/g and fd.1 ³.k/g are realvalued signals.24.k/ D cos and G i .1 z 1 1 /.5.255) 3. 1. gi .2k C 1/i 2N k D 0. N 1 (3. : : : . ž If fx.256) .z/ D Z[g i . 1/i z ³Á z 2 cos N N / cos 1 ³ Á i 2N 2 (3.257) Cz .3. Adaptive ﬁlter in the frequency domain. : : : . LMS algorithm in a transformed domain 215 Figure 3.
: : : .259) Ignoring the gain factor cos. 29. Adaptive transversal ﬁlters Correspondingly. 2. we have p N 1 2X x. N 1 (3. 30].25. if all the signals are real. We note that.5 General observations ž Orthogonalization algorithms are useful if the input has a large eigenvalue spread and fast adaptation is required. 3. even the ﬁltering operation determined by G i . 3.k/ D N mD0 z i .³=2N /i/. System model in which we want to identify the relation between x and z.6.1 System identiﬁcation We want to determine the relation between the input x and the output z of the system illustrated in Figure 3. that can be included in the coefﬁcient ci . Figure 3. assumed statistically independent of x.216 Chapter 3.k/ D X 2 N 1 x. .k N mD0 m/ ³.. 3. ž In general. they require larger computational complexity than the standard LMS. ž If the signals exhibit timevarying statistical parameters.k z 0 . We note that the observation d is affected by additive noise w.6 Examples of application We give now some examples of applications of the algorithms investigated in this chapter [1. 2 having zero mean and variance ¦w . 25.5. usually these methods do not offer any advantage over the standard LMS algorithm.25. the scheme can be implemented by using real arithmetic.2m C 1/i 2N i D0 (3.z/ can be implemented recursively [12].258) m/ cos i D 1.
k D hT x.A).k/ and x.k/ D d.k/ D and the estimation error e. d.3.260) We analyze the speciﬁc case of an unknown linear FIR system whose impulse response has Nh coefﬁcients.k 1 .k/ C ¼e.k/ 1/ C Ð Ð Ð C h Nh 1 x.262) y. : : : . h 1 .6. h Nh In this case.264) For N ½ Nh . 11 Typically x is generated by repeating a PN sequence of length L > N (see Appendix 3.26.261) N 1 X i D0 ci .k/ (3.k/x. we introduce the vector h with N components. h . known to both systems.0/.k/ C w. Assuming N ½ Nh . hT D [h 0 .263) 1// C w. 0] (3.k/ .k/ (3. Examples of application 217 Linear case Assuming the system between z. and assuming the input x is white noise11 with statistical power rx .k/x.0/I (3. Adaptive scheme to estimate the impulse response of the unknown system.k C 1/ D c. Using an input x.k/xŁ . : : : . we get R D E[xŁ .k i/ D cT .26 can be adopted to estimate the ﬁlter impulse response. c.k/ D h 0 x. 0. the experiment illustrated in Figure 3.k/ C h 1 x.k/ The LMS adaptation equation follows.Nh (3.k/ can be modelled as a FIR ﬁlter.k/] D rx .265) Figure 3.k/ (3. we determine the output of the transversal ﬁlter c with N coefﬁcients y.k/xT .
h Nh 1] T (3.0/. the smaller ¼ must be so that c. h N C1 .k/ is orthogonal to x.k/. The larger the power of the noise.267) coincides with the ﬁrst N coefﬁcients of h. Adaptive transversal ﬁlters and p D E[d. the approximation xT .0/ jj h. : : : .1/ 6D 0.267) we see that the noise w does not affect the solution copt .k/ ' N rx .k/].k/ copt .k/.k/jj2 ] where E[jj c.0/h Then the Wiener–Hopf solution to the system identiﬁcation problem is given by copt D R and 2 Jmin D ¦w 1 (3.1/jj2 (3. Let be deﬁned as in (3.k/ D E[je.266) pDh (3.267) (3. h. h N .270) As the input x is white. the noise inﬂuences the convergence process and the solution obtained by the adaptive LMS algorithm.1/ represents the residual error vector.k/jj2 ] D k 2¼/ (3.k/xŁ . consequently the expectation of (3.0/ holds.0/ Let c.0/ Jmin 1 1 k k½0 (3.0/jj2 ] C ¼2 N rx . the convergence behavior of the LMS algorithm (3.k/j2 ] D Jmin C rx .79): D 1 C rx . .k/ D c.1/ D [0.¼2 N rx . then we get J .218 Chapter 3.262) is easily determined.3.0/E[jj c.k/ is statistically independent of x.70) and the following assumptions: 1.1.k/ xŁ . if N < Nh then copt in (3. 3.272) The result (3.262) for k ! 1 (equal to copt ) is also not affected by w. 0.k/] D rx . Anyway. as seen in Section 3. emin .268) From (3.k/ approaches E[c. c.269) where h.271) E[jj c. In any case J .272) is obtained by (3. On the other hand. : : : . 2. and 2 Jmin D ¦w C rx .
273) A faster convergence and a more accurate estimate. this.06. which leads to a misadjustment equal to MSD D 0:26.269). ¼ D 0:1 is chosen.1/jj2 ] D ¼ and Â Ã N J .1 Consider an unknown system whose impulse response. . white. given in Table 1. and Gaussian with statistical power 2 ¦w D 0:01. the convergence curves of the meansquare error (estimated over 500 realizations) are shown in Figure 3. for ¼ rx . it results in E[jj c.274) N Jmin 2 (3. For the LMS algorithm. As discussed in Appendix 3. At convergence. for ﬁxed ¼. Convergence curves of the meansquare error for system identiﬁcation using LMS and RLS. Examples of application 219 Indeed.B.4 on page 26 as h 1 . The noise is additive.272) is an extension of (3. are obtained by choosing a smaller value of N .3. Example 3. Identiﬁcation via standard LMS and RLS adaptive algorithms is obtained using as input a maximallength PN sequence of length L D 31 and unit power.1/ D Jmin 1 C ¼ rx .27. as index of the estimate quality we adopt the ratio: 3n D 2 ¦w E[jj hjj2 ] (3.0/ 2 (3. has energy equal to 1. (3. Mx D 1.27.6. For a ﬁlter with N D 5 coefﬁcients.78). may increase the residual estimation error (3. however.275) Figure 3.0/ − 1.6.
28. . the two methods tend to give the same performance in terms of speed of convergence and error in steady state.k/. nonlinear relation between z. which can be identiﬁed by a table or randomaccess memory (RAM) method. At convergence.k/ and x.x. Adaptive transversal ﬁlters where h D c h is the estimate error vector.i/ 2 A.k// (3.k/ (3. the RLS algorithm usually yields a better estimate than the LMS. it results: ( 3:9 for LMS (3.k//j2 ] O (3.k 2/] D g. However.k 1/. Then z. given by z.28. that is for k D 30 in our example. Adaptive scheme to estimate the inputoutput relation of a system. As a result it is usually preferable to adopt the LMS algorithm.k/ and the gradient estimate is given by rg je. The cost function to be minimized is expressed as E[je. as it leads to easier implementation.x. Finite alphabet case Assume a more general.k/ assumes values in an alphabet with at most M 3 values. ﬁnite alphabet with M elements. x.k/j2 D O 2e.220 Chapter 3.278) Figure 3. x.277) where x.k/.276) 3n D 7:8 for RLS We note that.k/ D g[x. for systems with a large noise power and/or slow timevarying impulse responses. as illustrated in Figure 3.k/j2 ] D E[jd.279) g. even if the input signal is white.
the input vector x.2 Adaptive cancellation of interfering signals With reference to Figure 3. We note that. however.9) of d. w1 is ﬁltered by an adaptive ﬁlter with coefﬁcients fci g.k/ O O (3. Often.B.280) In other words. x. so that e. consisting of the noise signal w1 . 3. i D 0. according to the equation g.k/ D d. : : : . N ﬁlter output. the content of a memory location can be immediately identiﬁed by looking at the output.282) We note that this method is a block version of the LMS algorithm with block length equal to the input sequence. the input is determined on the domain with sampling period Tc .x. it is necessary to access each memory location several times to average out the noise.k/. Primary input. we consider two sensors: 1.k/ D N 1 X i D0 1. 1. and to update the RAM with the values of fd.x.k/ D s. with s ? w1 .x/ D E[g.k/g. d.x.i. 1. In the absence of noise.k/ identiﬁes a particular RAM location whose content is updated by adding a term proportional to the error.A.281) To complete the identiﬁcation process.4 In this section and in Appendix 3. and ¼ is given by the relative frequency of each address. Reference input. Observation 3.k/g is i. This is equivalent to considering g. : : : (3. Examples of application 221 Therefore the LMS adaptation equation becomes g. the value at each RAM location is scaled by the number of updates that have taken place for that location.29. where the RAM is initialized to zero. it is convenient to represent the estimate of h determined on Tc =F0 as F0 estimates determined on Tc . An alternative method consists of setting y. Using the polyphase representation (see Section 1.283) 2. consisting of the desired signal s corrupted by additive noise w0 . We assume that w0 and w1 are in general correlated.x. the observation d and the input x are determined on the same time domain with sampling period Tc . if the RAM is initialized to zero. given by y.k/ selects in the average each RAM location the same number of times.k/ D 0 during the entire time interval devoted to system identiﬁcation. In practice.k// D g. if the sequence fx.k/w1 .k// D g.k// C d..k// C ¼e.6. so that the ci .k/ C w0 . however.3. and the system output signal is determined on Tc =F0 .284) .k i/ (3.d.x/ C w] O (3.k/ O O k D 0.k/ with s ? w0 (3.6.
General conﬁguration of an interference canceller. Deﬁning the error e.287) for y.k/] D E[s 2 . is given by J D E[e2 .k/.z/ D Pdx . w1 and w0 are uncorrelated: min J D E[.30.285) the cost function.k/ D d.z/ Px .29.z/ (3. is the most accurate replica of w0 .k/ D w0 .k/ D s.k/] C E[. In this case e.286) y.k/ D d.s.289) .288) D E[. 1.222 Chapter 3. 2.w0 . In this case e.k/ C w0 .w0 .k/ C w0 .k/ y.k/ c c y.k/ y.k/.k//2 ] for y.k//2 ] (3. for a general input x to the adaptive ﬁlter.0/ (3. Adaptive transversal ﬁlters Figure 3.k/ C w0 . assuming realvalued signals and recalling that s is orthogonal to the noise signals.k/ D 0.k//2 ] D rs .50)) Copt .k/] c c (3. w1 and w0 are correlated: min J D rs .k/ D s. the Wiener–Hopf solution in the ztransform domain is given by (see (2.k/ and the noise w0 is not cancelled.k/.0/ C min E[.k/ (3.s.k/ We have two cases. General solution With reference to Figure 3.k//2 ] C min E[y 2 .
31.z/H .30.z/ (3.z/ D 1 H . Examples of application 223 Figure 3. Block diagram of an adaptive cancellation scheme. Speciﬁc conﬁguration of an interference canceller.289) becomes Copt .z/H Ł . 0 0 Adopting for d and x the model of Figure 3.290) Copt . in which w0 and w1 are additive noise signals uncorrelated with w and s.6.291) .z/ D 0 If w1 Á 0.z/ C Pw .31. and using Table 1.3. (3.290) becomes Pw .z/H Ł .1=z Ł / 0 Pw1 .1=z Ł / (3. (3.3. Figure 3.
We note that in this case x2 can be obtained as a delayed version of x1 .2³ f 0 kTc C '0 / (3. we take as reference signals x1 . It is easy to see that x2 is obtained from x1 via a Hilbert ﬁlter (see Figure 1.32.28).k/ (3.6.k/ D B cos.k/ D B sin.k/ c2 .32.295) (3.4 Disturbance cancellation for speech signals With reference to Figure 3.3 Let Cancellation of a sinusoidal interferer with known frequency d. As shown in Figure 3. The reference signal Figure 3.293) At convergence.6.k/x 1 .292) where s is the desired signal. Adaptive transversal ﬁlters 3. the primary signal is a speech waveform affected by interference signals such as echoes and/or environmental disturbances.33. and the sinusoidal term is the interferer.k/x 2 .296) (3.2³ f 0 kTc C '/ The adaptation equations of the LMS algorithm are c1 .k C 1/ D c2 .2³ f 0 kTc C '/ and x2 .k C 1/ D c1 .k/ C ¼e.k/ D s. .294) (3.k/ C A cos. Conﬁguration to cancel a sinusoidal interferer of known frequency.34. The relation between d and output e corresponds to a notch ﬁlter as illustrated in Figure 3.k/ C ¼e. the two coefﬁcients c1 and c2 change the amplitude and phase of the reference signal to cancel the interfering tone. 3.224 Chapter 3.
from the primary signal.3.33. The output signal is a replica of the speech waveform. Examples of application 225 Figure 3. which is correlated to the reference signal. the speech signal of user A is transmitted over a transmission line consisting of a pair of wires (local loop) [31] to the central ofﬁce A. the signal transmitted by user A and the signal received from user B. obtained by removing to the best possible extent the disturbances from the input signal.6. Figure 3.5 Echo cancellation in subscriber loops With reference to the simpliﬁed scheme of Figure 3. the adaptive ﬁlter output will attempt to subtract the interference signal. i. where the signals in the two directions of transmission. Disturbance cancellation for speech signals. are separated by a device called . At convergence.e.34. Frequency response of a notch ﬁlter. 3.35.6. consists of a replica of the disturbances.
the hybrids give origin to echo signals that are added to the desired speech signals. it is sufﬁcient to substitute for each sensor the ﬁlter with a single complexvalued coefﬁcient [32. Because of impedance mismatch. it is convenient to use several sensors.226 Chapter 3. as will be discussed in Chapter 16. For speech waveforms. 33] (see Section 8. For narrowband signals. to equalize the desired signal and remove interference. with the task of ﬁltering signals in space. the echo of signal A that is generated at the hybrid A can be ignored because it is not perceived by the human ear.6 Adaptive antenna arrays In radio systems. Transmission between two users in the public network.36. Adaptive transversal ﬁlters Figure 3. The case for digital transmission is different.18). e will consist of the speech signal B only.37. Figure 3. The signals of the array are then equalized to compensate for linear distortion introduced by the radio channel. . A similar situation takes place at the central ofﬁce B.e. A method to remove echo signals is illustrated in Figure 3. A general scheme for wideband signals is illustrated in Figure 3. Conﬁguration to remove the echo of signal A caused by the hybrid B. hybrid. At convergence.6. an antenna array. 3. with the roles of the signals A and B reversed.35. discriminating them through their angle of arrival. i. where y is a replica of the echo.36.
3. otherwise part of the desired signal would also be cancelled.7 Cancellation of a periodic interfering signal For the cancellation of a periodic interfering signal. is needed to decorrelate the desired component of the primary signal from that of the reference signal. Antenna array to ﬁlter and equalize wideband radio signals. where: ž we note the absence of an external reference signal. . ž a delay 1 D DTc . 3. On the other hand. to cancel a wideband interferer from a periodic signal it is sufﬁcient to take the output of the adaptive ﬁlter (see Figure 3. the reference signal is generated by delaying the primary input. Examples of application 227 Figure 3.38.39). we can use the scheme of Figure 3. where D is an integer.37.6.6.
Scheme to remove a sinusoidal interferer from a wideband signal.39.40. Scheme to remove a wideband interferer from a periodic desired signal. Adaptive transversal ﬁlters Figure 3. Figure 3.38.228 Chapter 3. . Figure 3. Scheme to remove a periodic interferer from a wideband desired signal.
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therefore all binary sequences of r bits are generated. ž The number of bits equal to “1” in a period is 2r “0” is 2r 1 1. In other words. but with different initial conditions. be the values assumed by the sequence in a period. 1.298) In both formulae the approximation is valid for a sufﬁciently large r. the elements of a sequence can be determined by any 2r consecutive elements of the sequence itself. except the all zero sequence. A practical example is given in Figure 3. and have period equal to L D 2r 1. which is generated by the recursive equation p.`/g. 1. which are generated by the same shiftregister. is still an rsequence.41 for a sequence with L D 15 (r D 4).3.A PN sequences In this Appendix we introduce three classes of deterministic periodic sequences having spectral characteristics similar to those of a white noise signal.`/ 2 f0.g. hence the name pseudonoise (PN) sequences. : : : . 35]. ž The sum of two rsequences. L 1. ž The linear span. ž Every nonzero sequence of r bits appears exactly once in each period. PN sequences 233 Appendix 3.299) ..` 4/ (3. Maximallength sequences Maximallength sequences are binary PN sequences. e. e.` 3/ ý p. It can be shown that the maximallength sequences enjoy the following properties [34. p. ` D 0.g.A. also called rsequences.297) and the relative frequency of a subsequence of length i < r with all bits equal to zero is 2r i 1 '2 2r 1 i (3. that are generated recursively. Let f p. while the remaining elements can be produced by a recursive algorithm (see.. is equal to r [36]. 1g. using a shiftregister (see page 877). and the number of bits equal to ž A subsequence is intended here as a set of consecutive bits of the rsequence. that determines the predictability of a sequence. The relative frequency of any nonzero subsequence of length i Ä r is 2r i '2 2r 1 i (3. the BerlekampMassey algorithm on page 891).`/ D p.
i. the all zero initial condition must be avoided.m/mod L D 0 otherwise (3. with the exception of the values assumed for .d. Assuming initial conditions p. also from the point of view of the autocorrelation function. mapping “0” to “ 1” and “1” to “C1”. even if deterministic and periodic.5. Spectral density (periodic of period L) Â Ã L 1 X 1 D Tc Pp m r p . 4/ D 1.300) Obviously. Mean L 1 1 X 1 p. Correlation (periodic of period L) L 1 1 X r p .299) we obtain the sequence 1 0 0 {z} {z} 0 1 0 0 1 1 0 1 0 1 1 1 {z} : : : p.i. sequence from the point of view of the relative frequency of subsequences of bits.` L `D0 n/mod L D 8 < : 1 1 L for . where ý denotes modulo 2 sum.n/e L Tc nD0 1 j2³ m L T nTc c 8 > 1 > Tc < L Â Ã D > > Tc 1 C 1 : L for . the spectral density of maximal length sequences is constant. 3/ D p.302) otherwise 3. The above properties make an rsequence. 1. In fact. 2/ D p. Adaptive transversal ﬁlters Figure 3.`/ p Ł .301) 2. To generate sequences with a larger period L we refer to Table 3.n/ D p. . Generation of a PN sequence with period L D 15. 1/ D p.L 1/ (3.303) We note that.234 Chapter 3. we get the following correlation properties.d. appear as a random i.n/mod L D 0 (3.0/ p.41.`/ D L `D0 L (3.1/ p. It turns out that an rsequence appears as random i.m/mod L D 0. applying (3.
L ` D 0.` p.`/ D p.`/ D p.`/ D p.3.`/ D p.` 1/ ý p.` p.` 2/ ý p.` p. 38. these sequences have the following properties.` p. Because of these characteristics they are also called polyphase sequences [37.` p. 39].` 8/ CAZAC sequences The constant amplitude zero autocorrelation (CAZAC) sequences are complexvalued PN sequences with constant amplitude (assuming values on the unit circle) and autocorrelation function r p .` 13/ ý p.` 14/ ý p.` 12/ 13/ 14/ 4/ ý p.` 17/ 18/ 17/ ý p. .` p.` 2/ ý p.`/ D p.` 14/ ý p.`/ D p.` p.` 16/ 11/ ý p.`/ D p.` p.` p.`/ D p.` 1 2/ 3/ 4/ 5/ 6/ 7/ 3/ ý p. 1.` p.`/ D p.` p.`/ D p. L 1 1 (3.` 6/ ý p.` 14/ ý p.` p.`C1/ L ej It can be shown that. The CAZAC sequences are deﬁned as.` p.` p. in both cases.`/ D p. Let L and M be two integer numbers that are relatively prime.`/ D e j p. : : : .305) M³ `.`/ D p.` p.` 11/ ý p.` 20/ 18/ ý p.` 2/ ý p.`/ D p.n/ equal to zero for . : : : .304) (3.` 3/ ý p.n/mod L 6D 0.`/ D p.` p.` p. for different values of r. 1.`/ D p.` 5/ ý p.`/ D p.` 2/ ý p. for L even for L odd p.` 9/ 10/ 11/ 10/ ý p.` Period L D 2r 1/ 1/ ý p.A.` p.` 11/ ý p. PN sequences 235 Table 3.` 12/ ý p.` p.` 7/ ý p.` 15/ 13/ ý p.` 12/ ý p.5 Recursive equations to generate PN sequences of length L D 2r 1.`/ D p.`/ D p.` 5/ ý p. r 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 p.` 11/ ý p.`/ D p.`/ D M³ `2 L ` D 0.` 9/ ý p.` 3/ ý p.`/ D p.` 19/ 14/ ý p.` 17/ ý p.
called preferred rsequences [36]. Let a D fa. whose CCS assumes only three values. that is: b D fb. is the set of Gold sequences [41.307) Gold sequences In a large number of applications. Spectral density Â Ã 1 D Tc Pp m L Tc (3. Construction of pairs of preferred rsequences. We deﬁne now another rsequence of length L D 2r 1 obtained from the sequence a by decimation by a factor M. as for example in spreadspectrum systems with codedivision multiple access (see Chapter 10). or.309) . 42]. ž The factor M satisﬁes one of the following properties: M D 2k C 1 or M D 22k 2k C 1 k integer (3. Correlation ( r p .308) 1 0 for .M`/mod L g We make the following assumptions.n/ D 3.`/g D fa. that have good autocorrelation and crosscorrelation characteristics. An important class of periodic binary sequences that satisfy these properties. four or maybe even a greater number of values. i.236 Chapter 3. ž Each sequence of the set must be easily distinguishable from its own time shifted versions. in other words. Mean L 1 1 X p. We show now the construction of a pair of rsequences. In general the crosscorrelation sequence (CCS) between two rsequences may assume three. that is r must be odd or equal to odd multiples of 2. rmod 4 D 2. sets of sequences having one or both of the following properties [40] are required.310) (3. ž rmod 4 6D 0.306) 2.n/mod L D 0 otherwise (3.e. Adaptive transversal ﬁlters 1.`/g be an rsequence with period L D 2r 1. ž Each sequence of the set must be easily distinguishable from any other sequence of the set and from its timeshifted versions.`/ D 0 L `D0 (3.
9.5. 7. 9.`/bŁ . PN sequences 237 ž For k determined as in the (3.3`/mod L g D .A. k/ D g:c:d:. 1.3. 9. The set (3. deﬁning g:c:d:. We deﬁne the set of sequences: G. Therefore e D g:c:d:. 1.r.a. 7.`/g D fa. 2/ D 1 and M D 22 C 1 D 5. 1.` L `D0 n/mod L r Ce 2 2 8 > > 1< D > L> : 1C2 1 1 2 r Ce 2 (value assumed 2r (value assumed 2r (value assumed 2r e 1 C2 e 2r e 1 times) 1 times) times) (3. 7. k/ as the greatest common divisor of r and k. 9. The sequence fb. b/ D fa. 9. A set of Gold sequences can be constructed from any pair fa. 1.316) contains L C 2 D 2r C 1 sequences of length L D 2r 1 and is called the set of Gold sequences. 31 1. 1. 1. a ý Z L 1 (3.r.314) (3. 1/ We note that. k/ D (3.n/g D 1 . 7.7. It can be proved [41. for the two sequences fa 0 . 36]: rab .5. : : : .1 (Construction of a pair of preferred rsequences) Let the following rsequence of period L D 25 1 D 31 be given: fa. a ý Z 2 b. we take k D 1.0001010110100001100100111110111/ The CCS between the two sequences. b. then e D g:c:d:.r. 9.312) r Ce 2 2 r Ce 2 2 Example 3.`/g D . 1. is: frab .n/ D L 1 1 X a.316) where Z is the shift operator that cyclically shifts a sequence to the left by a position. 1. if we had chosen k D 2. 1.A. a ý b.`/g of preferred rsequences of period L D 2r 1. a ý Z b.310). assuming “0” is mapped to “ 1”. let ( 1 r odd e D g:c:d:. 7. 1. Construction of a set of Gold sequences. 1. 1. 7.`/g obtained by decimation of the sequence fa. 7.`/g and fb.`/g . 7. 7.`/g is then given by fb.0000100101100111110001101110101/ (3. 1.`/g and fb0 . 42] that. 1/ D 1 and M D 2k C 1 D 21 C 1 D 3.313) As r D 5 and rmod 4 D 1.315) bg (3. or else M D 22Ð2 22 C 1 D 13.311) 2 rmod 4 D 2 Then the CCS between the two rsequences a and b assumes only three values [35.
`/g and fb0 . 1. 1. 1. 7. 7. 7. with the exception of zero lag. 9. 1.`/gD. 1.319) 1 .`/g: fa. 1. 1.a. 1. 1. 1. 9. 1. 1. 1. b/. 1. 9. 1. 1. 1. 1. 1.`/g D fa. 1. 1. 7. 1/ 1. 1/ (3. 1.321) frb0 . 1. 1. 1. 1.31.322) .`/ ý b. 7. 1. 1. 1. 1. 31 1 9. the two sequences (3. 9. 1. 1. hence L D 25 1 D 31. 1.n/gD 1 . 1. For example we calculate the ACS of fa.317) r C2 r C2 L> : 1 1C2 2 rmod 4 D 2 1 2 2 Clearly.313) and (3. 1. 9. 9. 1. 7. 1. 1.A. (3.n/gD frab0 . 1. 1. 7.`/gD. 7. 1. 1. 7.31. 1. 9. 1. 7. 1. 1.n/ D (3.1. 1. 7. 1. 1. 1. 1. 9. 1/ fb0 . 1. 1. 1/ fra . 9.318) (3.238 Chapter 3. 1. 1. 1. 1. the ACS of a Gold sequence no longer has the characteristics of an rsequence. 1.320) 1 . 1. 1. Example 3. 1. 1.` 2/g D a ý Z 2 b. Adaptive transversal ﬁlters belonging to the set G. and the CCS between fa. 1/ (3. 1. 1. 1. from which it is possible to generate the whole set of Gold sequences. 1. 1. 1. From Example 3. 1.2 (Gold sequence properties) Let r D 5. 1. 9. 1. 1. 7. 1. 1. 1.`/g and fb0 . 1. 7. assume only three values: 8 r C1 r C1 > 1C2 2 r odd 1 2 2 1< 1 ra 0 b0 . 1.A. 1. 1. 1. 1. 1. 31 1. 7. 1. 1. 1. 1. 7. 1. 1. 1. 7. 1. 7. 1. 1. 1. 1. 1. 1. 1. 7. 1. 1. 1.n/gD (3. 1. 7. 7. 1 31 1. 1. 1. 1. 1. 1. 7. 1. 1. 1. as is seen in the next example. the CCS as well as the ACS. 1. 9.314) are a pair of preferred rsequences.
L 1/. where we choose L ½ N . : : : .42.L C 1/N samples.N 1/ C . : : : .0/. i D 0.n/ D rx . obtained by repeating f p.1=L/w L .i/g. L 1.323) In practice. we consider the scheme illustrated in Figure 3. we have rdx . 1. we recall that if the input to a timeinvariant ﬁlter is periodic with period L. x. Identiﬁcation of a FIR system by PN sequences 239 Appendix 3.0/. in other words.k/mod L .42. To estimate the impulse response fh i g.B.A): ( r p . . : : : . and an input sequence x with length of at least . 1. i D 0.k/ is given by Figure 3. We assume a delay m 2 f0. to the impulse response fh i g. the output will also be periodic with period L. L 1. a rectangular window g Rc . N 1g. N 1. which describes the relation between input and output of an unknown system with impulse response fh i g. is used as input. In fact.n/ D1 '0 n D 0. 1. : : : (3. Correlation method to estimate the impulse response of an unknown system.264). : : : . instead of noise a PN sequence with period L. we observe that the crosscorrelation between d and x is then proportional. : : : n D 1. we take as an input signal white noise with statistical power rx .1 Correlation method With reference to (3.k/. L C 1.B. i D 0. and that the system is started at instant k D 0.i/g. To estimate the impulse response of a linear system.324) Moreover. the output v.0/h n (3. with a factor rx . L .k/ D .n/ D rzx . N 1. For k ½ . 2L .k/ D p.n/ D rx Ł h.B Identiﬁcation of a FIR system by PN sequences 3.3. We recall that the autocorrelation of a PN sequence is also periodic with period L and is given by (see Appendix 3. f p. 1. : : : .
k/ D L 1 X 1 u.k L `D0 `/ p Ł .k ` ` m/mod L C As L 1 1 X p.k ` m/mod L D r p . Variance L 1 1 X var[v.327) are obtained as follows.k/ ' h m (3.k L `D0 `/ p Ł .k/] D var w.k N 1 X i D0 `/ p .k m/mod L ` i/mod L pŁ .k/ D N 1 X i D0 h i r p .328) Mean and variance of the estimate of h m given by (3.329) " # `/ p .k L `D0 `/ p Ł .240 Chapter 3. 2.k/] D assuming w has zero mean.325) becomes v. Mean E[v.k L `D0 m/mod L (3.k # ` m/mod L Cw.`/j Ä 1. 1.m i/mod L C L 1 1 X w.327) If L × 1.324) we get v.326) (3.330) assuming w white and j p.k ` m/mod L L 1 X 1 D L `D0 " N 1 X i D0 h i p.k ` ` i/mod L pŁ .k ` m/mod L (3.325) D hi ` i/mod L pŁ .k Ł ` m/mod L ' 2 ¦w L (3. the second term on the righthand side of (3.k L `D0 `/ D L 1 X 1 d. Adaptive transversal ﬁlters v.m i/mod L (3. .327) can be ignored.k L `D0 L 1 1 X w.k L `D0 N 1 X i D0 h i r p .k Ł L 1 1 X p. hence observing (3.m i/mod L (3.
: : : .N 1/ d.L 1/ to 2.k m/mod L ' h m m D 0.k .N 1/ C .N 1/ samples. h 1 . However.L 1/ X kD. we get v. from (3. : : : .L 1// i D 0.43.332) we get 1/ C . h 1 .N 1/ . N 1 (3. it requires synchronization between the two PN sequences.i C . the samples at the correlator output from instant k D . Identiﬁcation of a FIR system by PN sequences 241 Figure 3.B. Using the scheme of Figure 3. at transmitter and receiver. : : : . varying m from 0 to N 1 it is possible to get an estimate of the samples of the impulse response of the unknown system fh i g at the output of the ﬁlter g Rc .N 1/ C .k/ D L 1 1 X d.43: with steps analogous to those of the preceding scheme.L 1/ C `/ p Ł .42.331) An alternative scheme is represented in Figure 3. after a transient equal to N 1 instants.L 1//mod L (3.3. give an estimate of the samples of the impulse response of the unknown system fh i g. h N 1 ] be the ﬁlter coefﬁcients to be estimated and hT D [h 0 .N 1/ C .k L `D0 . N 1 (3.k/g in a buffer and computing the correlation offline: rdx .m/ D O 1 L . O N 1 ] those estimated.332) After a transient of . : : : .333) O h i D v. this scheme has two disadvantages: 1.k/ pŁ . Correlation method via correlator to estimate the impulse response of an unknown system. it requires a very long computation time (N L). L consecutive output samples fd. Both problems can be resolved by memorizing.L 1//mod L ' h .N In other words. 1.` C . 1.L 1/. Let h be the estimation error vector h O hDh h .N 1/C. 2. Signaltoestimation error ratio O O O Let hT D [h 0 .
k/ O d. with an error given by z.h i O h i / x. As a consequence. we refer to the normalized ratio 3n D 2 ¦w 3e D 3 Mx E[jj hjj2 ] (3. Adaptive transversal ﬁlters The quality of the estimate is measured by the signaltoestimation error ratio 3e D jjhjj2 E[jj hjj2 ] (3.k/ (3.z. represented in Figure 3. In our case Mx D 1.k// C w.339) consists of two terms. is z.k/.340) where a PN sequence of period L D N .k n/mod L h n (3. .k/ the output of the identiﬁed system.2 Methods in the frequency domain System identiﬁcation in the absence of noise In the absence of noise (w D 0).335) where Mx is the statistical power of the input signal. is assumed as input signal x.42): 3D Mx jjhjj2 2 ¦w (3.338) we note that the difference d. Finally. we have to take into consideration the noise present in the observed system and measured by (see Figure 3. 3. the output signal of the unknown system.k i/ (3.337) O O the fact that h 6D h causes d.k/ D .k/.k/ D L 1 X nD0 x.334) On one hand.242 Chapter 3.k/ 6D z.k/ O d.336) measures the ratio between the variance of the additive noise of the observed system and the variance of the error at the output of the identiﬁed system. From (3. equal to the length of the impulse response to be estimated. O d.k/ O d.338) having variance Mx E[jj hjj2 ] for a white noise input.k/ D N 1 X i D0 .B. Let us consider the vector zT D [z.k/ D N 1 X i D0 O h i x. (3.k i/ (3.42.336) O We note that if we indicate with d. one due to the estimation error and the other due to the noise of the system.
k/ D z.k . x.k C 1/. in the frequency domain. Being M circulant. Mean O E[h] D h 2.346) are obtained as follows.348) . X m D DFT[x.k/ C s L w. Identiﬁcation of a FIR system by PN sequences 243 z. using the output samples fz.2. for k D L 1.346) 2 Assuming that w is zeromean white noise with power ¦w . : : : .k/. L 1 (3. After an initial transient of L 1 samples.i/j2 (3.342) can be rewritten in terms of the discrete Fourier transforms as Zm D Xm Hm from which we get h k D DFT or.k C . 2. 1.k/ kDL 1. 2.k/ D x L h. setting s. Variance " E[jj hjj2 ] D E L 1 X kD0 (3. and Hm D DFT[h k ]. from a computational complexity point of view.k 1/mod L .k/ D h k C s L w.342) Letting Zm D DFT[z. : : : .L 1//].B.341) can be solved very efﬁciently.k/]. : : : .k/ D s L z.344) hk D s L z. : : : .L 1//g we obtain a system of L linear equations in L unknowns.345) the expression of the output signal obtained in the presence of noise. : : : .k/].341) can be substituted by the circular convolution (see (1.3. L ½ 1 (3.105)) z. m 1 m D 0.k/ mod L .345) System identiﬁcation in the presence of noise Substituting in (3. the product in (3.L 1//mod L ].k/ (3.k/ D DFT 1 [1=X ].L 1/ (3.k/ C w. Because the input sequence is periodic.L 1/.347) # O jh k O h k j2 D L E[jh k 2 h k j2 ] D L ¦w L 1 X i D0 js. d. : : : . rather than inverting the matrix M. : : : .341) assuming k D L 1 in the deﬁnition of z and M. z. which in matrix notation can be written as z D Mh (3. and a circulant matrix M whose ﬁrst row is [x. (3. The system of equations (3. the system (3.k/ (3. x.341) admits a unique solution if and only if the matrix M is nonsingular.L 1/. z. mean and variance of the estimate (3.343) Ä Zm Xm k D 0. the estimate of the coefﬁcients of the unknown system is given by O hk D s L d.
336) becomes 3n D L C1 2L (3.N 1//]. observing (3.351) 2 1j (3. (3. we introduce the following quantities.1/.348). letting xT .L 1//g. In other words. . 44.244 Chapter 3. : : : .353) we see that the observation of L samples of the received signal requires the transmission of L T S D L C N 1 symbols of the training sequence fx.k .k/j2 (3. it gives an estimate with variance equal to the noise variance of the original system.337). we have that all terms jX j j2 are equal jX j j2 D L j D 0. (3.k/ D [x. 1.348) is equal to ¦w . it turns out X0 D 1 jX1 j2 D jX2 j2 D Ð Ð Ð D jX L hence. 45].B. the sum of squared errors at the output is given by ED where.0/. In the ﬁrst case.k/ k D .k/ D hT x.350) D L C1 For CAZAC sequences.k x.k/ D hT x. the noisy output of the unknown system can be written as d.353) With reference to the system of Figure 3..308). : : : .3. therefore 3n D 1.303). (3. x. x. The unknown system can be identiﬁed using the LS criterion O [43.k/. L 1 (3.352) 2 and the minimum of (3. : : : .k/ O d. O O d.3 The LS method 1/. x. 3.N 1/. : : : .k/ C w. if L is large.349) it is possible to particularize (3. from (3.N 1/ C . Adaptive transversal ﬁlters Using the Parseval theorem L 1 X i D0 js. in the best case.k/ As for the analysis of Section 2.354) . from (3. Although this method is very simple.N 1/ C .355) N X 1 1CL kDN 1 jd. CAZAC sequences yield 3 dB improvement with respect to the maximallength sequences.42. For a certain estimate h of the unknown system.i/j2 D L 1 L 1 1 X 1 X 1 jS j j2 D L jD0 L jD0 jX j j2 (3. from (3. it has the disadvantage that.336) for PN maximallength and CAZAC sequences.L 1/ From (3.
n D 0. Identiﬁcation of a FIR system by PN sequences 245 1.i. observing (3. : : : .356) 2.364) . : : : .k/j2 (3. N 1 (3.n/ D N X 1 1CL kDN 1 1/] (3. In some applications it is useful to estimate the O variance of the noise signal w that.0/.N where #.359) d.B.360) Then the cost function (3.362) 1 in the (3.363) 1 ϑ (3.¦w / D O2 1 Emin L (3.354) becomes E D Ed O hH ϑ O O O ϑ H h C hH h (3. Correlation matrix of the input signal D [8. for h ' h can be assumed equal to .357) x Ł .i.k/ x Ł .358) 3.k i/ x.k n/ (3. #. Energy of the desired signal Ed D N X 1 1CL kDN 1 jd.3. we can assume that is positive deﬁnite and therefore the inverse exists. n/ D N X 1 1CL kDN 1 i.362) can be precomputed and memorized.361) As the matrix is determined by a suitably chosen training sequence.k n/ (3. because We observe that the matrix it depends only on the training sequence. The solution to the LS problem yields O hls D with a corresponding error equal to Emin D Ed O ϑ H hls (3.339). n/] where 8. Crosscorrelation vector ϑ T D [#.
From (3.k/ (3.360).L 1/ (3.367) IHo (3.k/ xŁ .L 1/ 3 ::: x.139). we obtain the relation ϑ D hCξ (3.371) in (3.370) observing (3.4. We note the introduction of the new symbols I and o. in relation to an alternative LMMSE estimation method.B.I H I/ which coincides with (3.353) in (3. we have D IHI and O hls D . substituting (3.0/ 7 : :: : 5 : : 1// : : : x.366) ϑ D IHo (3.369) Substituting (3.N 1/ C . d. Adaptive transversal ﬁlters Formulation using the data matrix From the general analysis given on page 152.N where d. Desired sample vector o T D [d.L 1//] (3. L ð N observation matrix 2 6 I D4 x..371) Consequently.k/ xŁ .372) .369).k/ is given by (3.246 Chapter 3. the estimation error vector can be expressed as hD 1 ξ (3.362). : : : .362). Observing (2.160).N 2.353).357). (2. 1. which will be given in Section 3.N : : : 1/ C .. and letting ξD N X 1 1CL kDN 1 w. (3.131).k/ (3.359) can be rewritten as ϑ D N X 1 1CL kDN 1 d. we recall the following deﬁnitions. and (2.368) Computation of the signaltoestimation error ratio We now evaluate the performance of the LS method for the estimation of h. 1 x.365) 1/.
373) Therefore. Ł / 1 (3.379) Â Ã 1 N ðN 1 IC D L C1 L C1 N (3.376) yields 3n D L N are equal to (3. from (3. the correlation matrix can be written as D .376) is diagonal. We make the following observations.378) The (3. we get 3n D .376).44 the behavior of 3n is represented as a function of N . Now.377) 1 Using as training sequence a CAZAC sequence.375) and. 3n also doubles. ξ Ł is a zeromean random vector with correlation matrix 2 Rξ D E[ξ ∗ ξ T ] D ¦w Ł (3. The elements on the diagonal of 1=L.379) the inverse is given by 1 1 N ðN (3. for CAZAC sequences (solid line) and for maximallength sequences (dotteddashed line).L C 2 N / (3. Identiﬁcation of a FIR system by PN sequences 247 2 If w is zeromean white noise with variance ¦w .L C 1 N / N .381) In Figure 3. using as training sequence a maximallength sequence of periodicity L.380) which. yields 3n D . and 3n .374) In particular.3. the matrix D LI where I is the N ð N identity matrix.tr[ 1 ]/ 1 (3. . substituted in (3.L C 1/I From (3. and (3. h has mean zero and correlation matrix R h 2 D ¦w .L C 1/. Ł / 1 ] (3.378) gives a good indication of the relation between the number of observations L. the number of system coefﬁcients N .336). (3. and indicating with 1 N ðN the matrix with all elements equal to 1. with parameter L.B. doubling the length of the training sequence. For example. 2 E[jj hjj2 ] D ¦w tr[.
for L D 15 the maximallength sequence yields a value of 3n that is about 3 dB lower than the upper bound (3. it is necessary to set to zero all coefﬁcients whose amplitude is below a certain threshold. where the number of coefﬁcients may be large. the estimate is usually very noisy. ž For sparse systems.331) is adopted.371). if N is smaller than Nh . we get Â Ã 1 1 hD I hC ξ L L (3. we get 1 O hD ϑ L where ϑ is given by (3.382) . Observing (3. for various values of L.44.270)).378). for example. ž For a given value of L. On the other hand. the two sequences yield approximately the same 3n . N for CAZAC sequences (solid line) and maximallength sequences (dotteddashed line). Adaptive transversal ﬁlters Figure 3.248 Chapter 3. the estimate of the coefﬁcients becomes worse if the number of coefﬁcients N is larger than the number of coefﬁcients of the system Nh . but only a few of them are nonzero. ž If the correlation method (3. 3n vs. choosing L × N . after obtaining the estimate. because of the presence of the noise w. We note that the frequency method operates for L D N .359). The worst case is obtained for L D N . ž For a given N . Therefore. the estimation error may assume large values (see (3.
387) where 3 is deﬁned in (3.0/j2 <L 2 ¦w 3. We observe that using the correlation method. In fact.0/j2 (3.0/j2 / h i . and has a covariance matrix equal to .B.jjhjj2 C . for a CAZAC sequence. we obtain the same values 3n (3.4 The LMMSE method We refer to the system model of Figure 3.386) hence 3n D 1 NC 3 C .N 2/ jH. Let us assume that w and h are statistically independent random processes.42.362) coincides with (3.379) we get Â 1 L Ã I h 2 D 1 jj. and 3n is given by (3.1=L 2 / Rξ .N L2 2/ jH.3..1 L2 I/ hjj2 D X 1 N 1 jh i L 2 i D0 H. Moreover.377) the second term of the denominator in (3.0/j2 2/ 2 ¦w ½ (3.1=L/ I/h. using (3.384) Using a CAZAC sequence. as 1 is diagonal.0/ D PN 1 i D0 1 .324) is strictly true and the correlation method (3.384) vanishes.385) D where H.381) as the LS method. the LS method (3. from (3.373).N L Ä L jH. if L is large enough to satisfy the condition 3 C . In particular.383) 1 2 ¦w (3. we have tr[ ] D N L (3.B. it turns out E[jj hjj ] D and 3n D 1 tr[ ] C L2 Â 1 1 L Ã I h 2 2 Â 1 L Ã I h 2 C 2 ¦w tr[ ] L2 (3. from (3. whose secondorder statistic is known.331). Identiﬁcation of a FIR system by PN sequences 249 Consequently the estimate is affected by a BIAS term equal to . . Using instead a maximallength sequence.335).378).
390) (3. We note that in the LS method the observation was the transmitted signal x. which depends on the ratio between the noise variance and the variance of h. and (2.A.392) I Ł Rh ] T o (3. Recalling the deﬁnition (3. (3.250 Chapter 3.229). (3.Ro 1 Roh /T o .229) to the problem under investigation. the LMMSE estimator is given by O hLMMSE D .353). : : : . We conclude by recalling that Rh is diagonal for a WSSUS radio channel model (see (4.RŁ / w 1 I] 1 I H .394) C I H I] 1 IHo (3.N 1/ C .221)).L 1//] (3.N 1/. . from (3. we can write o D Ih C w Assuming that the sequence fx.391) (3.I Ł Rh I T C Rw / 1 (3.395) We note that with respect to the LS method (3.k/g is known.395). and the components of h are derived by the power delay proﬁle. we desire to estimate h using the LMMSE method given in the Appendix 2. Consequently. (3. The observation is now given by d and the desired signal is the system impulse response h.. Adaptive transversal ﬁlters For a known input sequence x.RŁ / h 2 If Rw D ¦w I.395) introduces a weighting of the components given by ϑ D I H o.389) be a random vector with noise components. which now will be denoted as o.368). while the desired signal was given by the system noisy output d. from the observation of the noisy output sequence d. we have Ro D E[oŁ o T ] D I Ł Rh I T C Rw and Roh D E[oŁ hT ] D I Ł Rh Then (3.366) of the observation vector o.388) where we have assumed E[o] D 0 and E[h] D 0. w.388) provides an estimate only of the random (i. Now.388) becomes O hLMMSE D [.393) can be rewritten as O hLMMSE D [. letting wT D [w. then Rh is also large and likely Rh 1 can be neglected in (3. the LMMSE method (3.e.RŁ / h 1 1 C I H . we have 2 O hLMMSE D [¦w . If the variance of the components of h is large. Consequently. the nondeterministic) component of the channel impulse response. some caution is needed to apply (2.393) Using the matrix inversion lemma (3.RŁ / w 1 o (3.176).
in general E[jj hjj2 ] D tr[R1h ] This result can be compared with that of the LS method given by (3.t/ D C1 X i D0 p. Identiﬁcation of a FIR system by PN sequences 251 For an analysis of the estimation error we can refer to (2. the scheme of Figure 3. is used to modulate in amplitude the pulse Ã Â t Tc =2 (3. we get 2 2 R1h D ¦w f[¦w . Basic scheme to measure the impulse response of an unknown system. is expressed by 1 rx . A PN sequence of period L. .B. repeated several times.399) g.RŁ / w 1 I]Ł g 1 (3.t/ D rect Tc The modulated output signal x is therefore given by x.i/mod L g. (3.401) Figure 3.396) C I H I]Ł g 1 (3.RŁ / h 1 1 C I H .RŁ / h 2 If Rw D ¦w I.45. which uses the error O vector h D hLMMSE h having a correlation matrix R1h D f[.Á/x Ł .t/ D wTc .42 can be modiﬁed to that of Figure 3.5 Identiﬁcation of a continuoustime system In the case of continuoustime systems.B. where the noise is neglected.400) The autocorrelation of x.Á t/dÁ (3.397) Moreover.t/ D L Tc Z LT C 2c L Tc 2 x.3.45 [46].233).398) 3.375). periodic function of period L Tc .t i Tc / (3.
406) Figure 3.t/ D L L Tc 2Tc 2 as shown in Figure 3.¾ /x.Á/dÁ t L Tc Z t Z C1 1 [h.0/ D 1 and r p . Adaptive transversal ﬁlters As g has ﬁnite support of length Tc .t/ given by (3. L 1.`/rg .402) where.403) in (3.− / D v− 0 − /dÁ (3.46.403) Substituting (3.t Tc `D0 `Tc / 0 Ä t Ä L Tc (3.402) and assuming a maximallength PN sequence.t/ D rect g Rc . : : : .t/ D 0 Ã Â Ã jtj t rect Tc 2Tc (3.Á/g.¾ /rx . Autocorrelation function of x.Á t/dÁ D Tc 1 rg .404) rx . in the case of g.Á L Tc t L Tc 0 Z C1 D h.t/. we obtain Ã Â Ã Â ÃÂ t L Tc 1 jtj 1 rect jtj Ä 1 C 1C (3.Á ¾ /d¾ ]x Ł .399).t/ D D 1 L Tc Z t u.t/ D L Tc L Tc L Tc we obtain v.t/ has a simple expression given by rx . and the result is ﬁltered by an ideal integrator between 0 and L Tc with impulse response Ã Â 1 1 t L Tc =2 (3.46 for L D 8.252 Chapter 3. rx .t/ D L 1 1 X r p . . x Ł .t − /.`/ D 1=L for ` D 1. If the output z of the unknown system to be identiﬁed is multiplied by a delayed version of the input.405) w L Tc . we have Â Z Tc g.− ¾ /d¾ D h Ł rx . with r p .
405).411) . We consider the function Z L Tc 1 rx 0 x . related to the clock frequency f 0 D 1=Tc of the transmitter by the relation Â Ã 1 0 (3. we can assume that rx 0 x .− / ' rx . the delay between the two sequence diminishes of the quantity .Á/] x.¾ /rx K 0 Â Ã Z C1 t L Tc ' ¾ d¾ h.409) L Tc t L Tc K If K is sufﬁciently large. given by (3.− / D [x 0 . and L is sufﬁciently large. so that Â Ã Z t 1 t L Tc 0 Ł for t ½ L Tc [x .− /.Á/]Ł dÁ v.¾ /x. Identiﬁcation of a FIR system by PN sequences 253 Figure 3. the output v− is approximately proportional to h.− / At the output of the ﬁlter g Rc .47 is an alternative to that of Figure 3. Therefore the output assumes a constant value v− equal to the convolution between the unknown system h and the autocorrelation of x evaluated in − .Tc0 Tc / D t=K .Á/]Ł x.Á − / dÁ ' rx 0 x − (3. Assuming 1=Tc is larger than the maximum frequency of the spectral components of h.Á ¾ / d¾ ][x 0 . As time elapses.47. In this latter scheme.Á − / dÁ L Tc 0 (3.410) (3. of simpler implementation because it does not require synchronization of the two PN sequences at transmitter and receiver.3. therefore we have Z t Z C1 1 [h.45. The scheme represented in Figure 3.t=Tc0 /. the output z of the unknown system is multiplied by a PN sequence having the 0 same characteristics of the transmitted sequence.408) where − is the delay at time t D 0 between the two sequences.t/ D L Tc t L Tc 0 Â Ã Z C1 t L Tc 0x ¾ d¾ D h. Sliding window method to measure the impulse response of an unknown system.407) f0 D f0 1 K where K is a parameter of the system. but a different clock frequency f 0 D 1=Tc0 .¾ /rx K 0 (3.B.
t L Tc /=K . Therefore the systems in Figure 3.45 are equivalent.K t 0 C L Tc / ' 0 C1 h. .254 Chapter 3.406).47 and in Figure 3.412) where the integral in (3.409) and (3. it can be shown that the approximations in (3.¾ /rx . with the substitution t 0 D . Adaptive transversal ﬁlters or.412) coincides with the integral in (3.t 0 ¾ / d¾ (3.410) are valid. Z y. If K is sufﬁciently large (an increase of K clearly requires a greater precision and hence a greater cost of 0 the frequency synthesizer to generate f 0 ).
1. An intermediate device may compensate for attenuation and/or disturbances introduced by the medium. respectively. where Z i denotes the source impedance. where the signal vi produces v1 through the source voltage divider. ampliﬁers. If Z 1 and Z 2 are respectively the input and output impedances of the 2port network and vo is the opencircuit voltage at the output... more generally. To be able to convey the information represented by these messages to a user situated at a certain distance from the source. many components (e. The transmission medium may consist. The word channel is often used in practice to indicate in abstract terms a transmission medium that allows the propagation of a signal. e. Z L the load impedance. . The corresponding abstract model is illustrated in Figure 4. It may consist of a simple ampliﬁer or. coaxial cable. cables. we obtain the equivalent electrical scheme of Figure 4. 4. waveguide. of one or more of the following media: twistedpair cable. and the signal v L is obtained from vo through the load voltage divider.g. v1 gives origin to vo according to the 2port network transfer characteristics. the input and output signals of the 2port network.2a. and v1 and v L are. radio link. of a repeater. ﬁlters. or optical ﬁber. The fundamental properties of various transmission media will be discussed in the remaining sections. A 2port network is a device that transfers a signal vi from a source to a load. or even—for data signals – of a regenerator that attempts to restore the original source message. that are expressed as voltage signals.2b. From an electrical point of view. The transmitter is a device that converts the source message into a signal that can physically propagate through the transmission medium.Chapter 4 Transmission media The ﬁrst two sections of this chapter introduce several parameters that are associated with the electrical characteristics of electronic devices. a transmission system can be conﬁgured as illustrated in Figure 4.g. : : : ) of a transmission system may be interpreted as a cascade of 2port linear networks.1 Electrical characterization of a transmission system Simpliﬁed scheme of a transmission system We consider a message source. as shown in Figure 4. where we identify input and output twoterminal devices. which could be for example a machine that generates speech signals and/or sequences of symbols. The task of the receiver is to yield an accurate replica of the original message.2c.
2c. or I L =I1 .1) where G1 o .2. however. the frequency response G Ch . f / D G1 o. we have GCh . In the cascade of several 2port networks. Therefore. For these cases. f / and G L .256 Chapter 4.2a.1. . f / (4. with reference to Figure 4. f / denote the frequency responses of the 2port network and of the load. f /G L . To analyze the characteristics of the network of Figure 4. We note that in some cases the frequency response could be deﬁned as VL =I1 .1). Figure 4. I L =V1 . f / of each network is given by the ratio VL =V1 . Transmission media Figure 4. Connection of a source to a load through a 2port linear network. f / will be different from (4. we will refer to the study of twoterminal devices. the expression of GCh . respectively. Simpliﬁed scheme of a transmission system.
the voltage and the current at the load. and an impedance Z b .4. Being I D Vb 1 Zb C Zc V D Vb Zc Zb C Zc (4. from (1. If v and i are.− / i.1. f /] (W/Hz) (4.3) t!1 t t=2 assuming that v. If Pvi is the crossspectral density between v and i.3 that consists of a generator with an opencircuit voltage vb .t/ i.5) is called average power density transferred to the load and expresses the average power per unit of frequency. the average power (W) transferred to the load is deﬁned by the relation:1 Z 1 t=2 P D lim v. The device is connected to a load with impedance Z c . Active twoterminal device (voltage generator) connected to a load.t/] (4. f / D Re[Pvi . the crosscorrelation rvi is a real function. respectively.4) PD 1 1 In fact. We now obtain Pvi in terms of Pvb using the method shown on page 49. we have that: Z C1 Z C1 Pvi .− / i.t/i. f / d f D Re[Pvi .6) Figure 4. P is sometimes deﬁned as PD 1 1 lim 2 t!1 t Z t=2 t=2 v.230).− / d− D E[v.3. Deﬁnition 4.t/ is an ergodic process in mean (see (1.442)).2) . f /] d f (4. 1 In propagation theory [1].1 The function p. modelled as a random WSS process with statistical power spectral density Pvb (V2 /Hz). Electrical characterization of a transmission system 257 Characterization of an active device We consider the twoterminal device of Figure 4. Hence Pvi is Hermitian with even real part and odd imaginary part.− / d− (4.
f / 4G b (4. f / D Pv .9) Zc jZ b C Z c j2 (4.4.9) with respect to Z c and is obtained for Z c D Z b : pd . the following relation holds p. f / D Pvb . f / 4Rb Rb D Re[Z b ] (4.7) hence Pvi . Transmission media then Ł V I Ł D Vb Vb Zc jZ b C Z c j2 (4.12) where G b D Re[Yb ]. For the circuit of Figure 4.11) is a parameter of the active device that expresses the maximum power per unit of frequency that can be delivered to the load.8) In general. that Ł consists of a current source with admittance Yb D G b C j Bb and a load with Yc D Yb . . and Pib (A2 /Hz) is the PSD of the signal i b . if v is the voltage at the load impedance Z c . the available power per unit of frequency is given by pd . f / D Pvb . f / D Pvb .2 The available power per unit of frequency of an active twoterminal device is deﬁned as Ł the maximum of (4. f / Rc jZ b C Z c j2 Rc D Re[Z c ] (4. f / D Pib . f / Rc jZ c j2 (4.258 Chapter 4.4.11) We note that (4. f / and p. Figure 4. Active twoterminal device (current generator) connected to a load.10) Deﬁnition 4. Active twoterminal device as a current generator.
f / Zc. f / Zi . f / (4.1. In a connection between a source and a load.2. According to Heaviside conditions (1.144). f / C Zb. the condition for maximum transfer of power Ł Z c D Z b is easily veriﬁed. 2. Electrical characterization of a transmission system 259 We have a simple relation between the circuit of Figure 4.11 on page 46). f / Z2. f / D K1 Zc. f / and (4.4. f / D Ł Z b . the conditions for the absence of signal distortion. f / D Zc.14) 1 Rb D Zb jZ b j2 j Xb jZ b j2 (4. as well as for maximum transfer of power to the load. f / ZL. regarding the impedances as complexvalued functions of the frequency within the passband B. For a broadband signal vb . the only way to verify the conditions Z c . for which the impedances are regarded as complexvalued constants within the passband B.18) .3 and consider the relation between the load voltage v and the source voltage vb .16) is that both the load impedance (Rc ) and the source impedance (Rb ) are purely resistive. 1.4 and that of Figure 4. f / D Z1. Distortion is not a problem because G. the frequency responses of the various blocks are given by Gi . f / C Z L .10 on page 29 and 1. Characterization of a 2port network With reference to the circuit of Figure 4. f / V. f / C Z 1. the voltage vb is transferred to the load without distortion if Zb. in this case the frequency response is given by G. f / f 2B (4.3 for Z b D Rb C j X b .17) (4. f/ D Vb . f / f 2B (4.15) where B is the passband of vb (see Deﬁnitions 1.16) where K 1 is a constant. are veriﬁed in the following two cases. from Norton theorem we get Yb D and Ib D Vb 1 Zb (4. For a narrowband signal vb . Note that for Rb D Rc also the condition for maximum transfer of power is satisﬁed. and the phase term is equivalent to a delay. f / is a complex constant in the passband of vb .13) Conditions for the absence of signal distortion We now return to the circuit of Figure 4. f / D GL .
f / we get g. f / D Pv L . deﬁned as gt . Note that also in this case the presence of a constant delay factor is possible. Transmission media and G1 o. f / (4.26) where pi. f /j2 Pv1 . f / D Pv1 . f / 1 jZ 1 j2 RL R L jZ 1 j2 D jGCh . f / (4.25) (4. f /G1 Vi o. f / D jGCh .d . f / (4.260 Chapter 4. f /j2 R1 jZ L j2 jZ L j2 Pv1 .d . f / and po . f / V1 . f / D po . f / D VL D Gi . Deﬁnition 4. f / pi . f / D po . f / R1 (4. f/ D Vo . f / 2 jZ 1 j jZ 1 C Z i j2 (4. f / is the available power per unit of frequency at the source. and observing (see Figure 4. f / and po .24) In the presence of match for maximum transfer of power only at the source. f / be the average power densities of source and load.21) Deﬁnition 4.4 A 2port network is said to be perfectly matched if the conditions for maximum transfer of power are established at the source as well as at the load: Z 1 D Z iŁ and Ł Z L D Z2 (4. f / (4.27) .19) The conditions for the absence of distortion between vi and v L are veriﬁed if G. f / pi. f / RL RL D Pvo .23) Using the expressions of the various frequency responses. f / D Pv L .3 The network power gain is deﬁned as the ratio g. we introduce the notion of transducer gain. f / jZ L j2 jZ L C Z 2 j2 (4. f /G L .2c) Pv L .20) is a constant in the passband of vi . Let pi .22) R1 R1 D Pvi . respectively: pi .
f / D Go constant f 2B (4.30) In particular for Z 1 D Z 2 . that is when input and output network impedances coincide.28) (4.ad /dB D .33) Deﬁnition 4.d .gd /dB D 10 log10 gd and .t/ where.4.25) we get gd . .31) are not veriﬁed. f / 1 4Ri 1 4R2 (4.d .ad /. f / D Pvi . we speak of available attenuation of the network : 1 (4.t/ D Go Ž.25)).34) (4.36) We note that. Go D 1 p gd D p ad (4. the impulse response is given by gCh . in case the conditions leading to (4. the relation between Go and gd is more complicated (see (4.d . observing (4.32) ad D gd In dB. gd . f / (4. f / po. .31). for an ideal distortionless network with power gain (attenuation) gd . in this case. If instead gd < 1 then the network is passive. f / D jGCh . f /j2 (4. f / pi. we will assume that the frequency response of the network is GCh .31) If in the passband B of vi we have gd > 1 the network is said to be active.d .35) Consequently.1. f / D po.gd /dB (4.d . f /. Electrical characterization of a transmission system 261 In this case the powers become available powers: žat the source žat the load pi. from (4.37) (4.29) Deﬁnition 4.5 The available power gain is deﬁned as the ratio between po. f / D Pvo .6 Apart from a possible delay.d . f / and pi.
1.P/dBW D 10 log10 . Transmission media Measurement of signal power Typically gd and ad are expressed in dB.P/dBm D 10 log10 .P/dBm 30 (4.40) 3 dBW. we note that . or dBrn: .1 For P D 0:5 W we have .Go /d B D . which expresses the power in dBrn of a signal ﬁltered according to the mask given in Figure 4. With reference to (4. The ﬁlter reﬂects the perception of the human ear and is known as Cmessage weighting. the power P is expressed in W.P in mW/ .42) For telephone signals.P/dBrn D 10 log10 Some relations are . dBm.gd /d B . Reproduced with permission of Lucent Technologies. Frequency weighting known as Cmessage weighting.P in pW/ 3 W).37). In fact.Go /d B D 20 log10 Go D 10 log10 gd D . as Go denotes a ratio of voltages.262 Chapter 4. Inc.gd /d B (4. mW (10 or pW (10 12 W).5.P/dBW D . a further power unit is given by dBrnc.39) (4.P/dBW D 120 D .5 [2].P in W/ .] ./Bell Labs. and . it follows that .41) .38) (4. Figure 4. [ c 1982 Bell Telephone Laboratories. or in dBW. (4.P/dBm D 27 dBm.P/dBrn Example 4.
at an absolute temperature of T Kelvin.4.43) J/K is the Boltzmann constant and Â hf Ã 1 hf . . Such noise is very important because it determines the limits of the system. the sources of the two motions do not interact with each other. Between two consecutive collisions the electron produces a current that is proportional to the projection of the velocity onto the axis of the conductor. the typical behavior is represented in Figure 4. In addition to interference caused by electromagnetic coupling between various system elements and noise coming from the surrounding environment. there is also noise generated by the transmission devices themselves. the power spectral density of the open circuit voltage w at the conductor terminals is given by Pw . f / D 2kTR . Thermal noise Thermal noise is a phenomenon associated with Brownian or random motion of electrons in a conductor.6. Representation of electron motion and current produced by the motion. an orderly motion is superimposed on the disorderly motion of electrons.6a where the changes in the direction of the electron motion are determined by random collisions with atoms at the set of instants ftk g.2. Actually. if we represent the motion of an electron within a conductor in a twodimensional plane. Noise generated by electrical devices and networks 263 4.6b. For a conductor of resistance R. For example. If a current ﬂows through the conductor. f / where k D 1:3805 Ð 10 23 (4. its motion between collisions with atoms produces a short impulse of current. f/ D e kT 1 kT (4.2 Noise generated by electrical devices and networks Various noise and disturbance signals are added to the desired signal at different points of a transmission system. Although the average value (DC component) is zero. As each electron carries a unit charge. We will analyze two types of noise generated by transmission devices: thermal noise and shot noise.44) Figure 4. the behavior of instantaneous current for the path of Figure 4.6a is illustrated in Figure 4. the large number of electrons and collisions gives origin to a measurable alternating component.
8.43). we get . f /. f / D 2kTR (4. only the resistive component of the impedance gives origin to thermal noise. the available noise power per Figure 4.t/ with PSD 1 P j .2 Because at each instant the noise voltage w. Let us consider the scheme of Figure 4. i. f / D 2kT R . a suitable model for the amplitude distribution of w. Therefore the PSD of w is approximately white. Observing (4.8.t/ is the Gaussian distribution with zero mean. Pw . 2 An equivalent model assumes a noiseless conductor in parallel to a generator of noise current j . (a) Electrical circuit (b) Equivalent scheme Figure 4. where a conductor is modelled as a noiseless device having in series a generator of noise voltage w.t/ is due to the superposition of several current pulses. for f − kT= h D 6 Ð 1012 Hz (at room temperature T D 290 K). In the case of a linear twoterminal device with impedance Z D R C j X at absolute temperature T. Transmission media where h D 6:6262 Ð 10 34 Js is the Planck constant.264 Chapter 4. We note that. Electrical circuit to measure the available source noise power to the load. Electrical model of a noisy conductor.7.e.7.45) We adopt the electrical model of Figure 4. f / ' 1.11). where R D Re[Z ]. . In other words. the spectral density of the open circuit voltage w is still given by (4. where a noisy impedance Z D R C j X is matched to the load for maximum transfer of power. Note that the variance is very large. because of the wide support of Pw .
but rather by an equivalent function called noise temperature.d .Pw /dBm D 174 C 10 log10 B (dBm) (4.49) We also note from (4.d .48) We note that a noisy impedance produces an open circuit voltage w with a root meansquare (rms) value equal to p p p ¦w D Pw 2B D pw.8 has a bandwidth B. Noise generated by electrical devices and networks 265 unit of frequency is given by pw. the power delivered to the load is equal to Pw D kT 2B D kTB (W) 2 (4.4. can also be modelled as Gaussian noise with a constant PSD given by Pishot . For T D 290 K.46) At room temperature T D 290 K. in this case it is convenient to use the electrical model of Figure 4. pw.d be the available power per unit of frequency. due to the presence of noise in a device. In any case. the total output noise power of a device is usually not described by p. f / D pw.d 4R2B D kT 4R B (V) (4.50) Shot noise Most devices are affected by shot noise.pw. f / D kT (W/Hz) 2 21 (4. which is due to the discrete nature of electron ﬂow: also in this case the noise represents the instantaneous random deviation of current or voltage from the average value.4. Shot noise. The noise temperature is deﬁned as Tw .52) .51) where e D 1:6 Ð 10 19 C is the electron charge and I is the average current that ﬂows through the device. Noise temperature of a twoterminal device Let pw. f / D 2 Ð 10 . expressed as a current signal.d .48) that the total available power of a thermal noise source is proportional to the product of the system bandwidth and the absolute temperature of the source.47) 177 (dBm/Hz) If the circuit of Figure 4. f / k=2 (4. in [2] shot noise is evaluated for a junction diode and shot and thermal noise for a transistor. f / D eI (A2 /Hz) (4.d . .2. Noise in diodes and transistors Models are given in the literature to describe the different noise sources in electronic devices. Speciﬁcally. f //dBm D (W/Hz) and (4.
Assuming that the source. f / D gd . the network also introduces noise. we will have a total output noise Figure 4. .S/ is equal to wo D wo. with noise temperature T S . Tw represents the absolute temperature that a thermal noise source should have in order to produce the same available noise power as the device. where both the source impedance and the load impedance are matched for maximum transfer of power. which if measured at the output is equal to wo. with available power pwo.53) pwo.30).A/ . This concept can be extended and applied to the output of an ampliﬁer or an antenna. the noise voltage generated at the network output because of the presence of wi .9a. and gd is the available power gain of the 2port network deﬁned in (4. expressing the noise power in terms of effective noise temperature. We note that if a device at absolute temperature T contains more than one noise source. with available power at the load given by kT S (4.9. Transmission media In other words.S/ . Noise temperature of a 2port network We will consider the circuit of Figure 4. Noise source connected to a noisy 2port network: three equivalent models. generates a noise voltage wi .266 Chapter 4.A/ .S/ . then Tw > T. f / 2 If in addition to the source.S/ .
57) Deﬁnition 4.e. f /Twi Then p wo . f /.A/ . f /.55) and denotes the temperature of a thermal noise source connected to a 2port noiseless network that produces the same output noise power. f / D k Tw 2 o (4. The effects on the load for the three schemes of Figure 4. f / 2 (4. Note that the dependency on frequency of T A and T S is determined by gd . p wo .7 The effective noise temperature T A of the 2port network is deﬁned as TA. f / k 2 (4. f /.9 The effective output temperature of a system consisting of a source connected to a 2port network is Two D gd . f / D gd . f / gd . Assuming the two noise signals wo. Then (4.4. f / [T S C T A ] 2 (4.56) Deﬁnition 4.A/ . the available power at the load will be equal to the sum of the two powers.2. on the other hand. considers all noise sources at the output.A/ .S/ C wo. we introduce the equivalent circuits illustrated in Figure 4.8 The effective input temperature of a system consisting of a source connected to a 2port network is Twi D T S C T A (4. i.9 are the same. .9c. 3 To simplify the notation we have omitted indicating the dependency on frequency of all noise temperatures. f / D g d .54) becomes:3 k pwo . In particular the scheme of Figure 4.A/ are uncorrelated.9b and 4. The scheme of Figure 4.58) Equivalentnoise models By the previous considerations.A/ . f / kT S C pwo.S/ and wo.59) (4.9c.S/ . Noise generated by electrical devices and networks 267 signal given by wo D wo. and pwi.54) Deﬁnition 4. pwo.9b assumes the network to be noiseless and an equivalent noise source is considered at the input. f / D pwo.
61) the noise power of the 2port network can be expressed as pwo. a useful relation to determine F. To determine the noise ﬁgure let us assume as source an impedance.A/ .63) From the above considerations we deduce that to describe the noise of an active 2port network. for which gd < 1. f / kT0 . f / D 1 C TA T0 (4. as the source and network noise signals are generated by uncorrelated phenomena.S0 / 4 F. f / Pw 0 . equivalent to (4. f / D1C Pw 0 .61).S0 / . f / (4. From (4. f / D Pwo. This is obtained by disconnecting the source and setting as an input to the 2port network an impedance Z i equal to the source impedance. f / o. Transmission media Noise ﬁgure of a 2port network Usually the noise of a 2port network is not directly characterized through T A .61) D1C Being pwo. f / Pwo . f / pwo.62) We note that F is always greater than 1 and it equals 1 in the ideal case of a noiseless 2port network.A/ the expression (4.A/ .S / (4. f / pwo. F is a parameter of the network and does not depend on the noise temperature of the source to which it is connected. Applying Thevenin theorem to the network output. We now see that for a passive network at temperature T0 . Recognizing that pwo. as for example a transmission line.A/ C pwo.S 0 / will only be thermal noise with noise temperature equal to T0 . we obtain 2 the important relation F.S0 / and that due only to the source pwo. Let us consider a passive network at temperature T0 .S / o.9)) F. Moreover. which is matched to the network for maximum transfer of power. leads to the following experiment. we must assign the gain gd and the noise ﬁgure F (or equivalently the noise temperature T A ). at temperature T0 .A/ . the system is equivalent to a twoterminal device with impedance Z 2 at temperature T0 . but through the noise ﬁgure F. We set the source at a noise temperature equal to the room temperature: T S 0 D T0 D 290 K. f / pwo.S0 / .S0 / . f / D k .268 Chapter 4. now the noise wi . 4 Given an electrical circuit. f / D gd .A/ does not depend on T S . it is sufﬁcient to assign only one of the two parameters.55). and substituting for pwo. The noise ﬁgure is given by the ratio between the available power at the load due to the total noise power pwo D pwo. is given by (see (4.60) . f / D p wo .F 2 1/T0 gd (4. that employs the PSDs of the output noise signals.
f / D .61) we have F.e. where the antenna is modelled as a resistance with noise temperature T S .10a. f / D 1 D ad gd (4. T A .F and the available noise power at the load is given by p w0 . we can determine the effective noise temperature of the network. Antennapreampliﬁer conﬁguration and electrical model.1 Let us consider the conﬁguration of Figure 4.4. Hence from the ﬁrst of (4. f / kTwi 2 (4.65) Example 4.S0 / . i.66) 1/T0 (4.2. from (4. Twi is given by (4. f / D g d .10b. where an antenna with noise temperature T S is connected to a preampliﬁer with available power gain g and noise ﬁgure F. An electrical model of the connection is given in Figure 4. Noise generated by electrical devices and networks 269 Assuming the load is matched for maximum transfer of power. Z 2 D Z Ł . f / D gd pwi. in a connection between a source and a 2port network. and pw0.kT0 =2/.10. If the impedances of the two devices are matched.2.62).46) L at the output we have pw0 . given F. pwi. Summarizing. Note that also in this case. f / D .65). . On the other hand. the effective input temperature of the system can be expressed as Twi D T S C T A D T S C .S0 / .64) where ad is the power attenuation of the network. according to (4.kT0 =2/. (a) (b) Figure 4.S0 / .
we consider the cascade of two 2port networks A1 and A2 . f / Simplifying (4. Then the noise ﬁgure of the overall network is given by pwo. we wish to determine the parameters of a network equivalent to the cascade of the two networks. i D 1.270 Chapter 4. characterized by gains gi and noise ﬁgures Fi .S0 / . the noise power at the output of the ﬁrst network is given by pwo. we obtain Friis formula of the total noise ﬁgure F D F1 C . the overall network has a gain g equal to the product of the gains of the individual networks: g D g1 g2 (4.F2 1/ . with available power gains g1 and g2 and noise ﬁgures F1 and F2 .68) At the output of the second network we have pwo. f /g2 C k .1 . from (4. N .67) With regard to the noise characteristics. f / D kT0 F1 g1 2 (4. Ai . f / FD D pwo.72) 1 Figure 4.2 .2. .F2 2 2 kT0 g1 g2 2 1/g2 (4. Transmission media Cascade of 2port networks As shown in Figure 4. For a source at room temperature T0 .2 .1 .11.F3 1/ . it is sufﬁcient to determine the noise ﬁgure of the cascade of the two networks. : : : .71) kT0 kT0 g1 g2 F1 C .F2 2 1/T0 g2 (4.69) using (4.F N 1/ C C ÐÐÐC g1 g1 g2 g1 g2 : : : g N (4.63) to express the noise power due to the second network only. Equivalent scheme of a cascade of two 2port networks.F2 1/ g1 (4. respectively. Assuming the impedances are matched for maximum transfer of power between different networks. With regard to the power gain.11.70) Extending this result to the cascade of N 2port networks.70) we get F D F1 C . f / D pwo.66) for T S D T0 .
. is called a repeater section. Then.75) Therefore the N sections have overall unit gain and noise ﬁgure F D Fsr C .2. in (4. with gain g A and noise ﬁgure F A . Substituting (4. To compensate for the attenuation of the cable we choose g A D ac . cascaded with an ampliﬁer. N . i D 1.76) 1 gA D 1 ac (4. : : : . Each section of the cable. in particular.72). We note that the output noise power of N repeater sections is N times the noise power introduced by an individual section.77) where Fsr is given by (4. we have that the equivalent noise temperature of the cascade of N 2port networks.Fsr 1/ C 1 ' N Fsr (4. with power attenuation ac and noise ﬁgure Fc D ac (see (4.74) Example 4.64)).Fsr 1/ .2.2 The idealized conﬁguration of a transmission medium consisting of a very long cable where ampliﬁers are inserted at equally spaced points is illustrated in Figure 4. the more F will be reduced.12. each section has a gain gsr D and noise ﬁgure Fsr D Fc C FA 1 D ac C ac .62). Transmission channel composed of N repeater sections.12. that relates the noise ﬁgure to the effective noise temperature. Noise generated by electrical devices and networks 271 We observe that F strongly depends on the gain and noise ﬁgure parameters of the ﬁrst stages.4.73) 1 Obviously the total gain of the cascade is given by g D g1 g2 : : : g N (4. Figure 4.F A gc 1/ D g A F A (4.Fsr 1/ C ÐÐÐ C gsr gsr gsr : : : gsr D N . is given by T A D T A1 C T A2 T A3 T AN C C ÐÐÐ C g1 g1 g2 g1 g2 : : : g N (4.76). characterized by noise temperatures T Ai . the smaller F1 and the larger g1 .
t/] D 2 . Therefore we also introduce the following signaltonoise ratio of average powers Z C1 ps .t/ (4. assuming pw is constant within the passband of w.84) 2 and. f / D Pw . if the term Rc =jZ b C Z c j2 is a Ł constant within the passband of s and w.3. f / D Ps . deﬁned as the ratio of the statistical powers Z C1 Ps . 3D Ps E[s 2 .t/ D s. one of the most widely used methods considers the signaltonoise ratio (SNR).78) To measure the level of the desired signal with respect to the noise. 2 D that is the condition for maximum transfer of power is satisﬁed. f / Rc jZ b C Z c j2 Rc jZ b C Z c j2 (4. where the source vb generates a desired signal s and a noise signal w: vb .85) where Ps is the available average power of the desired signal. from (4. from (4.3 Signaltonoise ratio (SNR) SNR for a twoterminal device Let us consider the circuit of Figure 4.4Rb /.t/] kTw B E[w (4.272 Chapter 4. the effects of the two signals on a certain load Z c are measured by the average powers.80). ps . Hence it is sufﬁcient that Rb is constant within the passband of s and w to have 3 D 3s D 3 p (4. f / d f 1 where. with bandwidth B.79) Mw E[w 2 .t/ C w.83) Moreover. then the two SNRs coincide. Later we will often use this relation.t/] 1 3s D D Z C1 D (4.t/] Pw .82) pw . we have k Pw D Tw 2B (4. f / d f Ps 1 D Z C1 3p D (4. and Tw is the noise temperature of w.9). f / d f Ms E[s 2 . Transmission media 4.80) Pw pw . . Note that if Z b D Z c . f / d f 1 On the other hand. then Rc =jZ b C Z c j 1=. However.81) (4. f / Therefore 3s and 3 p are in general different.
t/] D o 2 .88) We indicate with B the passband of the network frequency response. we observe that the power of wi could be very high if Twi is constant over a wide band.t/ C wo .86) and wi has an effective noise temperature Twi D T S C T A .2b. respectively. From the expressions (4.4. The open circuit voltage of the network output is given by vo . From (4. Under matched load conditions (that is Ł Z L D Z 2 ). and with B its bandwidth.89) 1 B and Z P wo D 2 B pwi .t/ D so .90) Assuming now that g.3. Therefore pwi . With reference to the above conﬁguration.t/ C wi .t/] kTwi B E[wo (4.50). where vi has a desired component s and a noise component wi (see Figure 4.93). f / d f (4. and Twi D T S C . we have k Tw g2B (4.T S D T0 / (4.91) and (4. f /g. f / d f (4.t/ D s. usually equal to or including the passband of s. Finally we get Pso D Ps g and Pwo D 3out D 2 Ps E[so .T S D T0 / In (4.83) holds. BjMHz denotes the bandwidth in MHz.Pwo /dBm D .t/ (4.87) where so and wo depend on s and wi .Pwi /dBm D 114 C 10 log10 B jMHz C.Pwi /dBm C . at the network output we obtain 3out D 2 Ps E[so .F/d B .94) .t/ (4. f /g.9b): vi . f / is constant within B.Twi D FT0 / equal to .4) and (4. (4.t/] P wo E[wo (4. and assuming (4. but wo has much smaller power since its passband coincides with that of the network frequency response.t/] D 2 . f / D kTwi =2.g/d B .30) Z C1 Z Pso D pso .F 1/T0 is the effective noise temperature including both the source and the 2port network. f / d f D 2 ps .93) and the average power of the effective output noise is given by .91) 2 i assuming that also the source is matched for maximum transfer of power. the effective input noise due to the connection sourcenetwork has an average power for T S D T0 .92) where Ps is the available power of the desired signal at the network input. Signaltonoise ratio (SNR) 273 SNR for a 2port network Let us consider now the connection of a source to the linear 2port network of Figure 4.
S0 / .95) 1. The two receiver ampliﬁers can be modelled as one ampliﬁer with gain: .S0 / .g A /d B D .274 Chapter 4. Transmission media Example 4.100) .F2 1/T0 . 2. f / pso .Pso =Pwo / ½ 20 dB we get .T S C T A /g A B D 1:38 ð 10 D 2:91 ð 10 23 5 .98) Relation between noise ﬁgure and SNR For a source at room temperature T S D T0 .3. the average power of the thermal noise at the receiver output. it follows Ps ½ 73 W (4. We want to ﬁnd: 1. Given the average power of the noise generated by the source at room temperature Pwi.96) (4.91) the average power of the output noise is Pwo D k. receiving signals from a satellite. The transmitted signal bandwidth is 1 MHz.Pso =Pwo / ½ 100. it can be shown that FD ps .1010=10 1/290 D T A1 C D 125 C D 151 K g1 g1 1020=10 (4. As Pso D Ps gsat . f / (4.g2 /d B D 20 C 80 D 100 dB and effective noise temperature: T A D T A1 C T A2 .S0 / D kT0 2B 2 (4. f / is a constant within the passband B of the network. From 3out D .60 C 151/ 10100=10 106 15:36 dBm WD (4.1=a` / gant g A D Ps 10 44=10 . The satellite has an antenna with a power gain of gsat D 6 dB and the total attenuation a` due to the distance between transmitter and receiver is 190 dB.1 A station. given that pwi. the minimum power of the signal transmitted by the satellite to obtain a SNR of 20 dB at the receiver output.g1 /d B C .99) A more useful relation is obtained assuming that g. The preampliﬁer is followed by an ampliﬁer with a noise ﬁgure F2 of 10 dB and a gain g2 of 80 dB. f /=pwo . f / D kT0 =2. The antenna feeds a preampliﬁer with a noise temperature T A1 of 125 K and a gain g1 of 20 dB.97) 2. has an antenna with gain gant of 40 dB and a noise temperature T S of 60 K (that is the antenna acts as a noisy resistor at a temperature of 60 K). f /=pwi. From (4.
4. We now develop the basic transmission line theory.4 4.1 the typical values of F. and gain g are given for three devices.1 Transmission lines Fundamentals of transmission line theory In this section.T S D T0 / F In other words.13. A uniform transmission line consists of a twoconductor cable with a uniform crosssection.4. let Figure 4. 1].4.7 7. In the last column the frequency range usually considered for the operations of each device is also given. With reference to Figure 4. Examples of transmission lines are twistedpair cables and coaxial cables. the principles of signal propagation in transmission lines are brieﬂy reviewed. from (4. that supports the propagation of transverse electromagnetic (TEM) waves [3.102) . . 3out D Ps Pwi.13.S0 / (4. Transmission lines 275 Table 4.1 Parameters of three devices.101) 4. F is a measure of the reduction of the SNR at the output due to the noise introduced by the network.0 T A (K) 11 250 1163 g (dB) 20 ł 30 20 ł 30 50 Frequency 6 GHz 3 GHz Ä70 MHz and 3in D then. Device maser TWT ampliﬁer IC ampliﬁer F (dB) 0.92). T A . we have 3in (4.16 2. which illustrates a uniform line. In Table 4. Uniform transmission line of length L.
104) Substituting @ 2 i=@t@ x in the ﬁrst equation with the expression obtained from the second. t/ and i D i. however. Primary constants are in general slowly timevarying functions of the frequency. conductance and capacitance of the line per unit length. depicted in Figure 4. They deﬁne.x. x denote the distance from the origin and L be the length of the line.14. inductance.x. `. resistance. The parameters r. g. To determine the law that establishes the voltage and current along the line. t/ and i.105) .14. respectively. The model of Figure 4. Ideal transmission line We initially assume an ideal lossless transmission line characterized by r D g D 0. we obtain 8 > @ 2v > > < @x2 D > > @ 2i > : D @t@ x @i @t @v . t/ as a function of distance x. let us consider a uniform line segment of inﬁnitesimal length that we assume to be time invariant. t/ be. c are known as primary constants of the line. they will be considered time invariant.` dx/ (4.x. The termination is found at distance x D 0 and the signal source at x D L.x. Line segment of inﬁnitesimal length dx. we get the wave equation @ 2v 1 @ 2v @ 2v D `c 2 D 2 2 @x2 @t ¹ @t (4. Voltage and current variations in the segment dx are given by 8 > @v dx D > < @x > @i > : dx D @x Differentiating the ﬁrst equation with respect time.103) to distance and the second with respect to @ 2i @ x@t @ 2v c 2 @t ` (4.14 is obtained using the ﬁrst order Taylor series expansion of v.cdx/ @t .276 Chapter 4. the voltage and current at distance x at time t. Let v D v. respectively. in this context. Transmission media i rdx ldx i+ ð i dx ðx v gdx cdx v+ ð v dx ðx Figure 4.
t/ D 1 h '1 t `¹ xÁ ¹ '2 t C x Ái C '. t/ D 1 h '1 t Z0 xÁ ¹ '2 t C x Ái ¹ (4.106) yields ` @i D @t 1 0 ' t ¹ 1 xÁ 1 0 xÁ C '2 t C ¹ ¹ ¹ . that propagating in the negative direction is given by v . respectively.108) where '. consists of two waves that propagate in opposite directions: the wave that propagates from the source to the line termination is called the source or incident wave.x. Transmission lines 277 p where ¹ D 1= `c represents the velocity of propagation of the signal on a lossless transmission line. t/ D cos ! t Z0 x Ái ¹ i h jV j xÁ C Âp cos ! t C Z0 ¹ (4. The general solution to the wave equation for a lossless transmission line is given by xÁ xÁ C '2 t C (4.x.x.!t/ (4.t C x=¹/ C Â p ].109) c the expression for the current is given by i. The transmission line voltage is obtained as the sum of the two components and is given by i h h x Ái xÁ (4.110) From the general solution to the wave equation we ﬁnd that the voltage (or the current). We consider now the propagation of a sinusoidal wave with frequency f D !=2³ in an ideal transmission line.x.1=`/@v=@ x.113) . t/ D jV j cos[!. Noting that from (4. t/ D jVC j cos ! t ¹ ¹ The current has the expression h jVC j i.x/ ¹ (4. The voltage at distance x D 0 is given by v.x/ is time independent and can therefore be ignored in the study of propagation.x. (4. Integrating by parts (4. t/ D jVC j cos[!. t/ D V0 cos.0.103) @i =@t D (4.t x=¹/].107) 0 0 where '1 and '2 are the derivatives of '1 and '2 .112) C jV j cos ! t C C Âp v. considered as a function of distance along the line.107) we get i.106) v.4. that which propagates in the opposite direction is called reﬂected wave.4.111) The wave propagating in the positive direction of x is given by vC .x.x. t/ D '1 t ¹ ¹ where '1 and '2 are arbitrary functions. Deﬁning the characteristic impedance of a lossless transmission line as r ` Z 0 D `¹ D (4.
the reﬂected and incident waves. then V D jV je jÂ p .t x=¹/ is a constant.119) At the termination. By Kirchhoff laws.278 Chapter 4. We note that frequency f and wavelength ½ are related by ½D ¹ f (4.118) þV þ Z L C Z0 C and −D 2Z L Z L C Z0 (4. where Â p is the phase rotation between the incident and the reﬂected waves at x D 0. V D VC e jþx C V e jþx 1 ID . % D V =VC .112) and (4. it turns out þ þ þV þ Z L Z0 %D D þ þ e jÂ p (4. % D 0 and there is no reﬂection.114) (4. the termination voltage and the incident wave − D VL =VC . Let us consider a transmission line having as termination an impedance Z L . This point is seen by an observer as moving at velocity ¹ in the positive direction of the xaxis. % D 1 and V D VC . the line is opencircuited. we obtain P =PC D j%j2 . Let us consider some speciﬁc cases: ž if Z L D Z 0 .115) where þ D !=¹ denotes the phase constant.117) V > I L D VL D VC : ZL Z0 Z0 The reﬂection coefﬁcient is deﬁned as the ratio between the phasors representing. respectively. ž if Z L D 1. ž if Z L D 0. the voltage and current at the termination are given by 8 > VL D VC C V < (4. % D 1 and V D VC . It is useful to write (4. the line is shortcircuited. deﬁning the incident power as PC D jVC j2 =Z 0 and the reﬂected power as P D jV j2 =Z 0 . From (4. If VC is taken as the reference phasor with phase equal to zero. We deﬁne the wavelength as ½ D 2³=þ. .112) and (4. For sinuosoidal waves the velocity for which the phase is a constant is called phase velocity ¹. The transmission coefﬁcient is deﬁned as the ratio between the phasors representing. respectively. respectively.VC e jþx V e jþx / Z0 (4. where the phasors V and I represent amplitude and phase at distance x of the sinusoidal signals (4.117). the ratio between the power delivered to the load and the incident power is hence given by 1 j%j2 .113). the propagation in free space is characterized by ¹ D c D 3 Ð 108 m/s.113) in complex notation. Transmission media Let us consider a point on the xaxis individuated at each time instant t by the condition that the argument of the function F.116) In particular.
the real and imaginary parts of : Þ is the attenuation constant measured in neper per unit of length.123) the voltage at the load can be expressed as VL D VC . Frequency response Let us consider the transmission line of Figure 4. The solution of the differential equation for the voltage can be expressed in terms of exponential functions as V D VC e x 2 V (4.15. and þ is the phase constant measured in radians per unit of length.4. respectively.1 C %/.120) Differentiating and substituting in the ﬁrst equation the expression of d I =dx obtained from the second. we get d2V D dx 2 where p D ZY (4.4. with a sinusoidal voltage source vi and a load Z L .125) 1 VC e Z0 x V e x Ð (4.1 C %/ j%D1 D 2VC . The propagation constant and the characteristic impedance are also known as secondary constants of the transmission line. Transmission lines 279 Nonideal transmission line Typically.123) The expression of the current is given by I D where r Z0 D Z Y (4. Recalling that V = VC D %. For sinusoidal waves in steady state.121) CV e x D VC e Þx e jþx C V eÞx e jþx (4. in a transmission line the primary constants r and g are different from zero.122) is a characteristic constant of the transmission line called propagation constant. the changes in voltage and current in a line segment of inﬁnitesimal length characterized by an impedance Z and an admittance Y per unit length can be expressed using complex notation as 8 dV > > < dx D Z I > dI > : D YV dx (4. From (4.124) is the characteristic impedance of the transmission line. . we deﬁne the voltage Vo D VL j Z LD1 D VC . Let Þ and þ be.
Observing the above relations we ﬁnd the following frequency responses: GL D G1 D ZL 1 VL 1C% D D VC .126) where Z i denotes the generator impedance. Transmission media i(t) 1 Z i v 1 (t) v(t) i v L(t) ZL x = L x=0 Figure 4.1 %e 2 L/ (4.1 C %/ D Vo 2VC 2 Z L C Z0 Vo 2e D V1 1 C %e L 2 L (4. The input and output impedances of the 2port network are. respectively.e L %e L / Z0 L C %e L/ (4. For the voltage V1 and current I1 we ﬁnd 8 > V1 D Vi Z i I1 D VC .129) o (4. We now want to determine the ratio between the voltage VL and the voltage V1 .128) VC .e < V > : I1 D C . given by: Z1 D Z2 D 1 C %e V1 D Z0 I1 1 %e 2 L 2 L (4.1 C %e 2 L / C Z i .1 C %/ j%D1 Vooc D V D Z0 C I L sc %/ j%D 1 Z 0 .15.1 C %e 2 L / D D Vi Zi C Z1 Z 0 .127) (4.280 Chapter 4.1 where I L sc D I L j Z L D0 and Vooc D VL j Z L D1 .131) . deﬁned as GCh D VL = V1 .130) Gi D V1 Z1 Z 0 . Transmission line with sinusoidal voltage generator vi and load ZL .
25).136) where Þ D Re[ ].136) yields þ þ þ ZL þ þ . We note that.139).118) we get 1 C % ' 2Z L =Z 0 .ad .138) The relation between Þ and . f //d B . the attentuation in dB introduced by the transmission line is equal to . and %2 ' 1 4Z L =Z 0 .23).ad .132) (4. for a matched transmission line.ad .Gi D 1=2/ GCh D e ž Shortcircuited transmission line: % D 1 (4.4. f //d B L 1 (4.ad . from (4. Therefore (4.ad . from (4. f / D 1 je L j2 D e2ÞL (4. f //d B L 10 log10 4 þ (4.ad .1). the channel frequency response is given by: GCh D G1 Let us consider some speciﬁc cases: ž Matched transmission line: % D 0 for Z i D Z L D Z 0 . we can use the general equation (4. f / D 1 j%j2 e 1 j%j2 e 4ÞL 2ÞL (4.ad . as Q Q ad . L/ (4. f //d B D 8:68Þ Q (4.140) In a transmission line with a nonmatched resistive load that satisﬁes the condition Z L − Z 0 .135) To determine the power gain of the network. in any case.1 C %/e 1 C %e 2 L L (4. f //d B D .139) From (4. Þ expresses the attenuation in neper per unit of length. Alternatively.141) Q þZ þ 0 .137). f / D 10 10 . f //d B is given by Q .137) In (4.134) L o GL D . f //d B L Q (4.133) GCh D 0 ž Opencircuited transmission line: % D 1 GCh D 2e 1Ce L 2 L D 1 cosh. one can introduce an attenuation in dB per unit of length.ad . %2 e 4ÞL ' 0. Transmission lines 281 Then. . we obtain g. or observe (4. the available attenuation is given by ad .4. f //d B D .
these conditions are satisﬁed if Þ is a constant and þ is a linear function of the frequency. f / D K ³ f (neper/m) (4. at least within the passband of the source. using the approximation Â Ã1=2 r2 1 r2 1C 2 2 '1C 2 ! 2 `2 ! ` we ﬁnd r Þ' 2 r c ` p and þ ' ! `c (4. An expression of the propagation constant generally used to characterize the propagation of TEM waves over a metallic transmission line [1] is r r p ! ! C jK C j! `c . it can be shown that Heaviside conditions are equivalent to the condition r c D g` In the special case g D 0. shows that both the attenuation constant and the phase constant must include a term proportional to the square root of frequency. we obtain `c ÞD! 2 and `c þD! 2 r (Â r2 1C 2 2 ! ` Ã1=2 )1=2 C1 (4. For a matched transmission line. The secondary parameters of the transmission line can p p be expressed as D Þ C jþ D .r C j!`/=.145) Impulse response of a nonideal transmission line For commonly used transmission lines.148) .g C j!c/.142) For frequencies at which r − !`.147) is valid for both coaxial and twistedpair cables insulated with plastic material.282 Chapter 4. that takes into account the variation of r with the frequency due to the skin effect.147) 2 2 where K is a constant that depends on the transmission line. f/ D K (4.g C j!c/. The expression (4.146) (4. The attenuation constant of the transmission line is therefore given by p Þ. a more accurate model of the propagation constants. For a matched transmission line. Transmission media Conditions for the absence of signal distortion We recall that Heaviside conditions for the absence of signal distortion are satisﬁed if GCh . and Z 0 D . f / has a constant amplitude and a linear phase.144) r (Â r2 1C 2 2 ! ` Ã1=2 )1=2 1 (4.143) (4.r C j!`/.
08 gCh(t) 0. we may note that it follows the f < 10 kHz. f //d B D 8:68K ³ f (dB/m) Q (4. From the expression (4.1 0. .t/ (4.133) of the frequency response of a matched transmission p line.16.02 KL=5 KL=6 0 0 2 4 6 8 t (s) 10 12 14 16 Figure 4. Transmission lines 283 and the attenuation introduced by the transmission line can be expressed as p .2 we give the values of Z 0 and D Þ C jþ experimentally measured for some telephone transmission lines characterized by a certain diameter.149) We note that.12 KL=2 0. the impulse response has the following expression KL gCh .17 for four telep f law in the range of frequencies phone lines [2]. with given by (4. For some transmission lines this law is followed also for f > 100 kHz.K L/2 4t 1. we can obtain the value of K .4. Therefore it is possible to determine the attenuation constant at every other frequency. Secondary constants of some transmission lines In Table 4.4. given the value of Þ. Impulse response of a matched transmission line for various values of KL. We note a larger dispersion of gCh for increasing values of K L.06 KL=3 0.150) The pulse signal gCh is shown in Figure 4.16 for various values of the product K L.ad . f / at a certain frequency f D f 0 . 0.147). without considering the delay `c introduced by the term p j! `c. which is usually indicated by a parameter called gauge. The behavior of Þ as a function of frequency is given in Figure 4.t/ D p e 2 ³t3 .04 KL=4 0.
0:9119/ 22 . Figure 4.] . Inc. Reproduced with permission of Lucent Technologies. Inc. Attenuation as a function of frequency for some telephone transmission lines: three are polyethyleneinsulated cables (PIC) and one is a coaxial cable with a diameter of 9.0:5105/ 26 .284 Chapter 4.17. Gauge diameter (mm) 19 . Reproduced with permission of Lucent Technologies. Transmission media Table 4.0:4039/ Frequency (Hz) 1000 2000 3000 1000 2000 3000 1000 2000 3000 1000 2000 3000 Characteristic impedance Z0 ( ) 297 217 183 414 297 247 518 370 306 654 466 383 j278 j190 j150 j401 j279 j224 j507 j355 j286 j645 j453 j367 Propagation constant Þ C jþ (neper/km) (rad/km) 0:09 C 0:12 C 0:15 C 0:13 C 0:18 C 0:22 C 0:16 C 0:23 C 0:28 C 0:21 C 0:29 C 0:35 C j0:09 j0:14 j0:18 j0:14 j0:19 j0:24 j0:17 j0:24 j0:30 j0:21 j0:30 j0:37 Attenuation ad D 8:68Þ Q (dB/km) 0:78 1:07 1:27 1:13 1:57 1:90 1:43 2:00 2:42 1:81 2:55 3:10 c 1982 Telephone Laboratories./Bell Labs./Bell Labs.2 Secondary constants of some telephone lines.0:6426/ 24 . [ c 1982 Bell Telephone Laboratories.525 mm.
For DSL applications it is therefore necessary to remove possible loading coils that are present in the local loops. causes Þ.18 for a transmission line with gauge 22 [2]. The digital subscriber line (DSL) technologies. where ½ BT satisﬁes the condition . Transmission lines 285 albeit with a different constant of proportionality.2n C 1/½ BT =4 D L BT .5 to force the primary constants to satisfy Heaviside conditions in the voice band.18./Bell Labs.4. A bridgedtap consists of a twisted pair cable of a certain length L BT . but considerably increases the attenuation outside of the voice band. formerly some lump inductors were placed at equidistant points along the transmission line. . In any case in the localloop. At the frequencies f BT D ¹=½ BT . at the connection point we get destructive interference between the reﬂected and incident component: this interference reveals itself as a notch in the frequency response of the transmission line. terminated by an open circuit and connected in parallel to a local loop. Attenuation constant Þ and phase constant þ for a telephone transmission line with and without loading. At the connection point. are given in Figure 4. introduced for data transmission in the local loop. Moreover. called inductive loading. : : : . f / to be ﬂat in the voice band. Reproduced with permission of Lucent Technologies. n D 0.] 5 By localloop we intend the transmission line that goes from the user telephone set to the central ofﬁce. Given Figure 4. The component propagating along the bridgedtap is reﬂected at the point of the open circuit: the component propagating on the transmission line must therefore be calculated taking also into consideration this reﬂected component. [ c 1982 Bell Telephone Laboratories. with and without loading.4. This procedure. Typical behavior of Þ and þ in the frequency band 0 ł 4000 Hz. the phase þ. the incident signal separates into two components. 1. f / may result very distorted in the passband. Inc. The frequency response of a DSL transmission line can also be modiﬁed by the presence of one or more bridgedtaps. which goes from 300 to 3400 Hz. require a bandwidth much greater than 4 kHz. up to about 20 MHz for the VDSL technology (see Chapter 17).
and 3) the characteristic impedance. Transmission lines conﬁguration for the study of crosstalk. 4.15 dB/100 m 8. 2) the attenuation of the nearend crosstalk signal. Signal attenuation at 16 MHz UTP3 UTP4 UTP5 13.4. which can be viewed as samples taken from the ensemble of frequency responses.1. where the terminals .2. 10 / belong to the disturbing transmission line and the terminals . As illustrated in Table 4.19. the transmission characteristics of unshielded twistedpair (UTP) cables commonly used for data transmission over local area networks are deﬁned by the EIA/TIA and ISO/IEC standards.3.286 Chapter 4. Transmission media Table 4. those of categories four and ﬁve (UTP4 and UTP5) are datagrade.3 Transmission characteristics deﬁned by the EIA/TIA for unshielded twisted pair (UTP) cables. to evaluate the performance of DSL systems we usually refer to a limited number of loop characteristics. the cables are divided into different categories according to the values of 1) the signal attenuation per unit of length.20 dB/100 m NEXT attenuation at 16 MHz ½23 dB ½38 dB ½44 dB Characteristic impedance 100 100 100 š 15% š 15% š 15% the large number of transmission lines actually in use. Let us consider the two transmission lines of Figure 4. or NEXT.85 dB/100 m 8.2 Crosstalk The interference signal that is commonly referred to as crosstalk is determined by magnetic coupling and unbalanced capacitance between two adjacent transmission lines. We note that the signal attenuation and the intensity of NEXT are substantially larger for UTP3 cables than for UTP4 and UTP5 cables. .19. In the study of the interference signal produced by magnetic coupling. On the other hand. Cables of category three (UTP3) are commonly called voicegrade. 20 / belong to the disturbed transmission line. that will be deﬁned in the next section. we consider Figure 4.
2Z 0 // D . Transmission lines 287 i1 1 v1 Z0 1’ im 2 Z0 2’ Figure 4. We assume that the impedance Z 0 is much smaller than the reactance of the capacitors that can be found on the bridge. The EMF produces a current j2³ f E Im D . the circuit of Figure 4. 2 0 To study the interference signal due to unbalanced capacitance.21a. we consider the circuit of Figure 4.4.2Zm// I1 . We will assume that the length of the transmission line is much longer than the wavelength corresponding to the maximum transmitted frequency and that the impedance Z 0 is much higher than the inductor reactance.2Z 0 // .1=.20.4. Interference signal produced by magnetic coupling. that can be expressed as Im D j2³ f m V1 .1=. 1 c 11 v1 c 12 1 2 Z0 2 1 (a) (b) c1 2 c1 2 c 22 Z0 c1 2 Z0 c1 2 c 12 Z0 v1 Z 0 c 11 2 c 12 ic 1 Z0 c 22 c 12 2 m Z0 Figure 4. Interference signal produced by unbalanced capacitance. Applying the principle of the equivalent generator we ﬁnd 0 Ic D V220 j Ic D0 D Z 220 1 1 c10 2 1 B c10 20 B @ 1 1 C c10 20 c120 C 1 C j2³ f V1 A 1 1 1 1 C C c10 2 c12 c12 C c10 2 c120 C c10 20 (4. that can be redrawn in an equivalent way as illustrated in Figure 4.21b. The induced electromagnetic force (EMF) is given by E D j2³ f m I1 .151) .20.21.1=. where I1 ' V1 =Z 0 .
from which we obtain Ic D c12 c10 20 c120 c10 2 j2³ f V1 D j2³ 1cV1 c12 C c10 2 C c120 C c10 20 (4. respectively. depending on whether the receiver side of the disturbed line is the same as the transmitter side of the disturbing line.155) .x/.154) To calculate the power spectral density of NEXT we need to know the autocorrelation function of the random process a p .153) be the nearend crosstalk coupling function at distance x from the origin.x/ dx (4. Transmission media Figure 4. the interference signals are called nearend crosstalk or NEXT. Nearend crosstalk Let a p .0/Ž. As illustrated in Figure 4.x/ 1c.152) Recalling that the current Ic is equally divided between the impedances Z 0 on which the transmission line terminates.x/] D r p . with autocorrelation ra p . the NEXT signal is expressed as Z Vp D Z0 I p D 0 L V1 e 2 x j2³ f a p .x/ Z0 C 2Z 0 2 (4. Illustration of nearend crosstalk (NEXT) and farend crosstalk (FEXT) signals.x/ D m. or farend crosstalk or FEXT.22. A model commonly used in practice assumes that a p .288 Chapter 4. We now evaluate the total contribution of the near and farend crosstalk signals for lines with distributed impedances.z/ p (4. we ﬁnd that the crosstalk current produced at the transmitter side termination is I p D Im C Ic =2. and the crosstalk current produced at the receiver side termination is It D Im C Ic =2.z/ D E[a p .22.x/ is a white stationary random process. In complex notation.x C z/a Ł . or the opposite side.
i C 2 /1x r p .158) To perform computer simulations of data transmission systems over metallic lines in the presence of NEXT. f /j2 ] ³ 3=2 r p . it is required to characterize not only the amplitude. then from (4.0/ f 3=2 .162) If we know the parameters of the transmission line K and k p .0/ 1x (4.x/ D 1 0 if x 2 [0.449).0/ K (4. denote statistically independent Gaussian random variables with zero mean and variance E[ai2 ] D A NEXT coupling function is thus given by L 1x 1 X i D0 p p p 1 2. f /j2 ]k p f 3=2 (4.4. : : : . the following stochastic model is used: L 1x 1 X i D0 a p .4.160) where ai . Transmission lines 289 For NEXT the following relation holds E[jV p . i D 0. but also the phase of NEXT coupling.449).1 K e4K p ³f L / ' E[jV1 .x i1x/e (4. jG p .148). f /j2 is also equal to the ratio between the PSDs of v p and v1 .157) Using (1.x/ D with ai w1x . 1x/ otherwise (4. f / D j2³ f ai w1x .156) where K is deﬁned by (4.x i1x/ (4.161) the variance of ai to be used in the simulations is given by E[ai2 ] D K kp ³ 3=2 1x (4.157) and (4.161) GNE X T . f /j2 ] (4. In addition to experimental models obtained through laboratory measurements. and kp D ³ 3=2 r p . f /j2 ] D E[jV1 . L=1x 1.159) ( w1x . f /j2 D E[jV p .K ³ f C j K ³ f C j2³ f `c/. f /j2 ] ' k p f 3=2 E[jV1 . . the level of NEXT coupling is given by6 jG p .163) 6 Observing (1.
2³ /2 rt . The level of FEXT coupling is given by jGt .6 dB. the maximum length of cables connecting stations is typically limited to 100 m.z/ D E[a t . nonhomogeneity of the transmission line.0/L (4. The amplitude of the frequency response obtained for a cable length L D 100 m is shown in Figure 4. . which deﬁnes the physical layer for data transmission at 100 Mb/s over UTP3 cables in Ethernet LANs (see Chapter 17). f /j2 ]e p 2K ³ f L (4. f /j2 ] D E[jV1 .1 For localarea network (LAN) applications.164) be the farend crosstalk coupling function at distance x from the origin.4. etc. we assume that at is a white stationary random process. the presence of connectors. the FEXT signal is given by Z L V1 e L j2³ f at .169). For the IEEE Standard 100BASET2.165) Vt D Z 0 It D 0 Analogously to the case of NEXT.z/ For the FEXT signal the following relation holds E[jVt . f / D 10 p 1:2 20 e . We note that for highspeed data transmission systems over unshielded twistedpair cables.2 dB has been included to take into account the attenuation caused by the possible presence of connectors. the following worstcase frequency response is considered: GCh .x/] D rt .x/ Z0 C 2Z 0 2 (4. In complex notation.290 Chapter 4.167) where L is the length of the transmission line.0/.0/Ž. a higher value than that indicated in Table 4. We note that the signal attenuation at the frequency of 16 MHz is equal to 14.x C z/a tŁ . A frequency independent attenuation of 1. NEXT usually represents the dominant source of interference.x/ 1c.2³ f /2 rt . Transmission media Farend crosstalk Let at . as it indicates a constant propagation delay. f /j2 ] D kt f 2 Le E[jV1 .3 for UTP3 cables.166) . f /j2 D E[jVt .0:00385 j f C0:00028 f /L p (4. In (4.x/ dx (4.x/ D m. with autocorrelation rat .147) may be caused by losses in the dielectric material of the cable. f /j2 ] p 2K ³ f L (4. Example 4.169) where f is expressed in MHz and L in meters. the term e j2³ f `cL is ignored.168) where kt D . Deviations from the characteristic expressed by (4.23 [4].
due to the factor f 3=2 . The level of NEXT coupling equal to 21 dB at the frequency of 16 MHz is larger than that given in Table 4. [ c 1997 IEEE.24 [8. we note that the useful interval for transmission is in the range from 800 to 1600 nm. and four realizations of NEXT coupling function. Amplitude of the frequency response for a voicegrade twistedpair cable with length equal to 100 m.4. The term “optical communications” is used to indicate the transmission of information by the propagation of electromagnetic ﬁelds at frequencies typically of the order of 1014 ł 1015 Hz.158) is illustrated in Figure 4. the wavelength rather than the frequency is normally used.23 as a dotted line.3 for UTP3 cables. Optical ﬁbers 291 0 Amplitude characteristic for 100 m cable length –10 –14. 6. to identify a transmission band. 4. that are found in the optical band and are much higher than the frequency of radio waves or microwaves. For indepth study of optical ﬁber properties and of optical component characteristics we refer the reader to the vast literature existing on the subject [5. the relation (4.] The level of NEXT coupling (4. We recall that for electromagnetic wave propagation in free space. we note the increase as a function of frequency of 15 dB/decade. f in MHz –20 (dB) –30 Amplitude –40 Four NEXT coupling functions –50 –60 0 5 10 15 20 f (MHz) 25 30 35 40 Figure 4. to the point that they now constitute a fundamental element of modern information highways. 9].6 dB –21. in this section we limit ourselves to introducing some fundamental concepts.116) holds: a frequency of 3 Ð 1014 Hz corresponds therefore to a wavelength of 1 µm for transmission over optical ﬁbers.23. The signal attenuation as a function of the wavelength exhibits the behavior shown in Figure 4.23.5.0 dB 16 MHz NEXT coupling envelope curve –21 + 15 log10 ( f/16 ) .162) are also shown in Figure 4. 7]. that corresponds .5 Optical ﬁbers Transmission systems using light pulses that propagate over thin glass ﬁbers were introduced in the 1970s and have since then undergone continuous development and experienced an increasing penetration. The amplitude characteristics of four realizations of the NEXT coupling function (4.
25 [10]. Transmission media Figure 4. each with a bandwidth of 6 MHz. equivalent to that needed for the transmission of ¾300:000 television signals. multiplexing techniques using optical devices have been developed. this phenomenon in turn causes intersymbol interference and limits the available bandwidth of the transmission . has an available bandwidth of 2 Ð 1012 Hz. made coherent light sources available for the transmission of signals. [From Li (1980). because they present less attenuation with respect to ﬁbers using plastic material. we note that. (1979). Optical transmission lines with lengths of over a few hundred meters use ﬁber glass. Moreover. the majority of optical communication systems employ as transmission medium an optical ﬁber.24. beginning in the 1970s. although the propagation of electromagnetic ﬁelds in the atmosphere at these frequencies is also considered for transmission (see Section 17. Three regions are typically used for transmission: the ﬁrst window goes from 800 to 900 nm. c 1980 IEEE.] to a bandwidth of 2 Ð 1014 Hz. see also Miya et al.292 Chapter 4. the second from 1250 to 1350 nm. Dispersion in the transmission medium causes “spreading” of the transmitted pulses.1). To efﬁciently use the band in the optical spectrum. A fundamental device in optical communications is represented by the laser. which acts as a waveguide. which. a system that uses only 1% of the 2 Ð 1014 Hz bandwidth mentioned above.2. We immediately realize the enormous capacity of ﬁber transmission systems: for example. such as wavelengthdivision multiplexing (WDM) and optical frequencydivision multiplexing (OFDM). Description of a ﬁberoptic transmission system The main components of a ﬁberoptic transmission system are illustrated in Figure 4. Attenuation curve as a function of wavelength for an optical ﬁber. and the third from 1500 to 1600 nm.
respectively. Optical ﬁbers 293 Figure 4. In this case the medium introduces signal distortion caused by the fact that propagation of energy for different modes has different speeds: for this reason multimodal ﬁbers are used in applications where the transmission bandwidth and the length of the transmission line are not large. In Table 4. thus eliminating the dispersion caused by multimode propagation.M C Mg / has values near 120. Elements of a typical ﬁberoptic transmission system. whereas the gradedindex (GRIN) ﬁber has a refraction index decreasing with the distance from the ﬁber axis. A measure of the pulse dispersion is given by 1− D . we note that the dispersion is minimum in the second window. As noticed previously. medium. and 1550 nm.170) where M is the dispersion coefﬁcient of the material. The stepindex (SI) ﬁber is characterized by a constant value of the refraction index. monomodal ﬁbers are preferred for applications that require wide transmission bandwidth and very long transmission lines.5. 0. Because in this case the dispersion is due only to the material and the geometry of the waveguide. Special ﬁbers are designed to compensate for the dispersion introduced by the material.4. these ﬁbers are normally used in very long distance connections. Multimode ﬁbers allow the propagation of more than one mode of the electromagnetic ﬁeld. to limit the number of modes . L denotes the length of the ﬁber and 1½ denotes the spectral width of the light source. are given for different types of ﬁbers. and 15 ps/(nmðkm) at wavelengths of 850. Mg is the dispersion coefﬁcient related to the geometry of the waveguide. the monomodal ﬁbers are characterized by larger bandwidths. The bandwidth of the transmission medium is inversely proportional to the dispersion.25. with values near zero around the wavelength of 1300 nm for conventional ﬁbers. normalized by the length of the optical ﬁber. Monomode ﬁbers limit the propagation to a single mode. 1300.4 typical values of the transmission bandwidth. because of the low attenuation and dispersion.M C Mg / L 1½ (4. The total dispersion .
Semiconductor laser diodes (LD) or lightemitting diodes (LED) are used as signal light sources in most applications. 4. 15. and R L is the resistance of the load that follows the photodetector. Some examples of radio transmission systems are: ž pointtopoint terrestrial links [11]. and therefore lead to a lower dispersion (see (4.I D C ² P Rc / C 4kTw B (4.4 Characteristic parameters of various types of optical ﬁbers. The transmitted waveform can therefore be seen as a replica of the modulation signal. The conversion from a current signal to an electromagnetic ﬁeld that propagates along the ﬁber can be described in terms of light signal power by the relation PT x D k0 C k1 i (4. Signal quality is measured by the signaltonoise ratio expressed as 3D gin .173) where gi is the photodetector current gain. P Rc is the power of the incident optical signal and ² is the photodetector response. Fiber multimode SI multimode GRIN multimode GRIN monomode monomode Wavelength (nm) 850 850 1300 1300 1550 Source LED LD LD o LED LD LD Bandwidth (MHzÐkm) 30 500 1000 >10000 >10000 to one. 13. e is the charge of the electron. ž mobile terrestrial communication systems [12.170)). . which convert the optical signal into a current signal according to the relation i D ² P Rc (4. 14. B is the receiver bandwidth.171) where k0 and k1 are constants.gi ² P Rc /2 R L 2e R L B.6 Radio links The term radio is used to indicate the transmission of an electromagnetic ﬁeld that propagates in free space.172) where i is the device output current. The more widely used photodetector devices are semiconductor photodiodes. I D is the photodetector dark current. Typical values of ² are of the order of 0. in this case the current signal. Transmission media Table 4. Laser diodes are characterized by a smaller spectral width 1½ as compared to that of LEDs. k is Boltzmann constant.294 Chapter 4. 16]. Tw is the effective noise temperature in Kelvin. n is a parameter that indicates the photodetector excess noise. We note that in the denominator of (4. the diameter of the monomodal ﬁber is related to the wavelength and is normally about one order of magnitude smaller than that of multimodal ﬁbers. these sources are usually modulated by electronic devices.173) the ﬁrst term is due to shot noise and the second term to thermal noise.5 mA/mW.
these are phenomena that permit transmission between two points that are not in lineofsight (LOS). In fact. one of the dimensions of the antenna must be at least equal to 1=10 of the carrier wavelength.26. this gives origin to the reﬂection of electromagnetic waves. We speak of diffusion or scattering phenomena if molecules that are present in the atmosphere absorb part of the power of the incident wave and then reemit it in all directions. among which the dimensions of the transmit antenna play an important role.6. requires an antenna of at least 30 m. A radio wave usually propagates as a ground wave (or surface wave). The earth and the ionosphere form a waveguide for the electromagnetic waves. . with carrier frequency f 0 D 1 MHz and wavelength ½ D c= f 0 D 300 m.4. to achieve an efﬁcient radiation of electromagnetic energy. Very low frequency (VLF) for f 0 < 0:3 MHz. Radio link model. : : : ). In particular. pressure. The choice of the carrier frequency depends on various factors. In any case. give also origin to signal reﬂection and/or diffusion.26. Recall that. This means that an AM radio station. if the atmosphere is nonhomogeneous (in terms of temperature. ž deepspace communication systems (with space probes at a large distance from earth). 4. or as a direct wave. buildings.. etc. At these frequencies the signals propagate around the earth. where we assume that the transmit antenna input impedance and the receive antenna output impedance are matched for maximum transfer of power.6. where c is the speed of light in free space. via reﬂection and scattering in the atmosphere (or via tropospheric scattering). Radio links 295 Figure 4. A radio link model is illustrated in Figure 4. Obstacles such as mountains. the electromagnetic propagation depends on the changes of the refraction index of the medium. We will now consider the types of propagation associated with frequency bands. humidity.1 Frequency ranges for radio transmission Frequencies used for radio transmission are in the range from about 100 kHz to some tens of GHz. ž earthsatellite links (with satellites employed as signal repeaters) [17].
with peak attenuation at 60 GHz.2 Narrowband radio channel model The propagation of electromagnetic waves should be studied using Maxwell equations with appropriate boundary conditions. Transmission media Medium frequency (MF) for 0:3 < f 0 < 3 MHz. Therefore these frequencies are adopted for satellite communications. We note that. They are also employed for lineofsight transmissions. which cause the signal to propagate over long distances with large attenuations. coverage is up to about r D 13 km. where the plot represents the limit on the power spectrum of the transmitted signal with reference to the power of a nonmodulated carrier. assuming the diameter of the rain drops is of the order of the signal wavelength. Radiation masks A radio channel by itself does not set constraints on the frequency band that can be used for transmission. which cause additional signal attenuation: 1. For f 0 > 30 MHz. High frequency (HF) for 3 < f 0 < 30 MHz. 3. a ﬁlter is usually employed at the transmitter frontend. the range p covered expressed in km is r D 1:3 h: for example. if h D 100 m. which . the signal propagates through the ionosphere with small attenuation. regulatory bodies specify power radiation masks: a typical example is given in Figure 4. for our purposes a very simple model. atmospheric conditions play an important role in signal propagation. In any case. Nevertheless. if the antennas are not positioned high enough above the ground. Extremely high frequency (EHF) for f 0 > 30 GHz. to prevent interference among radio transmissions. 2. the electromagnetic ﬁeld propagates not only into the free space but also through ground waves. Super high frequency (SHF) for 3 < f 0 < 30 GHz. due to water vapor: for f 0 > 20 GHz. ionospheric and tropospheric scattering (at an altitude of 16 km or less) are present at frequencies in the range 30–60 MHz and 40–300 MHz. Ultra high frequency (UHF) for 300 MHz < f 0 < 3 GHz. due to oxygen: for f 0 > 30 GHz. If h is the height of the tower in meters. The limit to the coverage is set by the earth curvature. using high towers where the antennas are positioned to cover a wide area. However.6. The waves are reﬂected by the ionosphere at an altitude that may vary between 50 and 400 km. To comply with these limits. The waves propagate as ground waves up to a distance of 160 km.296 Chapter 4. 4. At frequencies of about 10 GHz. We note the following absorption phenomena. respectively. with peak attenuation at around 20 GHz.27. Very high frequency (VHF) for 30 < f 0 < 300 MHz. due to rain: for f 0 > 10 GHz.
174) where 4³ d 2 is the surface of a sphere of radius d that is uniformly illuminated by the antenna. Obviously. At a distance d from the antenna. usually. The deterministic model is used to evaluate the power of the received signal when there are no obstacles between the transmitter and receiver. that is in the presence of line of sight: in this case we can think of only one wave that propagates from the transmitter to the receiver. is often adequate. which uniformly radiates in all directions in the free space. On a logarithmic scale (dB) this is equivalent to a decrease of 20 dBperdecade with the distance.4.27. the power density is 80 D PT x (W/m2 ) 4³ d 2 (4. Let PT x be the power of the signal transmitted by an ideal isotropic antenna. We observe that the power density decreases with the square of the distance. Radiation mask of the GSM system with a bandwidth of 200 kHz around the carrier. GT x D 1 for an isotropic antenna. . This situation is typical of transmissions between satellites and terrestrial radio stations in the microwave frequency range (3 < f 0 < 70 GHz).6. consists in approximating an electromagnetic wave as a ray (in the optical sense). In the case of a directional antenna.175) where GT x is the transmit antenna gain. Radio links 297 Figure 4. the power density is concentrated within a cone and is given by 8 D G T x 80 D GT x PT x 4³ d 2 (4. GT x × 1 for a directional antenna.
Later.GT x /d B and .176) where P Rc is the received power. The factor Á Rc < 1 takes into account the fact that the antenna does not capture all the incident radiation. . Transmission media At the receive antenna.ad /d B coincides with the free space path loss.177) where 32:4 D 10 log10 . Observing (4. A Rc is the effective area of the receive antenna and Á Rc is the efﬁciency of the receive antenna. we will use the following deﬁnition: Â Ã2 ½ P0 D PT x GT x G Rc (4. because of the factor A=½2 .178) ½2 where A is the effective area of the antenna.G Rc /d B in dB. we get Â Ã ½ 2 (4. and . is . f 0 is the carrier frequency and Á is the efﬁciency factor. The term . (4.179) does not take into account attenuation due to rain or other environmental factors. working at higher frequencies presents the advantage of being able to use smaller antennas.181): a) it increases with distance as log10 d.179) is known as the Friis transmission equation and is valid in conditions of maximum transfer of power. whereas for metallic transmission lines the dependency is linear (see (4. nor the possibility that the antennas may not be correctly positioned. In any case.179) P Rc D PT x GT x G Rc 4³ d GD The (4.½=4³ d/2 is called free space path loss. To conclude. (4.GT x /d B . ½ D c= f 0 is the wavelength of the transmitted signal. Usually Á 2 [0:5. while Á ' 0:8 for horn antennas.ad /d B D 10 log10 PT x D 32:4 C 20 log10 djkm C 20 log10 f 0 jMHz P Rc . b) it increases with frequency as log10 f 0 .181) d is expressed in km. It is worthwhile making the following observations on the attenuation ad expressed by (4. the power of the received signal is given by P Rc D PT x The antenna gain can be expressed as [1] 4³ A Á (4. 0:6] for parabolic antennas.180) 4³ which represents the power of a signal received at the distance of 1 meter from the transmitter.G Rc /d B A Rc GT x Á Rc 4³ d 2 (4.140)). expressed in dB. We note that. f 0 in MHz. because a part is reﬂected or lost. for a given G. For GT x D G Rc D 1.178) holds for the transmit as well as for the receive antenna.298 Chapter 4.4³=c/2 . . The available attenuation of the medium.178). The (4. the available power in conditions of matched impedance is given by P Rc D 8A Rc Á Rc (4.
92).atmosphere (4. Radio links 299 Equivalent circuit at the receiver We redraw in Figure 4.182) We note that there are two noise sources.F 1/T0 is the noise temperature of the ampliﬁer. The noise temperature of the antenna depends on the direction in which the antenna is pointed: for example T S.k=2/Tw . which implies using a directional antenna. introduced by the antenna (w S ) and by the receiver (w A ). The spectral density of the open circuit noise voltage is Pw .28 the electrical equivalent circuit at the receiver. Electrical equivalent circuit at the receiver.28.183) Multipath It is useful to study the propagation of a sinusoidal signal hypothesizing that the oneray model is adequate. The effective noise temperature at the input is Tw D T S C .6. From (4. for matched input and output circuits. Let sT x be a narrowband Figure 4. the signaltonoise ratio at the ampliﬁer output is equal to 3D available power of received desired signal P Rc D kTw B available power of effective input noise (4. f / D . and w represents the total noise due to the antenna and the ampliﬁer.10.4. and the available noise power per unit of frequency is pw . using a slightly different notation from that of Figure 4. and T A D . where T S is the effective noise temperature of the antenna.Sun > T S. f / D 2kTw Ri .F 1/T0 . T0 is the room temperature and F is the noise ﬁgure of the ampliﬁer. The ampliﬁer has a bandwidth B around the carrier frequency f 0 . . The antenna produces the desired signal s.
− / gCh .t/.bb/ ST x . in particular.− / D Re AT x (4.− / D Re AT x that is the channel attenuates the signal and introduces a delay equal to −1 . Transmission media transmitted signal.bb/ gCh .184) ] D Re[A Rc e j' Rc e j2³ f 0 t ] (4. hence A Rc / A T x =d. As the propagation delay is given by − D d=c.185) where −1 D d=c denotes the propagation delay. a part of its power is absorbed by the surface while the rest is retransmitted in another direction. Rc Reﬂection and scattering phenomena imply that the oneray model is applicable only to propagation in free space.188) indicates that the received signal exhibits a phase shift of ' Rc D 2³ f 0 −1 with respect to the transmitted signal.3 ns/m.bb/ f 0 was removed because the input already satisﬁes the condition . (4.189) 7 The constraint that GCh .186) gCh . (4.184).− −1 / (4. the total reﬂection factor is ai D Ki Y jD1 ai j (4. We will now consider the propagation of a narrowband signal in the presence of reﬂections.187) Limited to signals sT x of the type (4.a/ . A Rc is the amplitude of the received signal. If the ith ray has undergone K i reﬂections before arriving at the receiver and if ai j is a complex number denoting the reﬂection coefﬁcient of the jth reﬂection of the ith ray. f / D 0 for f < . the amplitude of the received signal decreases linearly with the distance.a/ e h . f / D 0 for f < f 0 . and the power of the received signal is given by P Rc D A2 =2.− −1 / (4. .186) can be rewritten as Ä ½ A Rc j' Rc . then A Rc D A0 =d.150) of h .188) Thus.t/ D Re[A Rc e j2³ f 0 .300 Chapter 4. the delay per unit of distance is equal to 3. and ' Rc D 2³ f 0 −1 D 2³ f 0 d=c is the phase of the received signal.185) has impulse response Ä ½ A Rc . if A0 is the amplitude of the received signal at the distance of 1 meter from the transmitter. If a ray undergoes a reﬂection caused by a surface. that is sT x . and is not adequate to characterize radio channels.− / D 2A Rc e AT x j2³ f 0 −1 Ž. As the power decreases with the square of the distance between transmitter and receiver. because of the propagation delay.t/ D Re[A T x e j2³ f 0 t ] The received signal at a distance d from the transmitter is given by s Rc .a/ h . such as for example the channel between a ﬁxed radio station and a mobile receiver.t −1 / (4. the baseband equivalent of gCh is given by7 . Choosing f 0 as the carrier frequency. the radio channel associated with (4. Using the deﬁnition (1.
186) we get " # Nc A0 X ai .− / D Re h .192) (4.− −i / (4. from (4. If Nc is the number of paths and di is the distance traveled by the ith ray.− / D Nc 2A0 X ai e A T x i D1 di j2³ f 0 −i Ž. . : : : .193) the resulting signal is given by the sum of Ai e j i . of the term A0 .bb/ gCh . undergo an attenuation due to reﬂections that is added to the attenuation due to distance.29.193) in the complex plane.6.t/ D Re[A Rc e j' Rc e j2³ f 0 t ] where now amplitude and phase are given by A Rc e j' Rc D A0 Nc X ai e j'i di i D1 (4.191) We note that the only difference between the passband model and its baseband equivalent is constituted by the additional phase term e j2³ f 0 −i for the ith ray. respectively. The total phase shift asociated with each ray is obtained by summing the phase shifts introduced by the various reﬂections and the phase shift due to the distance traveled. Limited to narrowband signals.190) A T x i D1 di where −i D di =c is the delay of the ith ray.4. Let Ai and i be amplitude and phase. corresponding to rays that are not the direct or line of sight ray.− −i / (4.193) with 'i D 2³ f 0 −i .ai =di /e j'i . The complex envelope of the channel impulse response (4. Radio links 301 Therefore signal amplitudes.190) around f 0 is equal to .a/ gCh . extending the channel model (4. the received power is 0 ψ3 ARc φR c ψ2 ψ 1 Figure 4. as represented in Figure 4. Nc . Representation of (4. i D 1. the received signal can still be written as s Rc .188) to the case of many reﬂections. extending the channel model (4.29. As P0 D A2 =2.
we get h2h2 P0 j1'j2 D PT x GT x G Rc 1 4 2 (4. Moreover. and considering that for the above assumptions the lengths of the two paths are both approximately equal to d.30. and 1d D 2h 1 h 2 =d is the difference between the lengths of the two paths.1 (Power attenuation as a function of distance in mobile radio channels) We consider two antennas. i. For small values of 1' we obtain: j1 e j1' j2 ' j1'j2 D 16³ 2 h2h2 1 2 ½2 d 2 (4. so that the inequalities between the antenna heights and the distance d are no LOS h1 h2 d Figure 4. We will now give two examples of application of the previous results. with height respectively h 1 and h 2 . the earth acts as an ideal reﬂecting surface and does not absorb power. . by substituting (4. one transmitting and the other receiving. Tworay propagation model. Observing (4.196) from which. that is 40 dB/decade instead of 20 dB/decade as in the case of free space. that are placed at a distance d. it is assumed that d × h 1 and d × h 2 (see Figure 4.195) P Rc ' 2 j1 e j1' j2 d where 1' D 2³ f 0 1d=c D 2³ 1d=½ is the phase shift between the two paths. Therefore the law of power attenuation as a function of distance changes in the presence of multipath with respect to the case of propagation in free space.6.197) d2 d We note that the received power decreases as the fourthpower of the distance d.195). Example 4.6.194). the received power is given by P0 (4. We consider the case of two paths: one is the straight path (LOS).180) in (4.302 Chapter 4. and the other is reﬂected by the earth surface with reﬂection coefﬁcient a1 D 1.30).e. but assume that transmitter and receiver are positioned in a room.194) and is independent of the total phase of the ﬁrst ray. P Rc D Example 4.2 (Fading caused by multipath) Consider again the previous example. Transmission media P Rc þ þ2 Nc þX a þ þ i j'i þ D P0 þ e þ þ i D1 di þ (4.
that the rays that reach the receive antenna are due. in the sense that by varying the position of the antennas the received power presents ﬂuctuations of about 20ł30 dB. We now analyze in detail the Doppler shift. It is assumed.200) θ P ∆l Rc Q Figure 4. the phases of the various rays change and the sum in (4.6. In fact. respectively. 4. moreover. The phase variation of the received signal because of the different path length in P and Q is 1' D 2³ v p 1t 2³ 1` D cos Â ½ ½ vp 1 1' D cos Â 2³ 1t ½ (4. 1t is the time required for the receiver to go from P to Q. With reference to Figure 4. to LOS. With these assumptions. The variation in distance between the transmitter and the receiver is 1` D v p 1t cos Â . and reﬂection from the ceiling.4.31. Illustration of the Doppler shift. Radio links 303 longer valid.198) P Rc D P0 þ þ þ i D1 di þ where the reﬂection coefﬁcients are a1 D 1 for the LOS path.3 Doppler shift In the presence of relative motion between transmitter and receiver.6. reﬂection from the ﬂoor. where v p is the speed of the receiver relative to the transmitter. and Â is the angle of incidence of the signal with respect to the direction of motion (Â is assumed to be the same in P and in Q). depending on the position.199) and hence the apparent change in frequency or Doppler shift is fs D T x (4. and the phase difference between the two rays remains always small. one ﬁnds that the power decreases with the distance in an erratic way. and a2 D a3 D 0:7. we consider a transmitter radio Tx and a receiver radio that moves with speed v p from a point P to a point Q. As a result the received power is given by þ þ 3 þX a e j'i þ2 þ þ i (4.193) also varies: in some positions all rays are aligned in phase and the received power is high. whereas in others the rays cancel each other and the received power is low. . In the previous example this phenomenon is not observed because the distance d is much larger than the antenna heights. known as a Doppler shift.31. the frequency of the received signal undergoes a shift with respect to the frequency of the transmitted signal.
instead.201) The (4. in particular.. must be much smaller than the inverse of the Doppler spread of the channel.204) The term indoor is usually referred to areas inside buildings. It is intuitive that the more the characteristics of the radio channel vary with time. is usually referred to areas outside of buildings: these environments can be of various types. f 0 f s /t ] (4. rural. The term outdoor. the received signal would undergo only one Doppler shift.55 km/h (26.202) þ where v p þkm=h is the speed of the mobile in km/h. and f 0 jMHz is the carrier frequency in MHz. etc. if v p D 100 km/h and f 0 D 900 MHz we have f s D 83 Hz. but a person or an object moves modifying the signal propagation. the received signal is s Rc . e. An important consequence of this observation is that the convergence time of algorithms used in receivers. the larger the Doppler spread will be.200) relates the Doppler shift to the speed of the receiver and the angle Â .200) the frequency shift f s depends on the angle Â . Therefore. For example.6. possibly separated by walls of various thickness. urban. the received signal is no longer monochromatic. the received frequency is 26:82 f Rc D f 0 C f s D 1850 ð 106 C D 1850:000166 MHz 0:162 ½D 8 (4. suburban. b) going away from the transmitter. which measures the dispersion in the frequency domain that is experienced by a transmitted sinusoidal signal. material. for example. if it moves away from the transmitter the Doppler shift is negative.304 Chapter 4. This phenomenon manifests itself also if both the transmitter and the receiver are static.3 (Doppler shift) Consider a transmitter that radiates a sinusoidal carrier at the frequency of f 0 D 1850 MHz. For a vehicle traveling at 96. But according to (4. c) perpendicular to the direction of arrival of the transmitted signal.t/ D Re[A Rc e j2³.184) is transmitted. thus enabling the adaptive algorithms to follow the channel variations. for Â D 0 we get f s D 9:259 10 4 v p jkm=h f 0 jMHz (Hz) (4. to perform adaptive equalization. The Doppler spectrum is characterized by the Doppler spread.203) (4. Transmission media This implies that if a narrowband signal given by (4. we want to evaluate the frequency of the received carrier if the vehicle is moving: a) approaching the transmitter.82 m/s). If the signal propagation were taking place through only one ray. We note that if the receiver moves towards the transmitter the Doppler shift is positive. . We now consider a narrowband signal transmitted in an indoor environment8 where the signal received by the antenna is given by the contribution of many rays. Example 4. and we speak of a Doppler spectrum to indicate the spectrum of the received signal around f 0 . The wavelength is 3 ð 108 c D D 0:162 m f0 1850 ð 106 a) The Doppler shift is positive. each with a different length. because of the different paths.g. and height.
In the literature (4. − / D Nc X i D1 gi .207) At a given instant t. i.e. (4. therefore there is no Doppler shift.bb/ GCh . If the channel is timeinvariant. Otherwise.bb/ GCh .206) models the channel as a linear ﬁlter having timevarying impulse response.t/ Q (4.6. or to changes in the surrounding environment.209) 9 If we normalize the coefﬁcients with respect to g1 . in this time interval the impulse response is only a function of − .bb/ gCh .t/. If the duration of the channel impulse response is very small with respect to the duration of the symbol period. or to both factors. the channel is equivalent to a ﬁlter with impulse response illustrated in Figure 4. which implies rapid changes of the received signal power over short distances (of the order of the carrier wavelength) and brief time intervals.− −i .t/ C g2 . the channel model (4.4. if the gain of the single ray varies in time we speak of a ﬂat fading channel.208) is called Rummler model of the radio channel. the transmitted signal is narrowband with respect to the channel. b) time dispersion of the impulse response caused by diverse propagation delays of multipath rays. where the channel variability is due to the motion of transmitter and/or receiver. c) Doppler shift.t.t// Q (4.t/e j2³ f −2 . which introduces a random frequency modulation that is in general different for different rays.t. an adequate model must include several rays: in this case if the gains vary in time we speak of a frequency selective fading channel.4 Propagation of wideband signals For a wideband signal with spectrum centered around the carrier frequency f 0 . (4.t/Ž. we rewrite the channel impulse response as a function of both the time variable t and the delay − for a given t: . For a given receiver location. f / D 1 C b e j2³ f − (4. In a digital transmission system the effect of multipath depends on the relative duration of the symbol period and the channel impulse response. Radio links 305 b) The Doppler shift is negative.6. − / D g1 . then the oneray model is a suitable channel model.206) where gi represents the complexvalued gain of the ith ray that arrives with delay −i . .− −2 . letting −2 D −2 −1 a simple tworay radio channel model has impulse response .209) becomes .t.t/Ž.191) is still valid. the received frequency is 26:82 f Rc D f 0 f s D 1850 ð 106 D 1849:999834 MHz 0:162 c) In this case cos.− / C g2 .bb/ gCh . Q Neglecting the absolute delay −1 . or at least it is timeinvariant within a short time interval.32 and frequency response given by:9 . f / D g1 .t/Ž. The transmitted signal undergoes three phenomena: a) fading of some gains gi due to multipath.Â / D 0. (4.t// (4.208) where b is a complex number.205) 4.
t/ C 2g1 .306 Chapter 4. Therefore.209) the following frequency response is obtained þ2 þ þ þ .211) From (4.206) the received power is P Rc D PT x Nc X i D1 jgi j2 (4. that is they do not interact with each other.t// shown in Figure 4. Transmission media Figure 4. the signal distortion depends on the signal bandwidth in comparison to 1=−2 . in the transmission of narrowband signals the received power is the square of the vector amplitude resulting from the vector sum of all the received rays. where g1 and g2 are assumed to be positive. In this case from (4.32.t/ C g2 .32.bb/ 2 2 Q (4. Physical representation and model of a tworay radio channel.210) þGCh . from (4.t/g2 . as the attenuation depends on frequency. Conversely.211) we note that the received power is given by the sum of the squared amplitude of all the rays.t/ cos. It is evident that the channel has a selective frequency behavior. In any case. Q Going back to the general case. the received power will be lower for a narrowband signal as compared to a wideband signal. rays with different delays are assumed to be independent. For g1 and g2 realvalued.2³ f −2 . . for a given transmitted power. for wideband communications. f /þ D g1 .t.
We deﬁne as power delay proﬁle. the simplest measure is the delay time that it takes for the amplitude of the ray to decrease by x dB below the maximum value. In other words. we use the (average) rms delay spread − r ms obtained by substituting in (4. A parameter that is normally used to deﬁne conveniently the MDS of the channel is the rootmean square (rms) delay spread. that is −r ms D where Nc X q −2 −2 (4. Statistical description of fading channels The most widely used statistical description of the gains fgi g is given by Q g1 D C C g1 Q gi D gi i D1 i D 2. and of the order of some tenths of ns in indoor channels.4.9.213) in place of jgi j2 its expectation. Radio links 307 Channel parameters in the presence of multipath To study the performance of mobile radio systems it is convenient to introduce a measure of the channel dispersion in the time domain known as multipath delay spread (MDS). all the other rays are assumed to have only a random component: therefore the distribution .212) jgi j2 −in n D 1. the EDS is not a very meaningful parameter. as a function of delay −i .5 power delay proﬁles are given for some typical channels.213) The above formulae give the rms delay spread for an instantaneous channel impulse response. whereas the ﬁrst ray contains a direct (deterministic) component in addition to a random component. Nc (4. −r ms . this time is also called excess delay spread (EDS).3 on page 67). Typical values of (average) rms delay spread are of the order of µs in outdoor mobile radio channels. With reference to the timevarying characteristics of the channels. E[jgi j2 ]. The MDS is the measure of the time interval that elapses between the arrival of the ﬁrst and the last ray. In this case − r ms measures the mean time dispersion that a signal undergoes because of multipath. : : : . which corresponds to the secondorder central moment of the channel impulse response. However.214) where C is a realvalued constant and gi is a complexvalued random variable with zero Q mean and Gaussian distribution (see Example 1.6. because channels that exhibit considerably different distributions of the gains gi may have the same value of EDS. also called delay power spectrum or multipath intensity proﬁle. 2 jgi j 2 −n D i D1 Nc X i D1 (4. In Table 4. the expectation of the squared amplitude of the channel impulse response.
a/ D 2.I .a/ N where I0 is the modiﬁed Bessel function of the ﬁrst type and order zero. In (4.1 C K /a 2 ]I0 [2a K .216) The probability density (4. 2³ /.t/ C C] cos 2³ f 0 t g1. In general for a model with more rays we take K D C 2 =Md .x/ D 2³ ³ (4.215) is given in Figure 4.215) 2 pjgi j . In this case the expression of the received signal is given by (4.Q . If C D 0. Q D1 Assuming that the power delay proﬁle is normalized such that Nc X i D1 E[jgi j2 ] D 1 (4. C D 1.218) .e.t/ D [g1. that is Md D iNc E[jgi j2 ]. and the Rayleigh distribution is obtained for all the gains fgi g.a/ D 2a e a 1. where Md is the P statistical power of all reﬂected and/or scattered components. it is K D 0.a/ N (4.217) p we obtain C D K =. i 6D 1.308 Chapter 4. For a oneray channel Q Q model.5 Values of E[jgi j2 ] (in dB) and −i (in ns) for three typical channels. which we rewrite as follows: Q s Rc . For K ! 1. Typical reference values for K are 3 and 10 dB. In p particular. i.192). we have N p pjg1 j . with no reﬂected and/or scattered components and. is equal to the ratio between the power of the direct component and the power of the reﬂected and/or scattered component. known as the Rice factor.1 C K / a exp[ K . To justify the Rice model for jg1 j we consider the transmission of a sinusoidal signal (4.e.t/ sin 2³ f 0 t Q (4. no direct component exists. Z ³ 1 e x cos Þ dÞ I0 . hence.214) the phase of gi is uniformly distributed in [0.33 for various values of K . Transmission media Table 4.184). we ﬁnd the model having only the deterministic component.K C 1/. letting gi D gi = E[jgi j2 ]. i. the parameter K D C 2 =E[jg1 j2 ].1 C K /]1. Standard GSM −i 0 200 500 1600 2300 5000 E[jgi j2 ] 3:0 0 2:0 6:0 8:0 10:0 Indoor ofﬁces −i 0 50 150 325 550 700 E[jgi j2 ] 0:0 1:6 4:7 10:1 17:1 21:7 Indoor business −i 0 50 150 225 400 525 750 E[jgi j2 ] 4:6 0 4:3 6:5 3:0 15:2 21:7 of jgi j will be a Rice distribution for jg1 j and a Rayleigh distribution for jgi j.
t/ Q Q2 (4. The Rice probability density function for various values of K.Q .t. − 1− / D E[g.5 3 Figure 4. A general continuoustime model is now presented.t.5 1 1.−. they can be approximated by independent Gaussian random processes with zero mean. The Rayleigh density function is obtained for K D 0. As the gains g1. in the assumption just formulated. Assuming that the signal propagation occurs through a large number of paths. − /g Ł .t 1t. − 1− /] (4. which in turn are subject to a very large number of random phenomena.t − /.t/ C C]2 C g1. The instantaneous envelope of the received signal is then given by q [g1. 4.2 0 0 0. In particular g.5 a 2 2.Q are due to the scattered component.6 K=5 1.219) which.33.4 0.4. and g1.8 0.5 Continuoustime channel model The channel model previously studied is especially useful for system simulations.I and g1.6.8 1. − / represents the channel output at the instant t in response to an impulse applied at the instant .I . where C represents the contribution of the possible direct component of the propagation Q Q Q signal. We now evaluate the autocorrelation function of the impulse response evaluated at two different instants and two different delays.I and g1.t. as will be discussed later.Q Q are given by the sum of a large number of random components.6 0. is a Rice random variable for each instant t.2 (a) K=2 1 1 p g  K=0 0. the (baseband equivalent) channel impulse response can be represented with good approximation as a timevarying complexvalued Gaussian random process g.4 1.220) .t. rg .6. t 1t. Radio links 309 2 K=10 1. − /.
− /.− − /2 M. . Transmission media According to the model known as the widesense stationary uncorrelated scattering (WSSUS). t 1t. where − r ms is the parameter deﬁned in (4. the autocorrelation is nonzero only for impulse responses that are considered for the same delay time.225): 1. as g is stationary in t. Power delay proﬁle For 1t D 0 we deﬁne the function M.− /Ž. Gaussian.− / C 1 Ž. Moreover.t. the autocorrelation only depends on the difference of the times at which the two impulse responses are evaluated. For a Rayleigh channel model.− / d− 1 where −D Z 1 1 − M. with equal power M. Therefore we have rg .t.− / D 3. As in the case of the discrete channel model previously studied.310 Chapter 4. that is called channel power delay proﬁle and represents the statistical power of the gain g. unilateral M. we deﬁne the (average) rms delay spread as Z 1 .− / D 1 − r ms e −=. unilateral r M.− r ms / 2− r ms / (4.− / d− (4.2− r ms / − ½0 (4.− 2 2 2.− / D E[jg.− / d− 1 2 Z 1 (4. the values of g for rays that arrive with different delays are uncorrelated.225) − r ms D M. and g is stationary in t.223) − ½0 (4.1− / (4. Exponential.t.224) The measure of the set of values − for which M.−. − /j2 ]. Two rays. − / for a given delay − .− / D 1 Ž. − 1− / D rg .221) In other words.222) 2 1 e ³ − r ms 2 − 2 =.− / is above a certain threshold is called (average) excess delay spread of the channel. three typical curves are now given for M.1t.226) The inverse of the (average) rms delay spread is called the coherence bandwidth of the channel. if the delay time is the same.
5/2 C .0:1/.1/.34.230) M( τ ) (dB) 0 10 20 30 0 1 2 5 τ ( µs) Figure 4. f. so that a suitably designed receiver can recover the transmitted information. in the presence of ﬂat fading the received signal may vanish completely. if the coherence bandwidth of the channel is lower than 5 times the modulation rate of the transmission system.5/ C . at frequencies f and f 1 f .0:01/.6.µ s/2 −N2 D 0:01 C 0:1 C 0:1 C 1 Therefore we get −r ms D N p 21:07 .t.t. However.34.0/ D 21:07 .227) Consequently the coherence bandwidth of the channel is equal to Bc D 146 kHz.213) we have −D N and . Radio links 311 For digital transmission over such channels. 1 and determine the coherence bandwidth. and.6.4 (Power delay proﬁle) We compute the average rms delay spread for the multipath delay proﬁle of Figure 4. deﬁned as Bc D 5−r ms .0:01/. .0:1/. otherwise the channel is ﬂat fading. t 1t. First we introduce the correlation function of the channel frequency response taken at instants t and t 1t.4:38/2 D 1:37 µ s (4. or larger then signal distortion is nonnegligible.228) .229) (4. whereas frequency selective fading produces several replicas of the transmitted signal at the receiver.2/ C .4. we observe that if − r ms is of the order of 20% of the symbol period.0:1/.1/2 C .0:1/. Example 4.0/ D 4:38 µ s 0:01 C 0:1 C 0:1 C 1 (4. then we speak of a frequency selective fading channel.1/ C . f 1 f /] (4. Multipath delay proﬁle. rG . Doppler spectrum We now analyze the WSSUS channel model with reference to time variations.t 1t. f /G Ł . f 1 f / D E[G. Equivalently.2/2 C . respectively. ¯ From (4.1/.
1t.1t/ Ð rg .1 f / is the Fourier transform of rg . rg . Now we introduce the Doppler spectrum D. in correspondence of the same delay − .1 f / D rg .½/].t.1t/ M.− / in (4.t.312 Chapter 4.234) represents a normalization factor such that Z C1 D.1t.1t.− / where d.½.236) with d.239) 10 In very general terms.230) the relation G.− / j2³ ½1t e D.1t/ (4. 1 f / D rG .238) With the above assumptions the following equality holds: D.½/ as the Fourier transform of the autocorrelation function of the impulse response.½/ d½ D 1 1 (4.− /. f / D Z C1 1 g.1t. so that Z C1 M.1t.½/ D PG .234) 1 rg .1t/ (4.0. (4.1t/ d. We deﬁne D.½.− / is the power delay proﬁle.237) and M.1t. − /.½/. we could have a different Doppler spectrum for each path.0/ (4.234) implies that rg .− / The term rg . evaluated at two different instants.1 f /e j2³ ½. . Transmission media Substituting in (4.½. − / e j2³ f − d− (4.1t.½/ D d. which represents the power of the Doppler shift for different values of the frequency ½.− / D d. moreover it holds Z C1 rG .− / D d.t. of the channel.1t.0.0/ D 1 (4. We recall that the Doppler shift is caused by the motion of terminals or surrounding objects. The Fourier transform of rG is given by Z 1 PG .235) We note that (4.233) 1 The time variation of the frequency response is measured by PG .231) we ﬁnd that rG depends only on 1t and 1 f . or gain g.1t/ D F 1 [D. that is:10 Z C1 rg . 0/.1 f /− d− 1 (4.0.232) that is rG .− / is a separable function.− / e j2³.− / d− D 1 1 (4.
0/.242) D. known as the Jakes model or classical Doppler spectrum. the corresponding autocorrelation function of the channel impulse response is given by rg . For indoor radio channels.4.− / D Nc X i D1 sinc. Radio links 313 Therefore.241) where J0 is the Bessel function of the ﬁrst type and order zero. we usually say that the channel is fast fading if f d T > 10 2 . Another measure of the support of D. If f d denotes the Doppler spread. The maximum frequency f d of the Doppler spectrum support is called the Doppler spread of the channel and gives a measure of the fading rate of the channel. thanks to the study conducted by a special commission (JTC).½/ can also be obtained as the Fourier transform of rG .240) : 0 otherwise For the channel model (4.206). f / D (4.− / D Nc X i D1 J0 .243) A further model assumes that the Doppler spectrum is described by a second or thirdorder Butterworth ﬁlter with the 3 dB cutoff frequency equal to f d .−i / Ž.½/ can be obtained through the rms Doppler spread or second order central moment of the Doppler spectrum.− −i / (4. to represent the Doppler spectrum is due to Clarke.6.− −i / (4. then 8 1 < 1 p j f j Ä fd ³ f d 1 . D. which in turn can be determined by transmitting a sinusoidal signal (hence 1 f D 0) and estimating the autocorrelation function of the amplitude of the received signal. The inverse of the Doppler spread is called coherence time: it gives a measure of the time interval within which a channel can be assumed to be time invariant or static.−i / Ž. Shadowing The simplest relation between average transmitted power and average received power is P Rc D P0 dÞ (4. f / D 2 f d : 0 elsewhere with a corresponding autocorrelation function given by rg . Let T be the symbol period in a digital transmission system.1t. The model of the Doppler spectrum described above agrees with the experimental results obtained for mobile radio channels.244) . and slow fading if f d T < 10 3 . it was demonstrated that the Doppler spectrum can be modelled as 8 < 1 j f j Ä fd (4. f = f d /2 D. Doppler spectrum models A widely used model.1t.2³ f d 1t/ M.1t.2 f d 1t/ M.
has a shadowing with ¦. whereas for the correct design of a network it is good practice to make use of the largest possible quantity of topographic data. with random amplitude and phase. we would have a model with ¦¾ D 0. shadowing should be considered in the performance evaluation of mobile radio systems. and also the material used for their construction. Final remarks A signal that propagates in a radio channel for mobile communications undergoes a type of fading that depends on the signal as well as on the channel characteristics. For indoor and urban outdoor radio channels the relation depends on the environment. Shadowing takes into account the fact that the average received power may present ﬂuctuations around the value obtained by deterministic models. . in general. that is the coherence time of the channel is smaller than the symbol period. centered at − D 0.¾ /d B . the shadowing can be reduced. this condition leads to signal distortion. Transmission media where Þ is equal to 2 for propagation in free space and to 4 for the simple 2ray model described before. the inverse of the transmitted signal bandwidth is much larger than the delay spread of the channel and g. A propagation model that completely ignores any information on land conﬁguration. for example. − / can be approximated by a delta function. in other words. whereas the delay spread due to multipath leads to dispersion in the time domain and therefore frequency selective fading. In general.¾ /d B D 12 dB. In a fast fading channel. the channel can be assumed as time invariant for a time interval that is proportional to the inverse of the Doppler spread. Improving the accuracy of the propagation model. the Doppler spread causes dispersion in the domain of the variable ½ and therefore time selective fading. The relation between ¦¾ and ¦.314 Chapter 4. these conditions occur when the inverse of the transmitted signal bandwidth is of the same order or smaller than the delay spread of the channel. which increases with increasing Doppler spread. Usually there are no remedies to compensate for such distortion unless the symbol period is decreased. variations of the average received power are lower in outdoor environments than in indoor environments. In the second case instead the channel has a timevarying frequency response within the passband of the transmitted signal and consequently the signal undergoes frequency selective fading. the impulse response changes much more slowly with respect to the symbol period. their dimensions.¾ /d B is ¦¾ D 0:23¦. If P Rc is the average received power obtained by deterministic rules. Hence. by using more details regarding the environmental conﬁguration. The ﬁrst type of fading can be divided into ﬂat fading and frequency selective fading.t. These ﬂuctuations are modelled as a lognormal random variable. In the ﬁrst case the channel has a constant gain. on the other hand. according to the number of buildings. where ¾ is a Gaussian random variable with zero mean and variance ¦¾2 . the impulse response of the channel changes within a symbol period. in practice shadowing provides a measure of the adequacy of the adopted deterministic model. In particular. this choice leads to larger intersymbol interference. In a slow fading channel. that is e¾ . in the presence of shadowing it becomes e¾ P Rc . The received signal consists of several attenuated and delayed versions of the transmitted signal. A channel can be fast fading or slow fading. however. in case we had an enormous amount of topographic data and the means to elaborate them. and therefore is based only on the distance between transmitter and receiver.
.g. where wi . in the case of ﬂat fading we choose Nc D 1.A.4.6. and 1=10 p f d TP Ä 1=5. which imposes the desired power delay proﬁle. Table 4. The interpolator output signal is then multiplied by a constant ¦i D M.− /. Discrete time model of a radio channel. 1.6 Discretetime model for fading channels Our aim is to approximate a transmission channel deﬁned in the continuoustime domain by a channel in the discretetime domain characterized by sampling period TQ . Radio links 315 4. the scheme of Figure 4. we need to obtain a sampled version of M.36.224)).206) must be multiples of TQ and consequently we need to approximate the delays of the power delay proﬁle (see. : : : . and are obtained as realizations of Nc random variables. In general. e.i TQ /. The discretetime model of the radio channel is represented.5). Starting from a continuoustime model of the power delay proﬁle (see. To generate each process gi .g. x(kTQ) TQ TQ TQ g (kTQ) 0 g (kTQ) 1 g (kT ) Q N1 c + y(kTQ) Figure 4. If the channel is time invariant ( f d D 0). as illustrated in Figure 4.7).6.35.36 is used. (4. all coefﬁcients fgi g.kTQ /. are constant. Model to generate the ith coefﬁcient of a timevarying channel. and h int is an interpolator ﬁlter (see Section 1. e. Figure 4. h ds is a narrrowband ﬁlter that produces a signal gi0 with the desired Doppler spectrum. .35. however. by a timevarying linear ﬁlter where the coefﬁcient gi corresponds to the complex gain of the ray with delay i TQ . We immediately notice that the various delays in (4. i D 0.. Nc 1. i D 0. : : : . Nc 1.`T P / is complexvalued Gaussian white noise with zero N mean and unit variance. fgi g are random processes. Ä Usually we choose f d TQ − 1.
242) in two ways: 1) implement a ﬁlter h ds such that jHds .!d TP =2/ where TP is the sampling period.211).246) Generation of a process with a preassigned spectrum The procedure can be generalized for a signal gi0 with a generic Doppler spectrum of the type (4. given by N gi the cascade of h ds and h int . if the channel model N N includes a Doppler shift f si for the ith branch.9.1 (4. Observing (4. We analyze the two methods. so that Nc 1 X i D0 E[jgi . f /j2 D D. Given !d D 2³ f d . a constant Ci must be added to the random component gi .1 C !0 C 2 !0 /2 1 4 (4.249) (4. it is M 0 D 1 if M wi D 1. f /.10 on page 72. : : : . where f d is the Doppler spread.1 C a1 C a2 / 11 Based on the Example 1. We give the description of h ds for two cases 1.z/ D 2 X n an z 1C nD1 where. 1) Using a ﬁlter. we have [18] a1 D 2 !0 / p 2 1 C !0 C 2 !0 2.1 C z 1 /2 ! (4.240) or (4.¦i2 C Ci2 / D 1 (4. 1. i D 0.kTQ /j2 ] D 1 (4. m D 1. then we need to multiply the term Ci C gi by the exponential function exp.a) Secondorder Butterworth ﬁlter.245) For example. in the range from f d to f d .316 Chapter 4. N f . the coefﬁcients fgi g. Transmission media If the channel model includes a deterministic component for the ray with delay −i D i TQ .247) Hds . . 2) generate a set of N f (at least 10) complex sinusoids with frequencies fš f m g. deﬁning !0 D tan. the equivalent interpolator ﬁlter.250) .248) a2 D c0 D 4 1 C !0 p 2 . : : : . to avoid modifying the average transmitted power. Nc 1. has energy equal to the interpolation factor T P =TQ . the above condition is satisﬁed if each signal gi0 has unit statistical power11 and f¦i g satisfy the condition Nc 1 X i D0 . j2³ f si kTQ /. are scaled. Furthermore. the transfer function of the discretetime ﬁlter is c0 .
f /d f .`T P / D a1 gi0 .252) The spacing between the different frequencies is 1 f m .z/ D B..`T P / D Nf X mD1 Ai. 1 fm D Z or.m / e j8i. 11 : 1:0000 e C 0 4:4153 e C 0 6:1051 e C 0 1:3542 e C 0 7:9361 e C 0 5:1221 e C 0 fbn g. Hds 2 ..I C e j . is an FIR shaping ﬁlter with amplitude characteristic of the frequency response given by the square root of the function in (4. Now h ds is implemented as the cascade of two ﬁlters.] Hds ..6 Parameters of an IIR ﬁlter which implements a classical Doppler spectrum. [From Anastasopoulos and Chugg (1997). Table 4. Radio links 317 The ﬁlter output gives gi0 . with cutoff frequency f d . : : : .251) C c0 .254) Table 4. 2) Using sinusoidal signals.m / e j8i.240). n D 0. as 1 fm D Kd N f D1=3 . 21 : 1:3651 e 4 8:1905 e 2:0476 e 3 9:0939 e 1:8067 e 3 1:3550 e 7:1294 e 5 9:5058 e 1:3321 e 5 4:5186 e 1:8074 e 5 3:0124 e 4 4 3 5 5 6 2:0476 e 6:7852 e 5:3726 e 7:1294 e 6:0248 e 3 4 4 5 5 2:7302 e 1:3550 e 6:1818 e 2:5505 e 4:5186 e 3 3 5 5 5 .6.2³ f m `TP C'i.z/.wi .z/ f d TP D 0:1 8:6283 e C 0 3:3622 e C 0 1:8401 e C 0 9:4592 e C 0 7:2390 e C 0 2:8706 e 1 fan g.z/=A. for m > 1 we have f m D f m 1 C 1 f m .m [e j . c 1997 IEEE. n D 0. : : : .` Q 1.` 2/T P / 2/T P // (4.`T P / C 2wi .` 1/T P / a2 gi0 .4.z/. Let gi0 .. The second. The ﬁrst. deﬁning K d D 0 fd fd Nf (4. letting f 1 D 1 f 1 =2. f m / m D 1. N f (4. Hds 1 .Q ] (4.2³ f m `TP C'i.` Q Q 1/T P / C wi .6 reports values of the overall ﬁlter parameters for f d TP D 0:1 [19]. is a Chebychev lowpass ﬁlter. Each 1 f m can be chosen as a constant.253) D1=3 . : : : .b) IIR ﬁlter with classical Doppler spectrum.
today are extensively used also for the transmission of data. We point out that the parameter − r ms provides scarce information on the actual behavior .Q are uniformly distributed in [0. N f . Transmission media Figure 4.37. p The amplitude is given by Ai.318 Chapter 4.I and 8i.t.35. Nine realizations of jgCh . f m /1 f m . This choice for 8i. f / presents some frequencies with large amplitude. − /j for a Rayleigh channel with an exponential power delay proﬁle having − rms D 0:5 T. the choice (4.255) The phases 'i.7.m must be generated as a Gaussian random variable with zero mean and variance D.m .7 4.bb/ of gCh . 2³ / and statistically independent. 8i. 4. for an exponential power delay proﬁle with − r ms D 0:5 T . Transmission of a signal over a telephone . In Figure 4. by the central limit theorem we can claim that gi0 is a Gaussian process. originally conceived for the transmission of voice.I and 8i. : : : .m D D. Ai.Q ensures that the real and imaginary parts of gi0 are statistically independent. fm f /2 D. f / is ﬂat. f m /1 f m . m D 1. f / d f (4.254) corresponds fm fm 1 Nf XZ mD1 . (bb) Suppose f 0 D 0 and f m D f m to minimizing the error 1 C 1 f m .37 are represented nine realizations of the amplitude of the impulse response of a Rayleigh channel obtained by the simulation model of Figure 4. if instead D. The Doppler frequency f d was assumed to be zero.1 Telephone channel Characteristics Telephone channels. which can scatter for a duration equal to 45 times − r ms . If D.
coaxial cables. a telephone channel is characterized by the following disturbances and distortions. Frequency offset It is caused by the use of carriers for frequency up and downconversion. The attenuation and envelope delay distortion are normalized by the values obtained for f D 1004 Hz and f D 1704 Hz.t/ is given by ( X. f C f off / f <0 Usually f off Ä 5 Hz. the signaltoquantization noise ratio 3q has the behavior illustrated in Figure 4.7. It is caused by electromechanical switching devices and is measured by the number of times the noise level exceeds a certain threshold per unit of time.257) are illustrated in Figure 4. and satellite links. f / D 20 log10 jGCh . The plots of the attenuation a. Therefore channel characteristics depend on the particular connection established. For a single quantizer. .2 and is present at a level of 20 ł 30 dB below the desired signal.t/ and output y.39. respectively. f/ D (4.256) and of the group delay or envelope delay (see (1. Thermal noise.258) X . Telephone channel 319 channel is achieved by utilizing several transmission media. optical ﬁbers. f f off / f >0 Y. Linear distortion The frequency response GCh . It is described in Section 4. f / of a telephone channel can be approximated by a passband ﬁlter with band in the range of frequencies from 300 to 3400 Hz.4. Nonlinear distortion It is caused by ampliﬁers and by nonlinear Alaw and ¼law converters (see Chapter 5).38 for two typical channels. f / 2³ d f (4.149)) −. such as symmetrical transmission lines. f / D 1 d arg GCh . The relation between the channel input x. It is introduced by the digital representation of voice signals and is the dominant noise in telephone channels (see Chapter 5). Noise sources Impulse noise. Quantization noise. As a statistical analysis made in 1983 indicated [2]. radio. f /j (4.
Attenuation and envelope delay distortion for two typical telephone channels.38.320 Chapter 4. Transmission media Figure 4. .
As illustrated in Figure 4. then it is practically indistinguishable from the original voice signal. there are two types of echoes: 1. .7. Telephone channel 321 Figure 4.39. Signal to quantization noise ratio as a function of the input signal power for three different inputs.6. If the echo is not very delayed.4. Phase jitter It is a generalization of the frequency offset (see (4.40. talker speech path talker echo listener echo Figure 4.270)). Three of the many signal paths in a simpliﬁed telephone channel with a single twotofour wire conversion at each end. Echo As discussed in Section 3. Talker echo: part of the signal is reﬂected and input to the receiver at the transmit side.5.40. it is caused by the mismatched impedances of the hybrid.
t/] (4. Listener echo: if the echo is reﬂected a second time. Let s. On terrestrial channels the roundtrip delay of echoes is of the order of 10ł60 ms. We note that the effect of echo is similar to multipath fading in radio systems.t/ is the instantaneous phase deviation. . The nonlinearity of a HPA can be described by a memoryless envelope model.t/ ½ 0 is the signal envelope. Power ampliﬁer (HPA) The ﬁnal transmitter stage in a communication system usually consists of a high power ampliﬁer (HPA). whereas on satellite links it may be as large as 600 ms. The HPA is a nonlinear device with saturation in the sense that.41.t/ be the input signal of the HPA. To mitigate the effect of echo there are two strategies: ž use echo suppressors that attenuate the unused connection of a fourwire transmission line.t/ D A. 4. We will now analyze the various blocks of the baseband equivalent model illustrated in Figure 4. expressed as s. in addition to not amplifying the input signal above a certain value.36. and '. Baseband equivalent model of a transmission channel including a nonlinear device. it introduces nonlinear distortion of the signal itself.259) where A.41. it returns to the listener and disturbs the original signal.t/ cos[2³ f 0 t C '. as illustrated in the scheme of Figure 3. ž use echo cancellers that cancel the echo at the source. Figure 4.8 Transmission channel: general model In this section we will describe a transmission channel model that takes into account the nonlinear effects due to the transmitter and the disturbance introduced by the receiver and by the channel. Transmission media 2.322 Chapter 4.
represent respectively the amplitude/amplitude (AM/AM) conversion and the amplitude/phase (AM/PM) conversion of the ampliﬁer. MsT x is the statistical power of the output signal sT x .t/e j'. the HPA are of two types.t/]/ (4. In practice.'.t/]/ It is usually more convenient to refer to baseband equivalent signals: s . Þ8 D 0:26 and þ8 D 0:25.260) (4. however.4.2³ f 0 t C '. As a rule. Transmission channel: general model 323 The envelope and the phase of the output signal. Þ8 and þ8 suitable parameters. and S and ST x are the amplitudes of the input and output signals. The (4.261) (4.264) OBO D 20 log10 p Ms T x where Ms is the statistical power of the input signal s. the point at which the ampliﬁer operates is identiﬁed by the backoff. without memory. TWT.42 for Þ A D 1. Here we assume S D 1 and ST x D G[1] for all the ampliﬁers considered.t/C8[A. depend on the instantaneous. The travelling wave tube (TWT) is a device characterized by a strong AM/PM conversion. First.t/. The solid state power ampliﬁer (SSPA) has a more linear behavior in the region of small signals as compared to the TWT. .t/]e j .bb/ sT x .263) IBO D 20 log10 p Ms Â Ã ST x (dB) (4. sT x .262) The functions G[A] and 8[A]. The conversion functions are G[A] D 8[A] D ÞA A 1 C þ A A2 Þ8 A 2 1 C þ8 A 2 (4.8.266) where Þ A . transformations of the input: sT x .265) and (4. SSPA.t/ D A.t/ D G[A.t/ D G[A.bb/ .t/ and . þ A D 0:25. respectively.t/] cos. i. that lead to saturation of the ampliﬁer.265) (4. called envelope transfer functions.266) are illustrated in Figure 4. For each type we give the AM/AM and AM/PM functions commonly adopted for the analysis. The AM/PM conversion is usually negligible. We adopt here the following deﬁnitions for the input backoff (IBO) and the output backoff (OBO): Â Ã S (dB) (4.e. we need to introduce some normalizations.t/ C 8[A. þ A .
324 Chapter 4. Transmission media 5 0 G[A] (dB) −5 −10 −15 −14 −12 −10 −8 −6 A (dB) −4 −2 0 2 (a) 25 20 15 Φ[A] (deg.43 the function G[A] is plotted for three values of p. Þ8 D 0:26 and þ8 D 0:25. AM/AM and AM/PM characteristics of a TWT for ÞA D 1.267) (4. þA D 0:25.269) 1 A½1 . In Figure 4. the superimposed dashed line is an ideal curve given by ( A 0< A<1 G[A] D (4.1 C A2 p /1=2 p (4.42.) 10 5 0 14 12 10 8 6 A (dB) 4 2 0 2 (b) Figure 4. Therefore the conversion functions are G[A] D A .268) 8[A] D 0 where p is a suitable parameter.
43.8. Transmission channel: general model 325 5 0 p=3 p=2 p=1 G[A] (dB) 5 10 15 14 12 10 8 6 A (dB) 4 2 0 2 Figure 4. AM/AM characteristic of a SSPA. AM/AM experimental characteristic of two ampliﬁers operating at 38 GHz and 40 GHz.44. 5 HPA 38 GHz HPA 40 GHz 0 G[A] (dB) −5 −10 −15 −14 −12 −10 −8 −6 A (dB) −4 −2 0 2 Figure 4. .4.
i. and ' j . and because of the dynamics and transient behavior of the PLL. to demodulate the received signal. can be neglected. .t/ consist of deterministic components and random noise. The phase noise is usually represented in a transmission system model as in Figure 4.t/] cos !0 t C ' j . of the oscillator. The recovered carrier is expressed as Ã Â dt 2 (4. a.t/ is the amplitude noise. depending on whether they use or not a carrier signal.270) v. Consider. Transmission medium The transmission medium is typically modelled as a ﬁlter. supply voltage. 12 Sometimes also called phase jitter. For example. For transmission lines and radio links the models are given respectively in Sections 4.t/ C 2 where d (longterm drift) represents the effect due to ageing of the oscillator. as well as the effect of ageing. Phase noise The demodulators used at the receivers are classiﬁed as “coherent” or “noncoherent”. The recovered carrier may differ from the transmitted carrier because of the phase noise. At the receiver input.4 and 4.t/. The power spectral density of the AWGN noise can be obtained by the analysis of the system devices.41. statistically independent of the desired signal. Often the amplitude noise a.6. all these noise signals are modelled as an effective additive white Gaussian noise (AWGN) signal.44 illustrates the AM/AM characteristics of two waveguide HPA operating at frequencies of 38 GHz and 40 GHz. Typically both phase and frequency are recovered from the received signal by a phase locked loop (PLL) system.326 Chapter 4. Figure 4. the noise introduced by a receive antenna or the thermal noise and shot noise generated by the preampliﬁer stage of a receiver. Transmission media It is interesting to compare the above analytical models with the behavior of a practical HPA. or by experimental measurements. The phase noise ' j . for example.t/ denotes the phase noise. Additive noise Several noise sources that cause a degradation of the received signal may be present in a transmission system. which ideally should have the same phase and frequency as the carrier at the transmitter.12 due to shortterm stability. frequency drift.t/ D Vo [1 C a. which employs a local oscillator. and the output impedance of the oscillator are included among deterministic components. temperature change.e.
f 2 D 2 MHz. c.4.8.t/ are in the range from 10 2 to 10 4 . The plot of (4. Depending on the values of a. with the exception of the frequency drift. −60 −70 −80 Pφ(f) (dBc/Hz) −90 −100 −110 (~ −20 dB/decade) − −120 −130 0 5 10 15 f (MHz) Figure 4.271) for f ` Ä f Ä f h . a D 65 dBc/Hz.272) is shown in Figure 4.45 for f 1 D 0:1 MHz. Transmission channel: general model 327 Ignoring the deterministic effects.272) where the parameters a and c are typically of the order of 65 dBc/Hz and 125 dBc/Hz. a PSD model of the of ' j . often used. and b is a scaling factor that depends on f 1 and f 2 and assures continuity of the PSD. f / D k 4 0 C k 3 0 C f4 f3  {z }  {z } random frequency walk ﬂicker frequency noise random phase walk or white frequency noise f2 k 2 0 f2  {z } f2 C k 1 0 C f  {z } ﬂicker phase noise white phase noise k0 f 2  {z0 } (4. A simpliﬁed model. that is it represents the statistical power of the phase noise. and c D 125 dBc/Hz. is given by 8 >a < P' j . . expressed in dB.45. dBc means dB carrier. b. respectively. f 1 and f 2 .t/ comprises ﬁve terms: f2 f2 P' j . with respect to the statistical power of the desired signal received in the passband. Simpliﬁed model of the phasenoise power spectral density. typical values of the statistical power of ' j . f / D c C >b 1 : f2 j f j Ä f1 f1 Ä j f j < f2 (4.
D. pp. “Fiber optic communications systems”. 2nd ed. H. J.328 Chapter 4. Jeunhomme. 1997. of the Technical Staff.. 1997. ed. NJ: PrenticeHall. 1992. [17] J. Miyashita. Electromagnetic waves. C. parameters. vol. 106–108. Rappaport. Whinnery. Pahlavan and A. Miya. 1996. Principles of mobile communication. J.. Levesque. Feher. 1980. 1990. 1175.. [15] W. C. Rao. Winston. Lee.). 1990. pp. Someda. 5th ed. Van Duzer. p. Wireless communications: principles and practice. NC: Bell Telephone Laboratories. Wireless information networks. IEEE. Gibson. Microwave mobile communications. NJ: PrenticeHall. “Ultimate lowloss singlemode ﬁbre at 1. Olcer. Upper Saddle River. Palais. Messerschmitt and E. Ungerboeck. Hoss. Digital communications by satellite. Digital communication. 1998. [2] M. 1995. in The Communications Handbook (J. [6] L. [9] T. “100BASET2: a new standard for 100 Mb/s ethernet transmission over voicegrade cables”. vol. 731–739. and T. NJ: PrenticeHall. S. 1994. 2nd ed. New York: IEEE Press. MA: Kluwer Academic Publishers. 1965. Fields and Waves in Communication Electronics. [7] J. L. Singlemode ﬁber optics. 1995. A. Englewood Cliffs. ¨ ¸ [4] G. and S. Englewood Cliffs. 1977. NJ: PrenticeHall. and transmission properties of optical ﬁbers”. New York: John Wiley & Sons. IEEE Communications Magazine. [8] T. Stuber. pp. Fiber optic communications. Transmission systems for communications. Jakes. Ramo. [10] J. 15. ch. MA: Kluwer Academic Publishers. Cherubini. Boston. IEE Electronics Letters. Palais. and T. Fiber optic communications. G. [13] T. S. Boca Raton: CRC Press. Terunuma. 1982. J. 1993. [3] S. Creigh. Hosaka. Oct. [16] G. G. [5] R. 1979. [14] K. 35. Norwell. J. Nov. T. Y. [11] D. 115–122. New York: John Wiley & Sons. C. 3rd ed. 1996.55 µm”. . Transmission media Bibliography [1] C. 54. Proc. Spilker. NJ: PrenticeHall. [12] K. R.. B. New York: Marcel Dekker. K. Li. London: Chapman & Hall. Englewood Cliffs. Wireless digital communications. Feb. Englewood Cliffs. “Structures. G.
4. Bibliography
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[18] A. V. Oppenheim and R. W. Schafer, Discretetime signal processing. Englewood Cliffs, NJ: PrenticeHall, 1989. [19] A. Anastasopoulos and K. Chugg, “An efﬁcient method for simulation of frequency selective isotropic Rayleigh fading”, in Proc. 1997 IEEE Vehicular Technology Conference, pp. 2084–2088, May 1997.
Chapter 5
Digital representation of waveforms
Figure 5.1a illustrates the conventional transmission of an analog signal, for example, speech or video, over an analog channel; in this scheme the transmitter usually consists of an ampliﬁer and possibly a modulator, the analog transmission channel is of the type discussed in Chapter 4, and the receiver consists of an ampliﬁer and possibly a demodulator. Alternatively, the transmission may take place by ﬁrst encoding1 the information contained in the analog signal into a sequence of bits using for example an analogtodigital converter (ADC), as illustrated in Figure 5.1b. If Tb is the time interval between two consecutive bits of the sequence, the bit rate of the ADC is Rb D 1=Tb (bit/s). The binary message is converted by a digital modulator into a waveform that is suitable for transmission over an analog channel. At the receiver, the reverse process occurs: in this case a digital demodulator restores the message, whereas the conversion of the sequence of bits to an analog signal is performed by a digitaltoanalog converter (DAC). The system that has as an input the sequence of bits produced by the ADC, and as an output the sequence of bits produced by the digital demodulator is called a binary channel (see Chapter 7). In this chapter the principles and methods for the conversion of analog signals into binary messages and viceversa will be discussed; as a practical example we will use speech, but the principles may be extended to any analog signal. To compute system performance, a fundamental parameter is the signaltonoise ratio. Let s.t/ be the original signal, s .t/ the Q Q reconstructed signal, and eq .t/ D s .t/ s.t/; then the signaltonoise ratio is deﬁned as 3q D E[s 2 .t/] 2 E[eq .t/] (5.1)
5.1
Analog and digital access
Analog access over a telephone channel in the public switched telephone network (PSTN) is illustrated in Figure 5.2. With reference to the ﬁgure, the word “modem” is the contraction of mod ulatordemodulator. Its function is to convert a binary message, or data signal, into
1
We bring to the attention of the reader that the terms “encoder” and “decoder” are commonly used to indicate various devices in a communication system. In this chapter we will deal with encoders and decoders for the digital representation of analog waveforms.
332
Chapter 5. Digital representation of waveforms
Figure 5.1. Analog vs. digital transmission.
an analog passband signal that can be transmitted over the telephone channel. In Figure 5.2, the source generates a speech signal or a data ﬁle; in the latter case, a modem is required to transmit the signal. The analog signal s.t/ that has a band of approximately 300–3400 Hz is sent over a local loop to the central ofﬁce (see Chapter 4): here it is usually converted into a binary digital message via PCM at 64 kbit/s; in turn this message is modulated before being transmitted over an analog channel. After having crossed several central ofﬁces where switching (routing) of the signal takes place, the PCM encoded message arrives at the destination central ofﬁce: here it is converted into an analog signal and sent over a local loop to the end user. It is here that the signal must be identiﬁed as a speech signal or a digitally modulated signal; in the latter case a modem will demodulate it to reproduce the data message. Figure 5.3 illustrates the concept of direct digital access at the user’s premises. An analog signal is converted into a digital message via an ADC. The user digital message is then sent over the analog channel by a modulator. At the receiver the inverse process is established, where the digital message obtained at the output of the demodulator may be used to restore an analog signal via a DAC. In comparing the two systems, we note the waste of capacity of the system in Figure 5.2. For example, for a 9600 bit/s modem, the modulated PCM encoded signal requires a standard capacity of Rb D 64 kbit/s. By directly accessing the PCM link at the user’s home, we could transmit 64000=9600 ' 6 data signals at 9600 bit/s.
5.1.1
Digital representation of speech
Some waveforms
Some examples of speech waveforms for an interval of 0.25 s are given in Figure 5.4. From these plots, we can obtain a speech model as a succession of voiced speech spurts (see Figure 5.5a), or unvoiced speech spurts (see Figure 5.5b). In the ﬁrst case, the signal
5.1. Analog and digital access
333
Figure 5.2. User line with analog access.
Figure 5.3. User line with digital access.
334
Chapter 5. Digital representation of waveforms
Figure 5.4. Speech waveforms.
Figure 5.5. Voiced and unvoiced speech spurts.
5.1. Analog and digital access
335
is strongly correlated and almost periodic, with a period that is called pitch, and exhibits large amplitudes; conversely in an unvoiced speech spurt the signal is weakly correlated and has small amplitudes. We note moreover that the average level of speech changes in time: indeed speech is a nonstationary signal. In Figure 5.6 it is interesting to observe the instantaneous spectrum of some voiced and unvoiced sounds; we also note that the latter may have a bandwidth larger than 10 kHz. Concerning the amplitude distribution of speech signals, we observe that over short time intervals, of the order of a few tenths of milliseconds (or of a few hundreds of samples at a sampling frequency of 8 kHz), the amplitude statistic is Gaussian with good approximation; over long time intervals, because of the numerous pauses in speech, it tends to exhibit a gamma or Laplacian distribution. We give here the probability density functions of the amplitude that are usually adopted. Let ¦s be the standard deviation of the signal s.t/; then we have gamma: Laplacian: Gaussian: ps .a/ D 3 8³ ¦s jaj 1 p !1
2 p
e
2jaj ¦s 1 a 2 ¦s Á2
3jaj 2¦s
p
ps .a/ D p e 2¦s ps .a/ D p e 2³ ¦s 1
(5.2)
As mentioned above, analog modulated signals generated by modems, often called voiceband data signals, are also transmitted over telephone channels. Figure 5.7 illustrates a
Figure 5.6. Spectrum of voiced and unvoiced sounds for a sampling frequency of 20 kHz.
336
Chapter 5. Digital representation of waveforms
1
s(t)
0
−1
0
0.06
t (s)
Figure 5.7. Signal generated by a modem employing FSK modulation for the transmission of 1200 bit/s.
Figure 5.8. Signal generated by modems employing PSK modulation for the transmission of: (a) 2400 bit/s; (b) 4800 bit/s.
signal produced by the 202S modem, which employs FSK modulation for the transmission of 1200 bit/s, whereas Figure 5.8a and Figure 5.8b illustrate signals generated by the 201C and 208B modems, which employ PSK modulation for the transmission of 2400 and 4800 bit/s, respectively. For the deﬁnition of FSK and PSK modulation we refer the reader to Chapter 6. In general, we note that the average level of signals generated by modems is stationary; moreover, if the bit rate is low, signals are strongly correlated.
5.1. Analog and digital access
337
Speech coding
Speech coding addresses persontoperson communications and is strictly related to the transmission, for example, over the public network, and storage of speech signals. The aim is to represent, using an encoder, speech as a digital signal that requires the lowest possible bit rate to recreate, by an appropriate decoder, the speech signal at the receiver [1]. Depicted in Figure 5.9 is a basic scheme, denoted as ADC, that provides the analogtodigital conversion (encoding) of the signal, consisting of: 1. an antialiasing ﬁlter followed by a sampler at sampling frequency 1=Tc ; 2. a quantizer; 3. an inverse bit mapper (IBMAP) followed by a paralleltoserial (P/S) converter. As indicated by the sampling theorem, the choice of the sampling frequency Fc D 1=Tc is related to the bandwidth of the signal s.t/ (see (1.142)). In practice, there is a tradeoff between the complexity of the antialiasing ﬁlter and the choice of the sampling frequency, which must be greater than twice the signal bandwidth. For audio signals, Fc depends on the signal quality that we wish to maintain and therefore it depends on the application, see Table 5.1 [2].
Figure 5.9. Basic scheme for the digital transmission of an analog signal.
Table 5.1 Sampling frequency of the audio signal in three applications.
Application telephone 300 broadcasting 50 audio, compact disc digital audio tape
Passband (Hz)
Fc (Hz)
3400 (narrow band speech) 8000 7000 (wide band speech) 16000 10 ł 20000 44100 10 ł 20000 48000
338
Chapter 5. Digital representation of waveforms
The choice of the quantizer parameters is somehow more complicated and will be dealt with in detail in the following sections. We will consider now the quantizer as an instantaneous nonlinear transformation that maps the real values of s in a ﬁnite number of values of sq . To illustrate the principle of an ADC, let us assume that sq assumes values that are taken from a set of 8 elements:2 Q[s.kTc /] D sq .kTc / 2 fQ
4 ; Q 3 ; Q 2 ; Q 1 ; Q1 ; Q2 ; Q3 ; Q4 g
(5.3)
Therefore sq .kTc / may assume only a ﬁnite number of values, which can be represented as binary values, for example, using the inverse bit mapper of Table 5.2. It is convenient to consider the sequence of bits that gives the binary representation of fsq g instead of the sequence of values itself. In our example, with a representation using three bits per sample, the bit rate of the system is equal to Rb D 3Fc (bit/s) (5.4)
The inverse process (decoding) takes place at the receiver: the bitmapper (BMAP) restores the quantized levels, and an interpolator ﬁlter yields an estimate of the analog signal.
The interpolator ﬁlter as a holder
Always referring to the sampling theorem, an ideal interpolator ﬁlter3 has the following frequency response Â Ã f (5.5) G I . f / D rect Fc
Table 5.2 Encoder inverse bitmapper.
Values Integer Binary representation sq .kTc / representation c.k/ D .c2 ; c1 ; c0 / Q 4 Q 3 Q 2 Q 1 Q1 Q2 Q3 Q4 0 1 2 3 4 5 6 7 000 001 010 011 100 101 110 111
2
The notation adopted in (5.3) to deﬁne the set reﬂects the fact that in most cases the set of values assumed by sq is symmetrical around the origin. 3 From the Observation 1.7 on page 71, if s .kT / is WSS, then the interpolated random process s .t/ is WSS q c q 2 2 and E[sq .t/] D E[sq .kTc /], whenever the gain of the interpolator ﬁlter is equal to one. As a result the signaltonoise ratio in (5.1) becomes independent of t and can be computed using the samples of the processes, 2 3q D E[s 2 .kTc /]=E[eq .kTc /].
5.1. Analog and digital access
339
g nTc
I
t T =1/Fc c
Figure 5.10. DAC interpolator as a holder.
Typically, however, the DAC employs a simple holder that holds the input values as illustrated in Figure 5.10. In this case Ã Â t Tc =2 D wTc .t/ (5.6) g I .t/ D rect Tc and G I . f / D Tc sinc Â f Fc Ã e
2³ f Tc =2
(5.7)
Unless the sampling frequency has been chosen sufﬁciently higher than twice the bandwidth of s.t/, we see that the ﬁlter (5.7), besides not attenuating enough the images of sq .kTc /, introduces distortion in the passband of the desired signal.4 A solution to this Q problem consists in introducing, before interpolation, a digital equalizer ﬁlter with a frequency response equal to 1= sinc. f Tc / in the passband of s.t/. Figure 5.11 illustrates the solution. A simple digital equalizer ﬁlter is given by G comp .z/ D 9 1 C z 16 8
1
1 z 16
2
(5.8)
whose frequency response is given in Figure 5.12.
g ~ ( kT ) sq c T c
I
g comp T c
wT
~ (t) sq
c
Figure 5.11. Holder ﬁlter preceded by a digital equalizer.
4
In many applications, to simplify the analog interpolator ﬁlter, the signal before interpolation is oversampled: for example, by digital interpolation of the signal sq .kTc / by at least a factor of 4. Q
340
Chapter 5. Digital representation of waveforms
Figure 5.12. Frequency responses of a threecoefﬁcient equalizer ﬁlter gcomp and of the overall ﬁlter gI D gcomp Ł wTc .
An alternative solution is represented by an IIR ﬁlter with: G comp .z/ D 9=8 1 C 1=8z
1
(5.9)
whose frequency response is given in Figure 5.13. In the following sections, by the term DAC we mean a digitaltoanalog converter with the aforementioned variations.
Sizing of the binary channel parameters
As will be further discussed in Section 6.2, in Figure 5.9 the binary channel is characterized by the encoder bit rate Rb . If B is the bandwidth of s.t/, the sample frequency Fc is such that Fc D 1 ½ 2B Tc (5.10)
If L D 2b is the number of levels of the quantizer, then the encoder bit rate is equal to Rb D bFc bit/s. Another important parameter of the binary channel is the bit error probability O Pbit D P[b` 6D b` ] (5.11)
If an error occurs, the reconstructed binary representation c.k/ is different from c.k/: Q consequently the reconstructed level is sq .kTc / 6D sq .kTc /. In the case of a speech signal, Q
5.1. Analog and digital access
341
Figure 5.13. Frequency responses of an IIR equalizer ﬁlter gcomp and of the overall ﬁlter gI D gcomp Ł wTc .
such an event is perceived by the ear as a fastidious impulse disturbance. For speech signals to have an acceptable quality at the receiver it must be Pbit Ä 10 3 .
5.1.2
Coding techniques and applications
At the output of an ADC, the PCM encoded samples, after suitable transformations, can be further quantized in order to reduce the bit rate. From [3], we list in Figure 5.14 various coding techniques, which are divided into three groups, that essentially exploit two elements: ž redundancy of speech, ž sensibility of the ear as a function of the frequency. Waveform coding. Waveform encoders attempt to reproduce the waveform as closely as possible. This type of coding is applicable to any type of signal; two examples are the PCM and ADPCM schemes. Coding by modeling. In this case coding is not related to signal samples, but to the parameters of the source that generates them. Assuming the voiced/unvoiced speech model, an example of a classical encoder (vocoder) is given in Figure 5.15, where a periodic excitation, or white noise segment, ﬁltered by a suitable ﬁlter, yields a synthesized speech segment. A more sophisticated model uses a more articulated multipulse excitation. Frequencydomain coding. In this case coding occurs after signal transformation to a domain different from time, usually frequency: examples are subband coding and transform coding.
342
Chapter 5. Digital representation of waveforms




Figure 5.14. Characteristics exploited by the different coding techniques.
Figure 5.15. Vocoder and multipulse models for speech synthesis.
5.1. Analog and digital access
343
Table 5.3 Voice coding techniques.
Bit rate (kbit/s) 1.2 2.4 4.8 8.0 9.6 16 32 64
Algorithm
Year
Codebook excited LP Multipulse excited LP Vector quantization Time domain harmonic scaling Adaptive transform coding Subband coding Residual excited LP Adaptive predictive coding Formant vocoder Cepstral vocoder Channel vocoder Phase vocoder Linear prediction vocoder Adaptive differential PCM Differential PCM Adaptive delta modulation Delta modulation Pulse code modulation
CELP MELP VQ TDHS ATC SBC RELP APC FORV CEPV CHAV PHAV LPCV ADPCM DPCM ADM DM PCM
1984 1982 1980 1979 1977 1976 1975 1968 1971 1969 1967 1966 1966
Various coding techniques are listed in Table 5.3. Table 5.4 illustrates the characteristics of a few systems, putting into evidence that for more sophisticated encoders the implementation complexity expressed in millions of instructions per second (MIPS), as well as the delay introduced by the encoder (latency), can be considerable. The various coding techniques are different in quality and cost of implementation. With respect to the perceived quality, on a scale from poor to excellent, three categories of encoders perform as illustrated in Figure 5.16: obviously a higher implementation complexity is expected for encoders with low bit rate and good quality. We go from a bit rate in the range from 4.4 to 9.6 kbit/s for cellular radio systems, to a bit rate in the range from 16 to 64 kbit/s for transmission over the public network. Generally a coding technique is strictly related to the application and depends on various factors: ž signal type (for example speech, music, voiceband data, signalling, etc.); ž maximum tolerable latency; ž implementation complexity. In particular, speech encoder applications for bit rate in the range 4–16 kbit/s are: ž long distance and satellite transmission; ž digital mobile radio (cellular radio);
344
Chapter 5. Digital representation of waveforms
Table 5.4 Parameters of a few speech coders.
Coder
Bit rate Computational Latency (kbit/s) complexity (MIPS) (ms) 64 32 16 8 4 2 0.0 0.1 1 10 100 1 0 0 25 35 35 35
PCM ADPCM ASBC MELP CELP LPC
Figure 5.16. Audio quality vs. bit rate for three categories of encoders.
ž modem transmission over the telephone channel (voice mail); ž speech storage for telephone services and speech encryption; ž packet networks with integrated speech and data.
5.2
5.2.1
Instantaneous quantization
Parameters of a quantizer
We consider a sample of a discretetime random process s.kTc /, obtained by sampling the continuoustime process s.t/ with rate Fc . To simplify the notation we choose Tc D 1, unless otherwise stated. With reference to the scheme of Figure 5.17, for a quantizer with L output values we have: ž input signal s.k/ 2 <; ž quantized signal sq .k/ 2 Aq D fQ L=2 ; : : : ; Q the alphabet Aq are called output levels;
1 ; Q1 ; : : : ; Q L=2 g;
the L values of
5.2. Instantaneous quantization
345
Figure 5.17. Quantization and mapping scheme: (a) encoder, (b) decoder.
ž code word c.k/ 2 f0; 1; : : : ; L 1g, which represents the value of sq .k/. The system with input s.k/ and output c.k/ constitutes a PCM encoder. The quantizer can be described by the function Q : < ! Aq (5.12)
For a given partition of the real axis in the intervals fRi g, i D L=2; : : : ; 1; 1; : : : ; L=2 S L=2 T such that < D i D L=2; i 6D0 Ri , Ri R j D ; for i 6D j, (5.12) implies the following rule Q[s.k/] D sq .k/ D Qi if s.k/ 2 Ri (5.13)
A common choice for the decision intervals Ri is given by: ( Ri D .−i ; −i C1 ] for i D L=2; : : : ; 1 Ri D .−i
1 ; −i ]
for i D 1; : : : ; L=2
(5.14)
where − L=2 D 1 and − L=2 D 1. We note that the decision thresholds f−i g are L 1, being − L=2 and − L=2 assigned. The mapping rule (5.12) is called the quantizer characteristic and is illustrated in Figure 5.18 for L D 8 and −0 D 0. The L values of sq .k/ can be represented by integers c.k/ 2 f0; 1; : : : ; L 1g or by a binary representation with dlog2 Le bits. For the quantizer characteristic of Figure 5.18, a binary representation is adopted that goes from 000 (the minimum level), to 111 (the maximum level); in this example the bit rate of the system is equal to Fb D 3Fc bit/s. Let eq .k/ D sq .k/ s.k/ (5.15) be the quantization error. From the relation sq .k/ D s.k/ C eq .k/ we have that the quantized signal is affected by a certain error eq .k/. We can formulate the problem as that of representing s.k/ with the minimum number of bits b, to minimize the system bit rate, and at the same time constraining the quantization error, so that a certain level of quality of the quantized signal is maintained. Observation 5.1 In this chapter the notation c.k/ is used to indicate both an integer number and its vectorial binary representation (see (5.18)). Furthermore, in the context of vector quantization the elements of the set Aq are called code words.
346
Chapter 5. Digital representation of waveforms
Figure 5.18. Threebit quantizer characteristic.
5.2.2
Uniform quantizers
A quantizer with L D 2b equally spaced output levels and decision thresholds is called uniform. For: ( i D L=2 C 1; : : : ; 1 −i C1 −i D 1 (5.16) i D 1; 2; : : : ; L=2 1 −i −i 1 D 1 8 > Qi C1 Qi D 1 < Q1 Q 1 D 1 > : Qi Qi 1 D 1 iD L=2; : : : ; 2 (5.17) i D 2; : : : ; L=2
where 1 is the quantization step size. Two types of characteristics are distinguished, midtread and midriser, depending on whether the zero output level belongs or not to Aq . Midriser characteristic. The quantizer characteristic is given in Figure 5.19 for L D 8: in this case the smallest value, in magnitude, assumed by sq .k/ is 1=2, even for a very small input value s. Let the binary representation of c.k/ be deﬁned according to the following rule: the most signiﬁcant bit of the binary representation of c.k/ denotes the sign .š1/ of the input value, whereas the remaining bits denote the amplitude. Therefore adopting the binary vector representation c.k/ D [cb
1 .k/; : : : ; c0 .k/]
c j .k/ 2 f0; 1g
(5.18)
5.2. Instantaneous quantization
347
7∆ 2 5∆ 2 3∆ 2 ∆ 2  4∆  3∆  2∆  ∆ 100
sq=Q[s]
010
011
001
000
 ∆ 2 3∆ 2 5∆ 2 7∆ 2
∆
2∆
3∆
4∆
s
101

110

111

Figure 5.19. Uniform quantizer with midriser characteristic (b D 3).
the relation between sq .k/ and c.k/ is given by sq .k/ D 1.1 2cb
1 .k// b 2 X jD0
c j .k/ 2 j C
1 .1 2
2cb
1 .k//
(5.19)
Midtread characteristic. The quantizer characteristic is shown in Figure 5.20. Zero is a value assumed by sq . Let the binary representation of c.k/ be the two’s complement representation of the level number. Then we have sq .k/ D 1 c.k/. Note that the characteristic is asymmetric around zero, hence we may use L 1 levels (giving up the minimum output level), or choose an implementation that can be slightly more complicated than in the case of a symmetric characteristic (see page 357).
Quantization error
We will refer to symmetrical quantizers, with midriser characteristic. An example with L D 23 D 8 levels is given in Figure 5.21: in this case the decision thresholds are −i D i1, i D L=2 C 1; : : : ; 1; 0; 1; : : : ; L=2 1, with, as usual, − L=2 D 1 and − L=2 D 1. The output values are given by Ã 8Â > iC1 1 > i D L=2; : : : ; 1 < 2 Ã (5.20) Qi D Â > > i 1 1 : i D 1; : : : ; L=2 2 Correspondingly the decision intervals are given by (5.14).
348
Chapter 5. Digital representation of waveforms
sq=Q[s]
3∆ 010 011
2∆ 001 ∆ 000 7∆ 2 5∆ 2  3∆ 2 111  ∆ 2  ∆ ∆ 2 3∆ 2
5∆ 2
7∆ 2
s
110
 2∆
101
 3∆
100
 4∆
Figure 5.20. Uniform quantizer with midtread characteristic (b D 3).
We note that if sq .k/ D Qi , then the b 1 least signiﬁcant bits of c.k/ are given by the binary representation of .jij 1/, and c.k/ assumes amplitude values that go from 0 to L=2 1 D 2b 1 1. If for each value of s we compute the corresponding error eq D Q.s/ s, we obtain the quantization error characteristic of Figure 5.21. We deﬁne the quantizer saturation value as −sat D −.L=2/
1
C1
(5.21)
that is shifted by 1 with respect to the last ﬁnite threshold value. Then we have jeq j Ä and ( eq D Q L=2 s for s > −sat Q L=2 s for s < −sat (5.23) 1 2 for jsj < −sat (5.22)
Consequently, eq may assume large values if jsj > −sat . This observation suggests that the real axis be divided into two parts: 1. the region s 2 . 1; −sat / [ .−sat ; C1/, where eq is called saturation or overload error .esat /;
5.2. Instantaneous quantization
349
Figure 5.21. Uniform quantizer (b D 3).
350
Chapter 5. Digital representation of waveforms
2. the region s 2 [ −sat ; −sat ], where eq is called granular error .egr /; the interval [ −sat ; −sat ] is also called quantizer range. It is often useful to compactly represent the quantizer characteristic in a single axis, as illustrated in Figure 5.21c, where the values of the decision thresholds are indicated by dashed lines, and the quantizer output values by dots.
Relation between
, b and τsat
The quantization step size 1 is chosen so that 2−sat D L1 Therefore, for L D 2b , 1D 2−sat 2b (5.25) (5.24)
If js.k/j < −sat , observing (5.22) this choice guarantees that eq is granular with amplitude in the range 1 1 Ä eq .k/ Ä 2 2 (5.26)
If js.k/j > −sat the saturation error can assume large values: therefore −sat must be chosen so that the probability of the event js.k/j > −sat is small. For a ﬁxed number of bits b, and consequently for a ﬁxed number of levels L, it is important to verify that, increasing −sat , 1 also increases and hence also the granular error; on the other hand, choosing a small 1 leads to a considerable saturation error. As a result, for each value of b there will be an optimum choice of −sat and hence of 1. In any case, to decrease both errors we must increase b with consequent increase of the encoder bit rate.
Statistical description of the quantization noise
In Figure 5.22 we give an equivalent model of a quantizer where the quantization error is modeled as additive noise. Assuming (5.26) holds, that is for granular eq , we make the following assumptions. 1. The quantization error is white, ( E[eq .k/eq .k n/] D Meq 0 nD0 n 6D 0 (5.27)
2. It is uncorrelated with the input signal E[s.k/eq .k n/] D 0 8n (5.28)
3. It has a uniform distribution (see Figure 5.23): peq .a/ D 1 1 1 1 ÄaÄ 2 2 (5.29)
5.2. Instantaneous quantization
351
Figure 5.22. Equivalent model of a quantizer.
p (a) e
q
1 __
∆
∆ − __ 2
∆ __ 2
a
Figure 5.23. Probability density function of eq .
We note that if s.k/ is a constant signal the above assumptions are not true; they hold in practice if fs.k/g is described by a function that signiﬁcantly deviates from a constant and 1 is adequately small, that is b is large. Figure 5.24 illustrates the quantization error for a 16level quantized signal. The signal eq .t/ is quite different from s.t/ and the above assumptions are plausible. If the probability density function of the signal to quantize is known, letting g denote the function that relates s and eq , that is eq D g.s/, also called quantization error characteristic, the probability density function of the noise is obtained as an application of the theory of functions of a random variable, that yields X ps .b/ 1 1 <a< (5.30) peq .a/ D jg 0 .b/j 2 2 1
b2g .a/
where g is the inverse of the error function, or equivalently the set of values of s corresponding to a given value of eq . We note that in this case the slope of the function g is always equal to one, hence g 0 .b/ D 1, and from (5.15) for 1=2 < a < 1=2 we get ¦ ² L L 1 g .a/ D Qi a; i D ; : : : ; 1; 1; : : : ; (5.31) 2 2 Finally, peq .a/ D
iD 2 X L 2 ; i 6D0 L
1 .Ð/
ps .Qi
a/
1 1 <a< 2 2
(5.32)
It can be shown that, if 1 is small enough, the sum in (5.32) gives origin to a uniform function peq , independently of the form of ps .
352
Chapter 5. Digital representation of waveforms
Figure 5.24. Quantization error, L D 16 levels.
Statistical power of the quantization error
With reference to the model of Figure 5.22, a measure of the quality of a quantizer is the signaltoquantization error ratio: 3q D E[s 2 .k/] 2 E[eq .k/] (5.33)
Choosing −sat so that eq is granular, from (5.29) we get Meq ' Megr ' 12 12 (5.34)
For an exact computation that includes also the saturation error we need to know the probability density function of s. The statistical power of eq is given by Z C1 2 Meq DE[eq .k/] D [Q.a/ a]2 ps .a/ da Z D
1 −sat − Zsat1 −sat
[Q.a/ [Q.a/
a]2 ps .a/ da C a]2 ps .a/ da :
Z
−sat
[Q.a/
1
a]2 ps .a/ da
(5.35)
C
In (5.35) the ﬁrst term is the statistical power of the granular error, Megr , and the other two terms express the statistical power of the saturation error, Mesat . Let us assume that ps .a/
5.2. Instantaneous quantization
353
is even and the characteristic is symmetrical, i.e. − i D −i and Q i D Qi ; then we get 8 9 L >X Z − > Z −sat <2 1 i = (5.36) .Qi a/2 ps .a/ da C .Q L a/2 ps .a/ da Megr D 2 2 > i D1 −i 1 > −L : ;
2
1
Z Mesat D 2
C1
−sat
.Q L
2
a/2 ps .a/ da
(5.37)
If the probability of saturation satisﬁes the relation P[js.k/j > −sat ] − 1, then Mesat ' 0; introducing the change of variable b D Qi a, as −i D Qi C 1=2 and −i 1 D Qi 1=2, we have L=2 X Z 1=2 b2 ps .Qi b/ db (5.38) Meq ' Megr D 2
i D1 1=2
If 1 is small enough, then ps .Qi and assuming 2. P L=2
i D0
b/ ' ps .Qi / for jbj Ä R C1
1
1 2
(5.39)
ps .Qi /1/ ' Megr D 2
ps .b/ db D 1, we get !Z
1=2 1=2
L=2 X i D1
ps .Qi /1
12 b2 db ' 1 12
(5.40)
In conclusion, as in (5.34), we have Meq ' 12 12 (5.41)
assuming that −sat is large enough, so that the saturation error is negligible, and 1 is sufﬁciently small to verify (5.39).
Design of a uniform quantizer
Assuming the input s.k/ has zero mean and variance ¦s2 , and deﬁning the parameter5 kf D ¦s −sat (5.42)
the procedure of designing a uniform quantizer consists of three steps. 1. Determine −sat so that the saturation probability is sufﬁciently small: Psat D P[js.k/j > −sat ] − 1 (5.43)
5
Often the inverse 1=k f D −sat =¦s is called loading factor.
354
Chapter 5. Digital representation of waveforms Ð For example, if s.k/ 2 N 0; ¦s2 , then6
Psat
8 > 0:046 > > > > Ã > Â < −sat D 0:0027 D 2Q > ¦s > > > > > 0:000063 :
−sat D2 ¦s −sat D3 ¦s −sat D4 ¦s
2. Choose L so that the signaltoquantization error ratio assumes a desired value 3q D 3. Given L and k f , we obtain 1D 2¦s 2−sat D L kf L (5.45) Ms ¦2 ' 2 s D 3k 2 L 2 f Meq 1 =12 (5.44)
Signaltoquantization error ratio
For L D 2b , observing (5.44) we have the following result Â Ã ¦s .3q /d B ' 6:02 b C 4:77 C 20 log −sat
(5.46)
Recalling that this law considers only granular error, if we double the number of quantizer levels for a given loading factor, i.e. increase by one the number of bits b, the signaltoquantization error ratio increases by 6 dB. Example 5.2.1 Let s.k/ 2 U[ smax ; smax ]. Setting −sat D smax , we get p smax −sat D D 3 H) .3q /d B D 6:02 b ¦s ¦s Example 5.2.2 Let s.k/ D smax cos.2³ f 0 Tc k C '/. Setting −sat D smax , we get p smax −sat D D 2 H) .3q /d B D 6:02 b C 1:76 ¦s ¦s
(5.47)
(5.48)
Example 5.2.3 For s.k/ not limited in amplitude, and assuming Psat negligible for −sat D 4¦s , we get .3q /d B D 6:02 b
6
7:2
(5.49)
The function Q is deﬁned in Appendix 6.A.
in particular for b D 3 it turns out −sat D 2:3 ¦s . The optimization of 3q for the uniform quantization of a signal with a speciﬁed amplitude distribution yields the results given in Table 5. that coincide with the curves given by (5.26. We also note that the quantizers obtained by the optimization procedure and by the method on page 353 are in general different. Example 5. The parameter b is the number of bits of the quantizer.25 for various values of b. . and of a ¼law (¼ D 255) quantizer (continuous lines). versus the statistical power of the input signal is illustrated in Figure 5.5 [4]. The expression of 3q for granular noise only is given by (5.46) and (5. We note that for values of ¦s near −sat the approximation Meq ' Megr is no longer valid because Mesat becomes nonnegligible.5 we have optimum performance for 1=¦s D 0:1881. whereas for b D 8 we obtain −sat D 3:94 ¦s .25.2. ¦s2 and b D 5.2. The optimum point depends on b: we have −sat D `¦s . Instantaneous quantization 355 45 40 35 30 b=8 Λq (dB) 25 7 20 6 15 5 10 b=8 7 6 5 −60 −50 −40 −30 σs/ τsat (dB) −20 −10 0 Figure 5. and consequently the value of 3q decreases. Signaltoquantization error ratio versus ¦s =−sat of a uniform quantizer for granular noise only (dashed lines). the optimum value of 3q is obtained by determining the minimum of .25. Assuming a Laplacian signal we obtain the curves also shown in Figure 5. The plot of 3q .k/ 2 N 0.4 Ð For s. For the computation of Mesat we need to know the probability density function of s and apply (5. respectively. where ` increases with b.35).46) for ¦s − −sat . We note that for the more dispersive inputs the optimum value of 1 increases. As shown in Figure 5. given by (5.Megr C Mesat /=Ms as a function of ¦s =−sat .46). observing Table 5.5. for a Laplacian signal (dasheddotted lines). and consequently −sat D 2b 1 1 D 3:05¦s .64) for a uniform and a ¼law quantizer.
10 36.34 L 3. L: Laplacian.0743 U 6. good enough for telephone communications.00 17.1657 0.14 48.96 20.17 max.95 8. Determination of the optimum value of 3q for b D 5 and s.07 11.4330 0.0541 0. G: Gaussian.2800 0.0.99 31.2130 0. in an unvoiced spurt ¦s2 can be reduced by 20–30 dB.14 1.a/ G 4.35 0. for example.0660 0. where ¦s2 is computed for a voiced spurt.k/ 2 N .5400 0.08 30.7320 0.76 4.0308 L 1.02 12.23 35.06 24.0569 0.7957 0.60 25. Digital representation of waveforms Table 5.a/ G 1.78 13.12 42. a voice signal.3459 0.0271 0.2165 0.36 30.5956 0.7309 0.13 40.4610 0.57 29.25 0.3 σs/τsat 0.a/ (U: uniform.5 Optimal quantization step size and maximum corresponding value 3q of a uniform quantizer for different ps .38 24.89 1.5860 0.25 14. setting −sat D 4¦s . . yields 3q ' 33 dB for b D 7. However.3q /d B ps .9957 0.0874 0.83 35. and consequently 3q is degraded by an amount equivalent to 3–5 bit.4142 1.0961 0.26.356 Chapter 5. We conclude this section observing that for a nonstationary signal. : gamma).1041 0.1881 0.1547 1. ¦s /.3352 0.44 15.1273 0.0135 6 x 10 5 M 4 eq /M s 3 Me gr /Ms 2 1 Me sat /Ms 0 0.4 2 Figure 5.49 22. [From Jayant and Noll (1984).04 18.16 26.] b (bit/sample) U 1 2 3 4 5 6 7 8 −3 1opt =¦s ps .1083 0.8660 0.0549 1.27 19.40 9.01 7.
V is compared with the output signal of a ramp generator with slope 1=− .k/ D 01 stop stop Figure 5. when the generator output signal exceeds the level V .5. Example of encoding one level at a time for b D 3. 1. The sign of s.27. . let us consider the case illustrated in Figure 5.k/ D 00 if V < −2 ) c. The ﬁrst implementation encodes one level at a time and is illustrated in Figure 5.k/j. Starting with the counter initialized to zero. which gives the PCM encoding of js. where − is the clock period of a counter with b 1 bits. Implementations of uniform PCM encoders We now give three possible implementations of PCM encoders.28.2. Set V D js. Instantaneous quantization 357 Figure 5.k/j. For example.28 for b D 3: if V < −1 ) c.k/ can be encoded as a separate bit.27. the number of clock periods elapsed from the start represents c. Uniform PCM encoder: encoding one level at a time.k/.
We conclude this section explaining that the acronym PCM stands for pulse code modulation. To determine the bits c0 and c1 we can operate as follows: if V < −2 ) c1 D 0 if V < −1 C c1 21 1 ) c0 D 0 otherwise c1 D 1 otherwise c0 D 1 Only two comparisons are made. 3. These encoders are called ﬂash converters.k/ D 11 stop stop 1.29.c1 . Digital representation of waveforms clock threshold adjust logic reference voltage s(k)=V logic + comparator serial code bits Figure 5.29. which encodes one bit at a time.k/ D .358 Chapter 5.k/. A second possible implementation. if V < −3 ) c. PCM is not a modulation. the code word length is 2. The last implementation.30. In this scheme V is compared simultaneously with the 2b 1 quantizer thresholds: the outcome of this comparison is a word of 2b 1 bit formed by a sequence of “0” followed by a sequence of “1”.3 Nonuniform quantizers There are two observations that suggest the choice of a nonuniform quantizer. and c. PCM encoder: encoding one bit at a time. is given in Figure 5. but rather a coding method. Generally the number of comparisons depends on V and it is at most equal to 2b 2.k/ D 10 if V > −3 ) c. neglecting the sign bit. for b D 3. We waited until the end of the section to avoid confusion about the term modulation: in fact. For example. The ﬁrst refers to stationary signals with a nonuniform probability density function: for such signals . which encodes one code word of (b 1) bit at a time. is given in Figure 5. In this case b 1 comparisons are made: it is as if we were to explore a complete binary tree whose 2b 1 leaves represent the output levels. but the decision thresholds now depend on the choice of the previous bits. 5. through a logic network this word is mapped to a binary word of b 1 bits that yields the PCM encoding of s. c0 /.
the variation of the average power over different links is also of the order of 40 dB. a compression function may precede a uniform quantizer: at the decoder it is therefore necessary to have an expansion of the quantized signal. for which the ratio between instantaneous power (estimated over windows of tenths of milliseconds) and average power (estimated over the whole signal) can exhibit variations of several dB. 2. whereas it is small if the signal is small: as a result the ratio 3q tends to remain constant for a wide dynamic range of the input signal. As shown in Figure 5. The second refers to nonstationary signals. depicted in Figure 5.30.31. with the techniques illustrated in Figures 5. Under these conditions a quantizer with nonuniform characteristics. speech.5. Encoding of the nonuniformly quantized signal yq is obtained by a lookup table whose input is the uniformly quantized value xq .3. with a step size equal to the minimum step size of the desired nonuniform characteristic. The characteristic of Figure 5.32.g. for example.31. is more effective because the signaltoquantization error ratio 3q is almost independent of the instantaneous power. As also illustrated in Figure 5. Flash converter: encoding one word at a time. for a nonuniform quantizer the quantization error is large if the signal is large. The most popular method.3. In Section 5. Nonuniform quantizers 359 τ1 s(k)=V τ2 2 τ3 b1 to b1 decoding logic (b1)bit code word τ2b1 Figure 5.29 and 5.1 we will analyze in detail the last two methods. Three examples of implementation 1. moreover. uniform quantizers are suboptimum.31 can be implemented directly. 3. .33. e.30. as that depicted for example in Figure 5.. employs a uniform quantizer having a large number of levels.
Nonuniform quantizer characteristic with L D 8 levels. Digital representation of waveforms Figure 5.32b illustrates in detail the principle of Figure 5.32 we assume −sat D 1. We ﬁnd that the ideal characteristics of F[Ð] should be logarithmic. At the receiver the bit mapper gives yq . that yields the signal y D F. that must be expanded to yield a quantized version of s sq D F 1 [Q[y]] (5. The signal y is uniformly quantized and the code word given by the inverse bit mapper is transmitted.51) This quantization technique takes the name of companding from the steps of compressing and expanding.3.50) In Figure 5.52) . F[s] D ln s We consider the two blocks shown in Figure 5.360 Chapter 5.34.s/ (5. (5. The signal is ﬁrst compressed through a nonlinear function F. If −sat 6D 1 we need to normalize s to −sat . 5.32a.31.1 Companding techniques Figure 5.
3.k/ sgn[s. Nonuniform quantizers 361 Figure 5. that is Let s. Here −sat D 1 is assumed. Encoding.54) .k/j (5.32.5.53) (5.k/] y.k/ D ln js. (b) nonuniform quantizer characteristic implemented by companding and uniform quantization. (a) Use of a compression function F to implement a nonuniform quantizer.k/ D e y.
observing (5.k/ D s.k/ yields yq .k/ and the sign of s.k/ is given by the inverse bit mapping of yq .55).k/eeq .33.k/ sgn[s. Decoder.57) (5.k/j sgn[s.k/ is correctly received. (b) decoder.k/ The value c.k/ (5.k/] D js.55) . Nonuniform quantization by companding and uniform quantization: (a) PCM encoder. Assuming c. Figure 5.k/]eeq .k/] D ln js.k/ is given by sq .56) (5. Nonuniform quantizer implemented digitally using a uniform quantizer with small step size followed by a lookup table. The quantization of y.k/ ' 1 C eq .k/ If eq − 1. and assume the sign of the quantized signal is equal to that of s.k/. Digital representation of waveforms Figure 5.34.k/. the quantized version of s.k/ D Q[y.362 Chapter 5. then eeq .k/ D e yq .k/j C eq .
k/ represents the output error of the system. thus an approximation of the logarithmic law is usually adopted.k/j.3 0.9 1 Figure 5.7 A=87. Regulatory bodies have deﬁned two compression functions: 1. we get 3q D Ms 2 .1 0 0 0.4 0.k/] Meq E[eq (5.58) where eq .5. is adopted in Europe.3. and hence with s.8 0.k/] E[eq D 1 1 D 2 .k/ (5.8 0. Nonuniform quantizers 363 and sq .Ajsj/ 1 > : Ä jsj Ä 1 1 C ln.6 0. Alaw.35. . As eq . The sign is considered separately: sgn[y] D sgn[s] 1 (5. 8 Ajsj 1 > > 0 Ä jsj Ä < 1 C ln.28)). illustrated in Figure 5.k/s.56 0.2 0.4 0.35 for two values of A. Alaw (A D 87:56). We note that a logarithmic compression function generates a signal y with unbounded amplitude.k/s 2 .6 F(s) 0.1 0.A/ A (5.k/ C eq .A/ A This law.7 0.k/ (see (5.3 0. For −sat D 1.2 0.9 0. Consequently 3q does not depend on Ms .k/ is uncorrelated with the signal ln js.59) where from (5.k/s.5 A=1 0.k/ D s.41) we have that Meq depends only on the quantization step size 1.60) y D F[s] D > 1 C ln.61) 0.5 s 0.
considering only the granular error. We note that. which is well veriﬁed for a uniform input in the interval [ −sat .36.60).1 C ¼/] 10 log10 1 C C 3 ¼¦s ¼¦s (5.4 µ =0 0. Digital representation of waveforms 1 0.1 C ¼/ as in the ideal case (5. the standard value of ¼ is equal to 255.1 C ¼jsj/ (5.6 F(s) µ =5 0.8 0.3 0. for µ s × 1. is adopted in the United States and Canada. ¼law.63) ln. y D F[s] D Signaltoquantization error ratio Assuming the quantization error uniform within each decision interval.2 0. ¼law (¼ D 255).2 0. For −sat D 1.6 0.1 0 0 0. we have ln[µ s] F[s] D (5.7 0.8 µ =50 0.5 0. 2.364 Chapter 5.64) Curves of 3q versus the statistical power of the input signal are plotted for ¼ D 255 in Figure 5.62) ln.3 0. Similar behavior is exhibited by (5.9 1 Figure 5. Note that in the saturation region they coincide with the curves obtained for a . ln.1 0.36 for four values of ¼.1 C ¼/ This law.9 µ =255 0. we have ( Â Â Ã Ã) −sat −sat 2 p . The compression increases for higher values of ¼. we can see that for ¼law.3q /d B D 6:02b C 4:77 20 log10 [ln. −sat ]. illustrated in Figure 5.54).7 0.5 s 0.25.4 0.
by increasing ¼. but the maximum value decreases. then we have a mapping of the 5 bits of xq to the 3 bits of yq using the mapper (sign omitted) of Table 5. For decoding.2 In the standard nonlinear PCM. including also the sign we have 8 bit/sample. We also note that.37. this leads to a bit rate of the system equal to Rb D 64 kbit/s. 3q ' 38 dB for a wide range of values of ¦s . For a sampling frequency of Fc D 8 kHz. An effect not shown in Figure 5.39. for each code word yq we select only one code word xq . we select for each compressed Figure 5.25 is that.37.7. Nonuniform quantizers 365 uniform quantizer with Laplacian input. Distribution of quantization levels for a 3bit ¼law quantizer with ¼ D 40. The relation between s.3. . the plot of 3q becomes “ﬂatter”.6.k/ and yq . as shown in Figure 5. Using the standard compression laws. We emphasize that also in this case 3q increases by 6 dB with the increase of b by one. which represents the reconstructed value sq . we need to approximate the compression functions by piecewise linear functions. Digital compression An alternative method to the compressionquantization scheme is illustrated by an example in Figure 5. For decoding.k/ is obtained through a ﬁrst multibit (5 in ﬁgure) quantization to generate xq .5. a quantizer with 128 levels (7 bit/sample) is employed after the compression. a mapper with 12bit input and 8bit output is given in Table 5. For encoding. if b D 8. Observation 5.
− . : : : .3.2 Optimum quantizer in the MSE sense Assuming we know the probability density function of the input signal s.67) .7. (5. : : : . we desire to determine the parameters of the nonuniform quantizer that optimizes 3q . : : : . The problem. Digital representation of waveforms Table 5. In the literature there are other nonlinear PCM tables. 1. Signaltoquantization noise ratio mask We conclude this section by giving in Figure 5. respectively.k/. 5. 1. − − L (5. − . Coding of xq 0000 0001 0010 0011 0100 0101 1000 1001 1010 1011 1100 1101 1110 1111 Coding of yq 00 01 Coding of sq 0000 0001 10 0100 11 1011 code word a corresponding linear code word.39 two masks that indicate the minimum tolerable values of 3q (dB) for an Alaw quantizer (A D 87:6). −sat D 3:14 dBm. as given in the third column of Table 5.k/ s.k/] s.Q[s. stationary with variance ¦s2 .sq . as a function of ¦s (dBm) for input signals with Gaussian and sinusoidal distribution.366 Chapter 5.6 Example of nonlinear PCM from 4 to 2 bits (sign omitted).66) 2 2 that minimize the statistical power of the error (minimum meansquare error criterion) fQi g iD Meq D E[. illustrated in Figure 5.k//2 ] (5.65) − L D 1 − L D C1 L 1 0 1 2 1 2 1 2 2 and the quantization levels L L . consists in choosing the decision thresholds ² ¦ Á. that differ in the compression law or in the accuracy of the codes [4].40. − .k//2 ] D E[. and b D 8 (sign included). these masks are useful to verify the quantizer performance. : : : .
Linear code .yq / 111WXYZ 110WXYZ 101WXYZ 100WXYZ 011WXYZ 010WXYZ 001WXYZ 000WXYZ Coding of sq 1WXYZ011111 01WXYZ01111 001WXYZ0111 0001WXYZ011 00001WXYZ01 000001WXYZ0 0000001WXYZ 0000000WXYZ Assuming ps .70) i D1 −i 1 .5.a/ da (5.xq / 1WXYZ01WXYZ001WXYZ0001WXYZ00001WXYZ000001WXYZ0000001WXYZ 0000000WXYZ Compressed code .7 Non linear PCM from 11 to 7 bits (sign omitted). : : : .68) 2 −0 D 0 L (5.a/ even. Piecewise linear approximation of the Alaw compression function (A D 87:6).69) i D 1.38. (5.3. : : : . because of the symmetry of the problem we can halve the number of variables to be determined by setting ( L − i D −i 1 i D 1. Table 5. Q i D Qi 2 and L=2 X Z −i Meq D 2 . The 12bit encoded input signals are mapped into 8bit signals.Qi a/2 ps . Nonuniform quantizers 367 Figure 5.
L 2 L 2 1 (5.71) (5.67) are @Meq @−i @Meq @Qi D0 D0 i D 1. Necessary but not sufﬁcient conditions for minimizing (5. i D 1.368 Chapter 5. : : : .72) . 3q versus ¦s for an Alaw quantizer (A D 87:56) and b D 8. : : : . Digital representation of waveforms (a) Gaussian test signal (b) Sinusoidal test signal 2 Figure 5.39.
78) . and (5.a/ da D 0 (5.73) 2Qi −i 2 Qi C1 −i2 C 2Qi C1 −i ] D 0 (5.a/ da −0 Q1 D Z −1 ps . the equation 1 @Meq D2 2 @Qi yields Z Qi D Z −i −i 1 −i 1 Qi C Qi C1 2 (5.76) aps .74) that is −i D Conversely. These two rules are illustrated in Figure 5.75) Z −i 1 −i .Qi a/ ps .41.a/ (b D 3). 1.77) to get −1 from the integral equation Z −1 aps . From 1 @Meq D .77) ps .a/ da (5.77) sets Qi as the centroid of ps .−i / (5.−i /[Qi C −i2 −i /2 ps .40. −i ].a/ da −0 (5. Fixed Q1 “at random”.71) gives 2 ps .a/ da −i In other words.75) establishes that the optimal threshold lies in the middle of the interval between two adjacent output values.Qi 2 @−i (5. (5.−i / .5. Max algorithm We present now the Max algorithm to determine the decision thresholds and the optimum quantization levels. we use (5.Qi C1 −i /2 ps .Ð/ in the interval [−i 1 . Decision thresholds and output levels for a particular ps . Nonuniform quantizers 369 p (a) s τ4 = τ3 τ2 τ1 τ0 τ1 τ2 τ3 Q4 τ4=+ 8 8 a Q4 Q3 Q2 Q1 Q1 Q2 Q3 Figure 5.3.
77) Z C1 aps .a/ da 1 then the parameters determined are optimum.a/ da −i Qi C1 D Z −iC1 ps . For i D . Lloyd algorithm This algorithm uses (5. . 3.82) .77) to obtain −i C1 by the equation Z −iC1 aps . : : : . We choose an initial partition of the positive real axis: P1 D f−0 . Digital representation of waveforms ps(a) τi1 Qi τi Qi+1 τi+1 a Figure 5. if − L=2 D C1 satisﬁes the last equation (5. 2.41. 2.81) ps . −1 . (5. We use (5.80) C QL 1 Now. We set a relative error ž > 0 and D0 D 1.75) and (5.77). : : : .81) is not satisﬁed we must change our choice of Q1 in step 1) and repeat the procedure. Optimum decision thresholds and output levels for a given ps .a/ da QL D 2 −L 1 Z2 C1 −L 2 (5. Otherwise.L=2/ 2 1 we obtain (5. but in a different order.79) The procedure is iterated for i D 2. 3. From (5.L=2/ Q L D 2− L 2 2 1 2. − L=2 D C1g such that −0 D 0 < −1 < Ð Ð Ð < − L=2 D C1. if (5.a/.a/ da −i (5.370 Chapter 5. 1.75) we obtain Qi C1 D 2−i C Qi for i D 1.
The considerations that follow have this objective. We evaluate the distortion associated with the choice of P j and A j . If Dj 1 Dj Dj <ž j C 1. (5.84) then we stop the procedure. Nonuniform quantizers 371 3.Qi 1/ a/2 ps . assuming that ps . L 2 (5. Expression of q for a very ﬁne quantization For both algorithms it is important to initialize the various parameters near the optimum values.Qi a/ da 2 ps . We derive the optimum partition P j D f−0 .86) . not necessarily to the absolute minimum. : : : . We observe that the sequence D j > 0 is nonincreasing: hence the algorithm is converging.83) 6.87) .−i −i If the Qi s are optimum.85) −i . We set the iteration index j at 1. 4. From (5.a/ ' ps .70). however. otherwise we update the value of j : j 7.5.Qi a/2 ps .Ð/. : : : .75). it must be @Meq @Qi D0 i D 1.a/ da (5. 2 D j D E[eq ] D 2 L=2 XZ i D1 −i 1 −i . in addition to determining the optimum value of 3q for a nonuniform quantizer. unless some assumptions are made about ps . 8.3. 5.−i we have that Meq D 2 '2 L=2 XZ i D1 L=2 X i D1 −i 1 1/ for −i 1 Ä a < −i (5.77). We go back to step 4. We obtain the optimum alphabet A j D fQ1 . at least approximately for a number of bits sufﬁciently high. −1 . Q L=2 g for the partition P j using (5. : : : . − L=2 D C1g for the alphabet A j 1 using (5.a/ da Z −i 1 (5.
90) It is now a matter of ﬁnding the minimum of (5.92) with Meq given by (5.88) −i Correspondingly. By setting to zero the partial derivative of (5.1−i / (5.1−i / ' 2 Z 0 C1 ps . introducing the length of the ith decision interval .92) with respect to .89) where −0 D 0 and − L=2 D C1.1−i /. with the constraint that the decision intervals cover the whole positive axis.94) in (5.1−i /3 12 (5. L 2 1 (5.95) 1=3 1/ .90) with respect to . this is obtained by imposing that L=2 X i D1 2 ps .1−i / D 2 ½ ps Substituting (5.−i that yields p .91) Using the Lagrange multiplier method.−i 1 / . ps .93) .372 Chapter 5.−i 4 1 // D0 i D 1. it follows that L=2 X i D1 Meq D 2 ps .1−i /2 1=3 C ½.1−i / D −i −i 1 i D 1. : : : .1−i /. the cost function is " ½. we obtain ps .−i 1/ . L 2 (5.94) .91) yields p K ½D 2L (5.−i 1=3 1 / .a/ da D K 1=3 (5. Digital representation of waveforms and Z Qi D Z −i −i a da 1 −i D da 1 −i C −i 2 1 (5.f1−i g min Meq C ½ K 2 L=2 X i D1 !# 1=3 ps .90).−i 1/ (5. : : : .
98).97) For a quantizer optimized for a certain probability density function.5.101) For L sufﬁciently large. L 2 1 (5. moreover.0.98) where f f is a form factor related to the amplitude distribution of the normalized signal s . Actually (5. as in the case of a quantizer granular error.−i 1/ (5. the quantization step size 1 of the uniform quantizer is related to the compression law according to the relation 1 D F.opt D K3 12L 2 (5.k/ 2 N . −i ].100) assumes that F 0 does not vary considerably in the interval . In this case.−i 1/ Ä 1 F 0 . and for a high number of levels L D 2b (so that (5. Nonuniform quantizers 373 hence . s .96) and the minimum value of Meq is given by Meq. 3³ /.90) we have Meq D 2 L=2 X i D1 ps . ¦s2 /.−i / F.0.3.−i 1 .−i 1/ i D 1. : : : .100) where F 0 is the derivative of F.99) p Q In the Gaussian case.k/ D s.−i 1 / ½2 .1−i / 12 (5.opt D 22b ff (5.32. and f f D 2=.a/ da Q 1=3 (5. 1/.−i 1/ ' .a/ da [F 0 . even of the Alaw and ¼law types.1−i / F 0 . Q Q K3 ff D 12 Q K D Z C1 1 ps .100) in (5.85) holds). according to (5.96) indicates that the optimal thresholds are concentrated around the peak of the probability density.3 Equation (5.90) can be used to evaluate approximately Meq for a general quantizer characteristic.102) . the optimum value of 3q .k/ 2 N .a/]2 (5. s. follows the increment law of 6 dB per bit. Obviously (5. we have 3q D Ms Meq. Observation 5. the intervals become small and we get Meq ' 12 2 12 Z 0 −sat ps . from Figure 5.1−i / D K ps L 1=3 . Substituting (5.k/=¦s .
Performance of nonuniform quantizers A quantizer takes the name uniform. Digital representation of waveforms where 1 is related to a uniform quantizer parameters according to (5.64). the ratio 3q D Ms =Meq has the expression given in (5. has a different type of input.20 . respectively. that is with longer tails. and gamma input signal. leads to less closely spaced thresholds and levels. and 5. optimized for a speciﬁc input distribution. Laplacian.099 0.800 0.0345 0.10 are given parameter values of three optimum quantizers obtained by the Max or Lloyd method. ¦s D 1). [From Jayant and Noll (1984). The optimum value of 3q follows the 6b law only in the case of nonuniform quantizers for b ½ 4. we consider what happens if a quantizer. if it is optimized for input signals having the corresponding distribution.128 2 1 1. best Table 5.798 0.30 14. and consequently to a decrease of 3q .618 7 2.374 Chapter 5.117 0.069 8 1 2.344 0. and various numbers of levels [4]. It is left to the reader to show that for a uniform signal s in [ −sat . Finally.152 1.00955 3q (dB) 4. Concerning the increment of . In Tables 5. a more dispersive distribution.453 0.245 0.844 1.258 0.501 0. 5.733 Meq 0.401 2. For example.657 4 1 2.388 3 1.942 5 1.62 20.13q /d B D 6:02 b maxf3q gd B (5.748 1.42 the deviation .437 1.8.522 0. Laplacian. even for a small number of levels. a uniform quantizer. with increasing b the maximum of 3q occurs for a smaller ratio ¦s =−sat (due to the saturation error): this makes 13q vary with b and in fact it increases.9.756 0.8 Optimum quantizers for a signal with Gaussian dis2 tribution (ms D 0.103) for both uniform and nonuniform quantizers [4]. Gaussian.25).510 1. or gamma. Note that.40 9.] L 2 4 8 16 i −i Qi −i Qi −i Qi −i Qi 1 1 0. In the case of uniform quantizers. quantized according to the ¼law.050 0.982 0.256 6 1. we show in Figure 5.3q /d B according to the 6b law.363 0. −sat ]. for Gaussian.
567 0. will have very low performance for an input signal with a very dispersive distribution.33 11.899 0.087 1.54 12.920 0.878 1.061 8 1 6.822 7 5.] L 2 4 8 16 i −i Qi −i Qi −i Qi −i Qi 1 1 0.307 5 2.111 5 1.387 3 3.420 0.380 1.073 2 1 2. can have even higher performance for a less dispersive input signal.0196 3q (dB) 1.12 Table 5.405 3 2.223 1.155 0.0154 3q (dB) 3.1761 0. a nonuniform quantizer.432 Meq 0.533 0. optimized for a speciﬁc distribution.3.10 Optimum quantizers for a signal with gamma distri2 bution (ms D 0.345 1.77 6.] L 2 4 8 16 i −i Qi −i Qi −i Qi −i Qi 1 1 0.707 1.422 2.64 18.591 0.478 0.233 0. on the contrary.673 0.0712 0.834 1.195 Meq 0.795 4 1 4.253 0.01 7.127 0. [From Jayant and Noll (1984).07 for a uniform input.121 1.051 0.313 0.268 0.833 0.230 0. ¦s D 1).527 0. [From Jayant and Noll (1984).729 4 1 3.959 6 3.128 4.0545 0.178 7 3.633 1.500 0.390 1.089 2.017 8 1 4.9 Optimum quantizers for a signal with Laplacian dis2 tribution (ms D 0.577 1.725 3. ¦s D 1).597 2. Nonuniform quantizers 375 Table 5.47 17.124 2 1 1.6680 0.264 0.5.057 1.2326 0.578 6 2. .
optimized for a speciﬁc probability density function of the input signal.376 Chapter 5. .] Figure 5. All quantizers have 32 levels (b D 5) and are optimized for ¦s D 1.43. Gaussian (G) and gamma (0) [4]. Input type: uniform (U). Performance comparison of uniform (dashed line) and nonuniform (continuous line) quantizers. for Laplacian input. Digital representation of waveforms 16 Γ 14 12 10 ∆ Λq (dB) L 8 Γ L 6 G 4 G 2 U 0 1 2 3 4 b 5 6 7 Figure 5. [From Jayant and Noll (1984). Comparison of the signaltoquantization error ratio for uniform quantizer (dasheddotted line). ¼law (continuous line) and optimum nonuniform quantizer (dotted line).42. Laplacian (L).
Only a logarithmic quantizer is independent of the input signal level. for a decrease in the input statistical power.21 the quantizer characteristic is deﬁned as Figure 5.4 Adaptive quantization An alternative method to quantize a nonstationary signal consists in using an adaptive quantizer. Adaptive quantization and mapping: general scheme. is called an adaptive PCM or APCM. A comparison between uniform and nonuniform quantizers with Laplacian input is given in Figure 5. with reference to Figure 5. .36 with −sat =¦s D 1.9 for the nonuniform Laplacian type quantizer.44. 5. as we can see from (5.3. The performance also does not change for a wide range of the signal variance. We note that the optimum nonuniform quantizer gives best performance.5 for the uniform Laplacian type quantizer.44. All quantizers have 32 levels (b D 5) and are determined using: a) Table 5.k/ if errors are introduced Q by the binary channel. For a uniform quantizer.5. performance decreases according to the law 10 log Ms D 20 log ¦s (dB). as their characteristic is of logarithmic type (see Section 5. where c. the idea is that of varying with time the quantization step size 1.1).k/ 6D c.4. Adaptive quantization 377 The 0 quantizers have performance that is almost independent of the type of input. which has parameters that are adapted (over short periods) to the level of the input signal.k/ so that the quantizer characteristic adapts to the statistical power of the input signal.98).43. even if this happens in a short range of values ¦s . c) the ¼ (¼ D 255) compression law of Figure 5. b) Table 5. General scheme The overall scheme is given in Figure 5.k/ is the quantization step size at instant k. If 1. The corresponding coding scheme.
1 2 L i D 1. for example.k/g 2 is generated with a constant statistical power. and 5.104) 1 If 1opt is the optimum value of 1 for a given amplitude distribution of the input signal assuming ¦s D 1 (see Table 5.k/ : 2 thresholds: −i .:::. : : : .5). : : : .opt ¦s . 0. 1.k/ iD L L C 1. we need to change the levels and thresholds according to the relations: Qi .k/ D Â > > i 1 1.9. 5.k/ > < 2 Ã output levels: Qi .k/ D Qi. : : : .opt g are given in Tables 5.k/ D 1opt ¦s .8.k/ −i .k/ (5. Therefore we let g.opt g and f−i. Digital representation of waveforms Ã 8Â > i C 1 1.105) For a nonuniform quantizer. 2 iD (5. ¦ y D 1.106) where fQi. both methods require computing the statistical power ¦s2 of the input signal.k/ (5.k/ (5. As illustrated in Figure 5. an alternative to the scheme of Figure 5.10 for various input amplitude distributions.k/ D i 1.107) However. The adaptive quantizers are classiﬁed as: Figure 5. 1. so that a signal fy. 2 2 L .45. and ¦s .k/ D −i. ﬁxed quantization and mapping.k/ is the standard deviation of the signal at instant k. . Adaptive gain.378 Chapter 5.45.44 is the following: the quantizer is ﬁxed and the input is scaled by an adaptive gain g.opt ¦s .k/ D 1 ¦s . then we can use the following rule 1.
4. 5.k//q (or gq .k/). (a) Figure 5. in Figure 5. Adaptive quantization 379 ž feedforward.k/g itself.k/ 6D sq .4. the signals at the output of the quantizer.46 and Figure 5. APCM scheme with feedforward adaptive quantizer: a) encoder. if ¦s is estimated by observing the signal fs.k//q .k//q (or gq . APCM scheme with feedforward adaptive gain and ﬁxed quantizer: a) encoder. . i. Q 2. if ¦s is estimated by observing fsq .44 and Figure 5. b) decoder. respectively. The main difﬁculty in the two methods is that we need to quantize also the value of ¦s .¦s . because of digital channel errors on both c. and how many bits must be used to represent . We emphasize that: 1.¦s . the system bit rate is now the sum of the bit rate of c.k/ and .k/ and . Figure 5.k/ so that it can be coded and transmitted over a binary channel.1 Feedforward adaptive quantizer The feedforward methods for the two adaptive schemes of Figure 5. that is what frequency is required to sample ¦s .k/g.45 are shown.46.47.¦s .k/.k/.47.k/) it may happen that sq .e.5.k/ D Q[s. ž feedback. b) decoder. we need to determine the update frequency of ¦s .k/]g or fc. 3.
k/ (5.k D/ D 1 K k X nDk . it must be ¦max ½ 100 ¦min (5.¦s . in order to keep 3q relatively constant for a change of 40 dB in the input level. for a > 0:9.1 a/=.2³ Tc / (Hz) 40 20 10 D 0:9688 D 0:9844 D 0:9922 . this introduces a latency in the coding system that is not always tolerable. for three values of a. the corresponding values of K 1 and B¦ for 1=Tc D 8 kHz.k/. whereas ¦max controls the saturation error level.k//q g or fgq . For speech signals sampled at 8 kHz. Table 5.11. Performance With the constraint that ¦s varies within a speciﬁc range. To determine the update frequency of ¦s2 . and coding the values given by (5. quantizing.k/ in (5.380 Chapter 5.n/ (5.108) where D expresses a certain lead of the estimate with respect to the last available sample: typically D D . from (1. For example.109) every K instants. however.109) Typically in this case we choose D D 0.K 1/ s 2 . If D D K 1.1. a K 1 1 1 2 2 2 5 6 7 Time constant 1 D 1=. from (1. Typically. For an exponential ﬁlter instead.12 shows the performance of different ﬁxed and adaptive 8level (b D 3) quantizers.109) is equal to B¦ D .471) with the length of the rectangular window. windows usually do not overlap. Digital representation of waveforms The data sequence that represents f. K samples need to be stored in a buffer and then the average power must be computed: obviously.1 a/s 2 .11 Time constant and bandwidth of a discretetime exponential ﬁlter with parameter a and sampling frequency 8 kHz. hence ¦s is updated every K samples. we prefer to determine a from the equivalence (1.K 1/=2 or D D K 1.462) we have ¦s2 .k/ are given in Section 1. The estimate of the signal power is obtained by a rectangular window with D D K 1. In Table 5.1 a/ (samples) 32 64 128 Filter bandwidth B¦ D . Two methods to estimate ¦s2 . ¦min Ä ¦s Ä ¦max . that gives a D 1 K 1 1 : this means decimating. the decimation and quantization of ¦s2 . we .k 1 D/ C .2³ Tc / .1 a/ recall that the 3 dB bandwidth of ¦s2 . using a rectangular window of K samples. Moreover.k/g is called side information.468) we have ¦s2 .110) Actually ¦min controls the quantization error level for small input values (idle noise).k/ Table 5.k D/ D a¦s2 .11 we give.
the feedback method estimates ¦s from the knowledge of fsq .48. −sat =¦s D 8) Gaussian (3q. If K − 128 the side information becomes excessive.4 – 12.5 are not considered.k/]g or fc.opt D 12:6 dB) uniform Q Gaussian (3q. where fs.n/g is substituted by fsq .4. We make the following observations: ž there is no need to transmit ¦s .opt D 11:4 dB) 9.8 11.3 14.k/.3 9.2 Feedback adaptive quantizers As illustrated in Figure 5.k/ b=3 3q (dB) Nonadaptive Adaptive K D 128 (16 ms) Adaptive K D 1024 (128 ms) nonuniform Q ¼ law (¼ D 100.7 13.109). APCM scheme with feedforward adaptive quantizer.5.k/. conversely there is a performance loss of 3 dB for K D 1024.108) or (5.1 12. this signal is available only for n Ä k 1: Figure 5.opt D 14:6 dB) Laplacian (3q.9 6.opt D 14:3 dB) Laplacian (3q. Adaptive quantization 381 Table 5. . we note that using an adaptive Gaussian quantizer with K D 128 we get 8 dB improvement over a nonadaptive quantizer.7 7.k/ affects not only the identiﬁcation of the quantized level. Although b D 3 is a small value to draw conclusions.k/ D Q[s.5 7. a possible method consists in applying (5.k/g.n/g. but also the scaling factor ¦s .3 11.48. Concerning the estimate of ¦s . However. 5. ž a transmission error on c. Speech s.4.4 – 15 13. therefore feedback methods do not require the transmission of side information.12 Performance comparison of ﬁxed and adaptive quantizers for speech.
n/. Likewise.opt Q4. Because of the lag in estimating the level of the input signal and the computational complexity of the method itself.K 1/ sq .k 1/ .2 0.4 0.382 Chapter 5.2 0.8 0. For example.49. Output levels and optimum decision thresholds for Gaussian s.k/j will be very different with respect to (5.6 0.opt τ0.a/ da D pc4 : P[jc. 0. : : : .k/ with unit variance (b D 3).49 for b D 3.3 If ¦s changes suddenly.7 0. the recursive estimate (5. L=2g.k/ D 1=K nD.1 0 ps (a) σs =1 2 Q1.k q q 2 1/ C .opt τ3. 2.1 8 > P[jc.109) becomes ¦s2 . the distribution of jc. while P[jc.6 0. if ¦s < 1 it will be P[jc.opt Q3.4 0.1 a/sq .5 0.111) : : > Z C1 > > > > ps . we present now an alternative method to estimate ¦s adaptively.a/ da D pc1 > > > −0 D0 < : : : : (5.111). Estimate of σs (k) For an input with ¦s D 1 we compute the discrete amplitude distribution of the code words for a quantizer with 2b levels and jc.8 0. As illustrated in Figure 5.3 0.3 0.5 0.108) becomes ¦s2 .opt τ2. Digital representation of waveforms Pk 1 2 this implies that the estimate (5.k/j D 4] D 2 −opt.1 0 τ1 (k) τ2 (k) τ3 (k) a Figure 5.k/j 2 f1. let Z −opt.k/j D 4] − pc4 .k/j D 1] × pc1 .opt τ1.k 1/.opt Q2.opt ps (a) σs<1 2 a 0. with q a lag of one sample. .7 0.k/ D a¦s2 .k/j D 1] D 2 > ps .
i D 1.k/ D E[ln p[jc.4. L=2. i D 1.116) 1/j] C ln ¦sq .112) in which fp[i]g. thus p[1] < 1. For example. what we do is vary ¦sq by small steps imposing bounds to the variations.115) pc L=2 D1 (5. thus p[L=2] > 1. L=2.k 1/j D L=2.p[L=2]/ In fact. then ¦sq must decrease to reduce the quantizer range.k 1/ (5. : : : . The objective is therefore that of changing ¦sq .k/ so that the optimal distribution is obtained for jc. In practice. that is: ¦min Ä ¦sq . Intuitively it should be .50. are suitable parameters.k 1/j]] C E[ln ¦sq .k 1/j]¦sq . then ¦sq must increase to extend the quantizer range.k 1/j D 1.k as in (5. from (5. : : : .50.k/ Ä ¦max (5.114) In steady state we expect that E[ln ¦sq .p[2]/ pc2 : : : .5.k/ D p[jc. if it is jc.k/] D E[ln ¦sq .k 1/ (5.k/ D ln p[jc. The algorithm proposed by Jayant [4]. is given by ¦sq .k from which E[ln ¦sq ].113) The problem consists now in choosing the parameters fp[i]g. 1/j]] D L=2 X i D1 1/].p[1]/ pc1 .117) .k E[ln p[jc. therefore it must be pci ln p[i] D 0 (5. if instead it is jc. Adaptive quantizer where ¦s is estimated using the code words. Adaptive quantization 383 Figure 5.k/j. illustrated in Figure 5.112) it follows that ln ¦sq .114).k 1/] (5.
Summarizing. From the input sample s. on the other hand. p 3 2 1 0 0 1 q Figure 5.106) is determined and. and by (5. At the reception of c.i/g D f1=7.9. 3=7.k/g are also known. i D 1.k/ the index i in (5. Then ¦sq . L=2 [4].k/ is produced by the quantizer characteristic (see Figure 5. Experimental measurements on speech signals indicate that this feedback adaptive scheme offers performance similar to that of a feedforward scheme.k/. i D 1. c. ¦sq . Interval of the multiplier parameters in the quantization of the speech signal as a function of the parameters fq. are in correspondence of the values of fq.106) the decision thresholds f−i .112) and the thresholds are updated: the quantizer is now ready for the next sample s. thus it can adapt very quickly to changes in the mean signal level.112). Jayant also gave the values of the parameters fp[i]g. the possible output values are also known from the knowledge of ¦sq .] . 4. we note that there is a large interval of possible values for p[i]. 5=7.k C 1/ is computed by (5.i/ D L 1 2 1 ² L 1 .k/. Digital representation of waveforms Based on numerous tests on speech signals.:::.k/ (see (5. 1g.52).i/g. : : : .1 ¦ (5. it is strongly affected by the errors introduced by the binary channel. i D 1.k C 1/ by (5.k C 1/. at instant k. [From Jayant and Noll (1984).384 Chapter 5. Therefore p[1] is in the range from 0. Let 2i q. : : : . L=2. consequently. and p[4] in the range from 1. For example.8 to 2.8 to 0. sq .i/g [4]. for L D 8 the values of fp[i]g. 1 L 3 1 .51 the values of fp[i]g are given in correspondence of fq.k/ is known. : : : .106)).118) In Figure 5.9.51. An advantage of the algorithm of Jayant is that it is sequential. in turn the receiver updates the value of ¦sq . especially if the index i is large. At the receiver.
k/g is involved. obtained by an ADC. To avoid introducing a new signal. the preliminary conversion by an ADC is omitted in all our schemes. even nonlinear predictors.k/ O (5.τ3(k) .120) ci z i (5. Differential coding (DPCM) 385 Q (k) 4 sq=Q[s] 010 011 p[4] Q (k) 3 p[3] 001 Q (k) 2 p[2] 000 Q (k) 1 p[1] .τ1(k) 100 τ1(k) τ2(k) τ3(k) s p[1] 101 p[2] 110 .Q (k) 2 p[3] 111 .k/ Considering the ztransform. For each output level the PCM code word and the corresponding value of p are given.k/ be the prediction signal :8 s .52. the ﬁnite number of bits of this preliminary quantization should not affect further processing. 5. 8 The considerations presented in this section are valid for any predictor. Inputoutput characteristic of a 3bit adaptive quantizer. as well as in some schemes of the previous section on adaptive quantization. it is desirable to perform the various operations in the digital domain on a linear PCM binary representation of the various samples. let s .k/ D O N X i D1 ci s.5.5.7 With reference to Figure 5.Q (k) 1 .81) the prediction error is deﬁned as f . for a linear predictor with N coefﬁcients. let C.53.121) 7 In the following sections.k/ D s.z/ D N X i D1 s .τ2(k) .Q (k) 4 Figure 5.5 Differential coding (DPCM) The basic idea consists in quantizing the prediction error signal rather than the signal O itself. Obviously. when processing of the input samples fs.k i/ (5. .119) From (2.Q (k) 3 p[4] .
In the case C.k/ and O s . Computation of the prediction error signal f.k/ and prediction s .k/ D s.123) (5. It is interesting to rearrange the scheme of Figure 5. then O S.124).386 Chapter 5.k/.k/ is called the reconstruction signal and is given by sq .z/] is the prediction error ﬁlter.120) and (5. s(k) +  f(k) f(k) sq(k) + f(k) + + sq(k) ^ s(k) c + ^ s(k) c (a) (b) Figure 5.k/ that is the reconstructed signal coincides with the input signal.54b shows how to obtain the reconstruction signal from f .125) 5. We will now quantize the signal f f .55. according to (5.54a illustrates how the prediction error is obtained starting from input s. it is easy to prove that in the scheme O of Figure 5. . (a) Prediction error ﬁlter.81).z/[1 C.z/ and F.k/ O (5.5.1 Conﬁguration with feedback quantizer With reference to the scheme of Figure 5.k/g.z/ D S.z/S.k/ coincides with the difference between two consecutive input samples.k/ D s . [1 C. (5.z/ D z 1 .54 we have sq . the following relations hold.k/.53 in the equivalent scheme of Figure 5. where sq . (b) Inverse prediction error ﬁlter.122) Recalling (2.124) Figure 5. f .54. Digital representation of waveforms s(k) +  f(k) c ^ s(k) Figure 5.z/ D C.k/.53. From (5.k/ C f . Figure 5.124).z/] (5.54. that is for a predictor with a single coefﬁcient equal to one.
129) f q .k/] sq .k/ O s . the quantized prediction error is transmitted over the binary channel.k/ D Q[ f .126) (5.k/ D s . f .k/ C f q .k/] (5.k/ C f q .k C 1 In other words.55.k/ s .k C 1/ D O Decoder: sq . DPCM scheme with quantizer inserted in the feedback loop: (a) encoder.132) be the quantization error and 3q.128) i/ (5.k C 1 (5. Differential coding (DPCM) 387 Figure 5. Let eq.130) i/ (5.k/ (5. f D E[ f 2 .k/ D s . f .127) (5.5.k/ D f q . f the signaltoquantization error ratio 3q.k/] 2 E[eq.133) .k/ D s.k/ O (5.131) ci sq . (b) decoder.k C 1/ D O N X i D1 N X i D1 ci sq . Encoder: f .k/ f .k/ O s .5.
f is maximized by selecting the thresholds and the output values according to the techniques given in Section 5.134) To summarize. that minimize M f . can be derived from (5. Therefore G p can be sufﬁciently high also for a predictor with a reduced complexity.133) 3q.k/ O (5.k/ D s.k/. f < Meq and the DPCM scheme presents an advantage over PCM.136). For example in the case N D 1.k/ C eq. From (5.k/g. assuming known Ms and G p .138) Regarding the predictor.k/ C f . useful in scaling the quantizer characteristic. Mf D Ms Gp (5.137) shows that to obtain a given 3q we can use a quantizer with a few levels. not of s.128) and (5.k/ D s. f M f Meq.k/ C eq. 3q. this will cause a deterioration of G p with O respect to the ideal case fsq . . Observing (5. f depends on the number of quantizer levels. (5.1/. assuming the distribution of f f . if M f < Ms then also Meq.139) ignoring the effect of the quantizer. the reconstruction signal is different from the input signal.k/ 6D s.91).k/. using (5.3.k/ D s . f is only a function of the number of bits and of the probability density function of f f .k/g is highly predictable. and the reconstruction error (or noise) depends on the quantization of f .k/g. f .k/g and not fs.137) Ms Mf (5. the optimum value of c1 is given by ². the correlation coefﬁcient of the input signal at lag 1. Digital representation of waveforms Recalling (5. Then we have Gp D 1 1 ² 2 .k/g. Consequently. We observe that the input to the ﬁlter that yields s .134) the signaltonoise ratio is given by 3q D Given Gp D called prediction gain.k/g is known. N . we have sq .k/g. : : : . it follows that 3q D G p 3q.126). In particular the statistical power of f f . with the normalization by the standard deviation of f f .k/g. f (5. sq . For the quantizer. f . This decrease will be more prominent the larger feq.k/g. f . which in turn determine the transmission bit rate.135) where observing (5.k/g.k/ D s.1/ (5.k/ in (5. i D 1. 3q. f (5.k/g. f g will be with respect to fs.136) Ms Ms M f D Meq. If we ignore the dependence of G p on feq.132).k/g. recalling (2.98) we know that for an optimum quantizer. that is for fsq .k/. once the number of coefﬁcients N is ﬁxed. whereas G p depends on the predictor complexity and on the correlation sequence of the input fs.129) is fsq . we need to determine the coefﬁcients fci g. provided the input fs.388 Chapter 5.
k/g.k/g instead of fs. Figure 5. c1 D ².55.1/ D 0:85.85 0.2 Alternative conﬁguration If we use few quantization levels.k/ s. O For a simple predictor with N D 1 and c1 D 1.56a.1 Figure 5.56.5. Evidently.5.56b.6 7. the DPCM scheme allows us to use a transmission bit rate lower than that of PCM. fsq . In the speciﬁc case. for an input with ². with (b) a 6 level quantizer.1/.90 0.1// (dB) 5. having as input fsq .13 the values of G p for three values of ².13 Prediction gain with N D 1 for three values of ².k 1/. can give poor performance because of the large quantization .1/. the maximum level of the quantizer is instead related to the slope overload distortion in the sense that if Q L=2 is not sufﬁciently large.1/ 0.k/g presents a slope different from that of fs. the predictor of the scheme of Figure 5.1/ D 0 there is no advantage in using the DPCM scheme. for an input having ².k/g in the instants of maximum variation. a simple predictor with one coefﬁcient yields a prediction gain equivalent to one bit of the quantizer: consequently.1 ² 2 .k/ D sq . We note that the minimum level of the quantizer still determines the statistical power of the granular noise in fsq . We note that. Differential coding (DPCM) 389 Table 5.k 1/j.56a illustrates the behavior of the reconstruction signal after DPCM with the sixlevel quantizer shown in Figure 5.5. as shown in Figure 5. being Q L=2 =Tc < max js. 5.95 G p D 1=. We give in Table 5. (a) Reconstruction signal for a DPCM. given the total 3q .2 10. the output signal cannot follow the rapid changes of the input signal. hence s .k/g.
k/ D f q .k/ C so .k C 1/ D O Decoder: O sq.k/ D Q[ f .k/ D s.k/ O (5. Digital representation of waveforms Figure 5.57.145) ci sq. noise present in fsq . However. s . from (5.57. In fact.k/ so .k/g.k/.k/ is obtained from the input signal without errors.o .144) i/ (5. Encoder: f .k/ D s.k/ s .k/ 6D s .k/] sq .390 Chapter 5.k/ s .k C 1 At the encoder.o . b) decoder.k C 1/ D O N X i D1 N X i D1 ci s.141) (5. DPCM scheme with quantizer inserted after the feedback loop: a) encoder.142) i/ (5.143) f q .140) (5.k C 1 (5. An alternative consists in using the scheme of Figure 5. the O O prediction signal reconstructed at the decoder is so . where the following relations hold.144) and O .
k/ C [Oo .148) We introduce the following vectors and matrices.k/ C f . and consequently the impulse response is very long.5. Vector of prediction coefﬁcients c D [c1 .k 1/ 6D s. : : : .k/ D so . which in the case of a feedback quantizer system is given by sq .55 because of errors introduced by the binary channel.k/ 6D s .i/.s.k/ s .k/. though to a lesser extent as compared to the scheme of Figure 5. depending on the function C.k/ D so . This is difﬁcult to achieve if the transfer function 1=[1 C.i/ is deﬁned in (1.147) where sq . the difference between O the prediction signals. M f D E[.k/g at the encoder is affected by a smaller disturbance.i/ D s .k/ D s.N /]T where ².k i/ (5. the inverse prediction error ﬁlter must suppress the propagation of such errors in a short time interval.k//2 ] O (5.4 Note that the same problem mentioned above may occur also in the scheme of Figure 5. For both conﬁgurations. so .k 1/ then O O sq.145).57.k/. ².k/ C eq. : : : .k/. f . Observation 5. as the signal f f q .k/ O D s. Differential coding (DPCM) 391 (5.k/ s .5. we choose the coefﬁcients fci g that minimize the statistical power of the prediction error. may be nonnegligible.k/] C eq.k/ C eq.150) (5.k/ is the reconstruction signal.k/ O (5.1/. for i Ä k 1.o . the prediction signal s .540). however.k/ O s . f .k/. and consequently so . As a result the output O O sq.k/ D O N X i D1 ci sq .k/ C s. f .k/ C eq.z/. even if by chance so .k/g ρ D [².146) can assume values that are quite different from s. 5. O O A difﬁculty of the scheme is that.3 Expression of the optimum coefﬁcients For linear predictors.z/] has poles near the unit circle. For the design of the predictor.k/ s s .5.k 1/. f .k 1/ 6D f . as f q . c N ]T Vector of correlation coefﬁcients of fs.149) . (5.o .k/ is given by O s .
2. Effects due to the presence of the quantizer Observing (5.1 T copt ρ/ (5.153) Recalling the analysis of Section 2.154) (5.156) In general it is very difﬁcult to analyze the effects of feq.158) 8 Assuming for fs.1 D ². f Žn rsq .n/ C 3q 1 Žn (5.n/ D rs . f .155) The difﬁculty of this formulation is that to determine the solution we need to know the value of 3q (see (5. except in the case N D 1.392 Chapter 5.153)).78) copt D ρ The corresponding minimum value of M f is obtained from (2. M f D Ms .79).N 2/ 6 3q D6 6 : : : :: : : : 6 : : : 6 Â : Ã 4 1 ². f .152) . We may consider the solution with the quantizer omitted.N 1/ 6 1 C 3q Â Ã 6 6 1 6 ².k/g and feq. Digital representation of waveforms Correlation matrix of sq .2.k/g the correlations are expressed by (5.2.1/ (5.k/g.154) becomes Â Ã 1 D ².k/g on G p . we get Dividing by Ms we obtain rsq .155).1/ : : : ².N 1/ ::: ².27) and (5.1/ 1 C 3q 3 7 7 7 7 7 7 7 7 7 5 (5.154) and (5.1=3q / (5.2.28). and depends only on the second order statistic of fs.n/ Ms D ².151) (5. In this case some efﬁcient algorithms to determine c and M f in (5.1/ 1 C .1 and 2.n/ C Meq.155) are given in Sections 2. hence 3q D 1.157) copt. the prediction gain is given by Gp D Ms D Mf 1 1 T copt ρ (5.1/ 1C : : : ². for which (5.1 1 C 3q Then copt. the optimum prediction coefﬁcients are given by the matrix equation (2. normalized by Ms 9 Ã 2Â 1 ².
For c1 D 1 we have 1 Â 1 1 ² 2 . f Ã (5.1// 1 3q. f D Ms =3q . and saturates for N ½ 2.s.k/sq . f Ã (5.1 c1 . f / is due only to the presence of the quantizer. in fact.1/ > 1=2.5.s. Note that (5. then copt. if the system is very noisy.5.1/ D 1 1 ² 2 .159) The above relations show that if 3q is small.163) We note that the choice c1 D 1 leads to a simple implementation of the predictor: however.1 ². f .k/ D O N X i D1 ci .1/ 1 C 1=3q (5.162) ² 2 . f depends only on the number of quantizer levels.1 D ². Differential coding (DPCM) 393 and Gp D 1 1 copt. f . f / 2 2c1 ². the prediction gain for a ﬁxed predictor is between 5 and 7 dB.1 ².1/.1 2. observing (5.160) we obtain Gp D 1 1 2 . Therefore we have s .135) it follows Meq.161) hence 3q. Only for 3q D 1 it is copt.4 Adaptive predictors In adaptive differential PCM (ADPCM). Various experiments with speech have demonstrated that for very long observations. 5.1/=3q.k 1/ C eq. For N D 1 and any c1 it is M f D E[.160) 2 2 2c1 ².k/ ' Ms . that is.1/ ² 2 .1 is small and G p tends to 1.164) . the expression is complicate and will not be given here. where from (5. f As from (5. It may occasionally happen that a suboptimum value is assigned to c1 : we will try to evaluate the corresponding value of G p . Â 1 1 Gp D 2. this choice results in G p > 1 only if ².137) 3q D G p 3q.1/ C c1 / C c1 Meq.c1 =3q. the predictor is timevarying.1/ we have Gp D where the factor .k i/ (5.1/ 3q. speech is a nonstationary signal and adaptive predictors should be used.k 1///2 ] (5. Rather we will derive G p for two values of c1 .1/ C c1 (5. of the order of one second. 1. For c1 D ².5.161) allows the computation of the optimum value of c1 for a predictor with N D 1 in the presence of the quantizer: however.
These quantities.59b and in Figure 5. c N ]T is chosen to minimize M f over short intervals within which the signal fs. the power measured on windows of 128 samples is shown in Figure 5. sq .60.2).478).¦ f /q : the system is now ready to encode the samples of the observation window in sequence.1.k/g to the decoder output fQq . We consider an observation window for the signal fs.59a: we note that the speech level exhibits a dynamic range of 30–40 dB and rapidly changes value. G p can reach the value of 20–30 dB.k coefﬁcient adaptation is given by c.59c. In particular. respectively. We note that. for some voiced spurts. : : : .k/g of K samples. Speech signals have slowlyvarying spectral characteristics and can be assumed as stationary over intervals of the order of 5–25 ms.i/g for a window of K samples and apply the same procedure as the feedforward method.k/sq . Deﬁning sq . Adaptive feedforward predictors The general scheme is illustrated in Figure 5.k/ D [sq . In general. Based on these samples the input autocorrelation function is estimated up to lag N using (1. even for unvoiced spurts that present small correlation. this method requires too many computations. The ﬁxed predictor is determined by considering the statistic of the whole signal. An alternative could be that of estimating at every instant k the correlation of fsq . must be sent to the receiver to reconstruct the signal. : : : .i/g. thus within certain windows the prediction gain is even less than 1. the observation is now available only for instants i < k.k/g is quasistationary. and consequently this method is not suitable to track rapid changes of the input statistic. and calculate c.0/ s (5.394 Chapter 5. The prediction gain in the absence of the quantizer is shown for a ﬁxed predictor with N D 3 and an adaptive predictor with N D 10 in Figure 5. give the parameters of the predictor cq and of the quantizer . together with the quantized parameters of the system.59.166) 1/. then we solve the system of equations (5.k/. is illustrated in Figure 5. The adaptive predictor is estimated at every window by the feedforward method and yields G p > 1. of the sequential adaptive type.58.k/ 0<¼< 2 N r2q . The digital representation of f f q . for speech signals sampled at 8 kHz. s The performance improvement obtained by using an adaptive scheme is illustrated in Figure 5. this system introduces a minimum delay from fs. for speech we choose K Tc ' 10–20 ms. Sequential adaptive feedback predictors Also for adaptive feedback predictors we could observe fsq .k/C µ f q .165) .k/g.k N /] (5. however. and N ' 10. after being appropriately quantized for ﬁnite precision representation. i < k. Another simple alternative. where the predictor is adapted by the LMS algorithm (see Section 3. Digital representation of waveforms The vector c D [c1 . however.k/g the procedure is repeated.k/g equal to K samples.k C 1/ D ¼1 c. For the next K samples of fs.154) to obtain the coefﬁcients c and the statistical power of the prediction error. Also for ADPCM two strategies emerge. Not considering the computation time.
in Q decoding the same equations are used with fQq .k/. where ¼1 Ä 1 controls the stability of the decoder if.14 gives the algorithm.k/.k/ in place of sq . while Figure 5.k/ 6D f q . Differential coding (DPCM) 395 Figure 5. ADPCM scheme with feedforward adaptation of both predictor and quantizer: (a) encoder. (b) decoder.5. it occasionally happens fQq .58.5. .k/ in place of f q . because of binary channel errors. Table 5.k/ and therefore sq .61 illustrates the implementation.
k/ O f q . This observation does not apply to speech.59.k/ O sq .0/ D 0 sq .k/ s .k C 1/sq .k/ D s . .k/C µ f q .5 For a stationary input. Digital representation of waveforms Figure 5.0/) s .0/ D s. the LMS adaptive prediction can be used to easily determine the predictor coefﬁcients.0/ D 0 O f . which presents characteristics that may change very rapidly with time.k C 1/ D ¼1 c. : : : Observation 5. (c) an adaptive predictor (N D 10).k/ D Q[ f .k/ C f q .14 Adaptation equations of the LMS adaptive predictor. (a) Speech level measured on windows of 128 samples. 1.k/ s . as for example a modem signal. Table 5.k C 1/ O For k D 0.k/sq .k/] c.0/ D 0 (or sq . it is better to switch off the adaptation. Initialization c.k/ D s.k C 1/ D cT . and corresponding prediction gain Gp for: (b) a ﬁxed predictor (N D 3). For these measurements the quantizer was removed. however. once the convergence is reached.396 Chapter 5.
Differential coding (DPCM) 397 Figure 5.60. .5. ADPCM scheme with feedback adaptation for both predictor and quantizer: (a) encoder. (b) decoder.5.
k/ D We consider two cases. The predictor refers to an input model whose samples are given by s.61.5.k i/ C w.167) which has only poles (neglecting zeros at the origin). 5. Obviously in both cases the quantizer is nonuniform.398 Chapter 5. LMS adaptive predictor. a 5bit ADPCM scheme yields the same quality as a 7bit PCM. Allpole predictor The predictor considered in the previous section implies an inverse prediction error ﬁlter or synthesis ﬁlter (see Figure 2. For further study on various signal models (AR. N X i D1 ai s.z/ D F. Digital representation of waveforms Figure 5.z/ (5. Performance Objective and subjective experiments conducted on speech signals sampled at 8 kHz have indicated that adopting ADPCM rather than PCM leads to a saving of 2 to 3 bits in encoding: for example.z/ 1 1 C.k/ (5.5 Alternative structures for the predictor In this section we omit the quantizer and we analyze alternative structures for the predictor.168) .12.z/ D S. we refer to Section 1. MA and ARMA).9) with transfer function H .
k/ D w. the synthesis ﬁlter has a FIR all zero transfer function H .168) implies an AR(N ) model for the input.k i/ (5.k/ f q .k/.z/ D 1 C q X i D1 bi z i (5. : : : .k/ D s.173) bi z ci z i (5. Differential coding (DPCM) 399 1.5.171) Incidentally we note that an approximate LMS adaptation of the coefﬁcients fbi g is given by bi .k//2 ] is C.174) i . The coding scheme that makes use of an allpole predictor is called linear predictive coding (LPC) and the prediction error f f . 2.k i/ C bi f .169) with P × N . Allzero predictor For an MA input model. The PN n and optimum predictor that minimizes E[. We observe that the two cases model in a simpliﬁed manner the input to an allpole ﬁlter whose output is unvoiced (case 1) or voiced (case 2). Then (5. In this case (5.s.170) Correspondingly from (5.k/g is a periodic sequence of impulses. fw.k/.z/ D O nD1 an z it yields f . In this case s .k/ C ¼b f q .k/g is a periodic signal of period PN n and it yields f .5.z/ D 1 q X i D1 p X i D1 p X i D1 q X i D1 ci s.k/.168) implies that also fs. the prediction signal is given by s .k/ D w. q) input model.172) Polezero predictor The general case refers to an ARMA( p. w. for sq .k/g is called LPC residual.k C 1/ D bi . we have 1C H .k/ D O q X i D1 bi f .k/ D O Correspondingly.k/ D A C1 X nD 1 Žk nP (5. q (5.k i/ (5.z/ D nD1 an z equal to a periodic sequence of impulses. P.124). The optimum predictor is still C.k/g is white noise.k i/ i D 1. fw.k/ s .
175) is the longterm estimate.k/ D s.k/ Then.721 standard at 32 kbit/s (see Table 5.62. The prediction error f . in (5.k/g as shown in the scheme of Figure 5.k/ s .178) is the shortterm estimate. In this case it is convenient to use the estimate O s . and P is the pitch period expressed in number of samples.172) for the coefﬁcients fbi g. For the LMS adaptation of the coefﬁcients in (5.k i/ (5.k/ C ss . Pitch predictor An alternative structure exploits the quasiperiodic behavior. the 5 bit quantizer is adapted by the Jayant scheme.k/ D s` . In (5.176) (5. The equations (5.175). we refer to (5.k i/ (5.k/ D þs.174) are illustrated in Figure 5.179) is related to the input fs.176) þ is the pitch gain.62. This conﬁguration was adopted by the ADPCM G. ss .63. of period P.177) ci f ` . Digital representation of waveforms s(k) + f(k) f(k) sq(k)=s(k) + ^ s(k) b b + ^ s(k) + c + c sq(k)=s(k) + (a) (b) Figure 5.173).k/g be the corresponding prediction error f ` . Polezero predictor: (a) analysis.166) for the coefﬁcients fci g and to (5. .k P/ (5.400 Chapter 5.k O P/ (5.173) and (5.k/ D s. of voiced spurts of speech.k/ D O N X i D1 þs.16).k/ O N X i D1 ci f ` .k/ O O where s` . Let f f ` .k/ D f ` . that uses an LMS adaptive predictor with 2 poles and 4 zeros. (b) synthesis.
even if not optimum. the autocorrelation sequence of f f ` . using only a onebit quantizer. . 10 For the adopted notation see Footnote 3 on page 441.154).180) It follows:10 P D arg max ². initially conducted by Atal [5].183) From the estimate of the autocorrelation sequence of fs.k/g at lag n.5. once P and þ are determined. Determination of the shortterm predictor through minimization of the cost function: min E[ f 2 .s. Cascade of a longterm predictor with an allpole predictor.k/g.k/] c (5.179) is in fact an allpole type with a very high order N . Differential coding (DPCM) 401 s(k) +  fl(k) +  f (k) ^ (k) ss cl longterm predictor cl (z)= βz P ^ sl (k) c shortterm predictor c(z)= Σ ci zi i=1 N Figure 5. The encoder and decoder schemes are given in Figure 5. The subdivision into two terms.k/g is easily computed. Then the coefﬁcients fci g of the longterm predictor can be obtained by solving a system of equations similar to (5. has the advantage of allowing a very simple computation of the various parameters.5.k P//2 ] (5. for speech signals sampled at 8 kHz the whole predictor in (5.P þs.k/g.k/ þ.64: they form the adaptive predictive coding (APC) scheme which differs from the DPCM for the inclusion of the longterm predictor. and þ D ². Experimental measurements. APC Because P is usually in the range from 40 to 120 samples. Computation of longterm predictor through minimization of the cost function min E[ f ` . where and ρ depend on the autocorrelation coefﬁcients of f f ` .k/2 ] D E[.63.n/ n6D0 (5.P/ (5.n/ represents the correlation coefﬁcient of fs. have demonstrated that adapting the various coefﬁcients by the feedforward method every 5 ms we get high quality speech reproduction with an overall bit rate of only 10 kbit/s.182) 2.181) where ². 1.
We note that plots in Figures 5. as shown in Figure 5. Adaptive predictive coding scheme.66. . whereas these lines are attenuated in the plot of Figure 5. due to the periodic behavior of the corresponding signals in the time domain. the improvement given by the longterm predictor to lowering the LPC residual is shown in Figure 5.402 Chapter 5.64. For voiced speech. Digital representation of waveforms Figure 5. Without the longterm predictor the LPC residual presents a peak at every pitch period P.66b exhibit some spectral lines.65c. The frequencydomain representation of the three signals in Figure 5.65.66a and 5.66c. these peaks are removed by the longterm predictor.65 is given in Figure 5.
66. (b) LPC residual. (a) Voiced speech. Figure 5. (b) LPC residual.5.65.65: (a) voiced speech. (c) LPC residual with longterm predictor. (c) LPC residual with longterm predictor. Differential coding (DPCM) 403 Figure 5. DFT of signals of Figure 5. .5.
180).6 5.t/. Digital representation of waveforms Other longterm predictors with 2 or 3 coefﬁcients have been proposed: although they are more effective. Observation 5.187) and power spectral density Px .1 Delta modulation Oversampling and quantization error For an input signal s. There are also numerous methods that are more robust and effective than (5.kTc / be the sampled version of s. it is important to have a signaltonoise ratio that is constant in the frequency domain: this yields the socalled spectral shaping of the error.247).404 Chapter 5. the determination of their parameters is much more complicate than the approach (5.nTc / D rs n F0 C1 X `D 1 T0 F0 1 2B (5. WSS random process with bandwidth B. are normally sent: for example reﬂection coefﬁcients (PARCOR).6.188) Equation (5.6 From the standpoint of perception. Let x.184) (5. Two improvements with respect to the basic APC scheme are outlined in the following observations [3]. obtained by ﬁltering the residual error so that it is reduced at frequencies where the signal has low energy and enhanced at frequencies where the signal has high energy.188) is obtained from (5.t/.186) (5. f / D Ps Â f ` F0 T0 Ã (5. thus partly assimilating the longterm predictor in the overall predictor.90) and the deﬁnition (1. : : : .185) (5. area functions or line spectrum pairs (LSP). To avoid this very laborious computation. . Observation 5. allpole predictors have been proposed with more than 50 coefﬁcients. parameters associated with the prediction coefﬁcients fci g i D 1.n/ D rs .7 In APC. 5. N .187) using (1. let the sampling period be Tc D where T0 D and F0 is the oversampling factor.181) to determine the pitch period P. with autocorrelation Â Ã T0 rx . t 2 <.k/ D s.
k/ D x.k/ (5. In particular. Effects of oversampling for two values of the oversampling factor F0 .k/g become more correlated. from (5.k/g presents images that are more spaced apart from each other.k/ be the quantized signal and eq .k/g. .67a we note that by increasing F0 the samples fx.6.187) and from Figure 5.67 shows the effect of oversampling on fx.188) we have that the spectrum of fx.k/ C eq . Delta modulation 405 Figure 5. let xq .5.k/ the corresponding quantization error xq .68. from (5. Let us now quantize fx.k/g. With reference to Figure 5.189) Figure 5. moreover.67.
the effective signaltonoise ratio is given by 3q. by increasing F0 the PSD of eq decreases in amplitude. f / C Peq . f / D Meq T0 F0 (5. In conclusion. .68. Rb D b 1 F0 Db Tc T0 (5.193) (5. For feq .o . at the output we will have yq .189). we have Peq . the encoder bit rate.k/ D eq Ł g.189). depending only on the number of quantizer levels (see (5.191) From (5.k/ where eq. by ﬁltering fxq .192) and.190) Consequently. f / D Meq Then Meq. However.194) T0 f rect F0 2B (5. from (5. but at the expense of quadrupling the bit rate. F0 D 4 improves 3q by 6 dB.k/ D x. Digital representation of waveforms Figure 5.o .195) where 3q D Mx =Meq is to the signaltonoise ratio after the quantizer.406 Chapter 5. f / (5.192). we have Pxq .k/g with an ideal lowpass ﬁlter g having bandwidth B and unit gain. for the noncorrelation assumption between x and eq .k/ has PSD given by Peq.98)).196) increases proportionally to F0 : for example. General scheme.k/g white with statistical power Meq .44) and (5.k/ C eq.o . f / D Px . with reference to (5. Therefore oversampling by large factors is used only locally before PCM or in compact disk (CD) applications to simplify the analog interpolation ﬁlter at the receiver. under the assumption that the quantization noise is white. oversampling improves performance by a factor F0 .o D Meq F0 (5.o D 3q F0 (5. Moreover.
thus simplifying the overall system. which is illustrated in Figure 5.197) and a quantizer with only two levels (b D 1). . Figure 5.5.69.6. For a predictor with only one coefﬁcient c1 . that onebit code words eliminate the need for framing of the code words at the transmitter and at the receiver. moreover.2 Linear delta modulation (LDM) Linear delta modulation is a DPCM scheme with oversampled input signal. 1 × 2B Tc (5.69. is called a linear delta modulator (LDM).6. the coding scheme.198) The high value of F0 implies a high predictability of the input sequence: therefore a predictor with a few coefﬁcients gives a high prediction gain and the quantizer can be reduced to the simplest case of b D 1. We note. The following relations hold. LDM coding scheme. Then the encoder bit rate is equal to the sampling rate. Rb D 1 (bit/s) Tc (5. Delta modulation 407 5.
i Ä k. which performs the function of the ﬁlter g. 1.k/ s . The accumulated value is proportional to sq .k/ ½ 0 1 . where the DAC is often a simple holder.200) (5. the integrator has a bandwidth B equal to that of the input signal.201) and (5.199) (5.k/ O 1 . Note that the decoder consists simply of an accumulator followed by a DAC. Letting b.k/ The system is based on three parameters (1=Tc .k/ D ( 1 1 c. Mixed analogdigital implementation. For example the choice of c1 D 1 considerably simpliﬁes (5.o .sq Ł g/. this implementation involves the accumulation of fb. Analog implementation.k/ D c1 sq .408 Chapter 5.203). also for the LDM we need to choose a small 1 to obtain low granular noise. However. As for the DPCM scheme.203).k/ D s.k/ D s.k/ D 0 1/ C f q . Digital representation of waveforms Encoder: f . An alternative to the previous scheme.70c.k/ D 1/ if f .202) (5.k/ D .k/ < 0 (5.k/ (5. that are appropriately selected.69 for c1 D 1 is given in Figure 5.k/ D 1 c. Choice of system parameters With reference to Figure 5.70a.203) (5. is obtained O by placing a DAC after the accumulator and carrying out the comparison in the analog domain. typically we set c1 Ä 1 so that random transmission errors do not propagate indeﬁnitely in the reconstruction signal (5.71.c.k/ D s .k/ O Decoder: f q .k/ D 0/ if f . which requires carrying out the operation f .i/g.k/ ( f q . LDM implementation Digital implementation.70a and Figure 5.k/ D sgn[ f . to eliminate the outofband noise.k/ C f q .70b.c. that become simple accumulator expressions.k/. At the receiver.k/ in the digital domain.k sq. and of the gain 1 of the quantizer step size. as illustrated in Figure 5.201) h i O s .k C 1/ D c1 s . by an updown counter (ACC). and c1 ). we will now establish a relation between the various parameters of an LDM.204) sq . In many applications it is convenient to implement the analog accumulator by an integrator: thus we obtain the implementation of Figure 5.k/]. to get instead a small slope overload . An implementation of the scheme of Figure 5.
Delta modulation 409 Figure 5. Graphic representation of LDM. .5.6.71. granular noise s(k) s(k1) slope overload distortion ∆ sq (k) ^ s(k) } fq(k) t 1 +1 +1 { 0 Tc +1 +1 +1 +1 +1 1 +1 1 b(k) Figure 5.70. LDM implementations.
In other words. as a consequence we have that LDM requires a very high oversampling factor F0 to give satisfactory performance.410 Chapter 5.205) ½ max þ þ t dt Tc and the reconstruction signal fsq .72. An alternative is represented by an adaptive scheme for the step size 1. which requires a sampling rate of the order of 200 kHz.k 1/ (5.d=dt/s. In speech applications with a bandwidth of about 3 kHz. . the Jayant algorithm uses the following relation 1. In particular. if we reduce 1 to decrease the granular noise we must also reduce Tc to limit the slope overload distortion. to have a 3q.2F0 / where ²x . the only possibility in the LDM is to reduce Tc .1// ln.k/g can follow very rapid changes of the input signal.1 ²x . Digital representation of waveforms distortion a large 1 is needed. 5.1/ D rs .o of 9 dB: 3 dB are due to ﬁltering of the outofband noise and 6 dB to the reduction of the granular noise. as 1 can be halved.Tc /=rs .72.t/j.t/þ (5.o of approximately 35 dB we need F0 ½ 33.3 Adaptive delta modulation (ADM) To reduce both granular noise and slope overload distortion. as shown in Figure 5.6.0/.206) 1opt D 2Ms .k/ D p1.207) Figure 5. The optimum value of 1 is given approximately by p (5. for a given value of maxt j. ADM coding scheme: (a) encoder. so that þ þ þ þd 1 þ s. We note that doubling the sampling rate we obtain an increment in 3q. (b) decoder.
k/ sq.k/b. (5.k 1/ D c.6.k/ (5.k/ D Þ1.k if c.k 1/ C D1 otherwise sq(k) s(k) s(k1) slope overload distortion (5. .k/ (5. Experiments on speech signals show that by doubling the sampling rate in the ADM we get an improvement of 10 dB in 3q .k/ D s.k 1/ f q .k/ D sgn f .216) sq . Encoder: f .210) (5. In some applications ADM encoding is preferred to PCM because of its simple implementation.o .k/ 6D c.k 1/ C f q .k/ s .k/ D p1. in spite of the higher bit rate.k/ D c.209) b. A graphic representation of ADM encoding is shown in Figure 5.k/ D c1 sq .k/ D c.217) Typical values for po and pg are given by 1:25 < po < 2 and po pg Ä 1.k/ O (5.k 1/ (5.k 1/ C D2 if c.k/ C f q .k/ D 1.k 1/ (slope overload ) 1/ (granular noise) (5.k 2/ (slope overload ) 1.k/ D sq Ł g.5.211) (5. The following relations between the signals of Figure 5.k/ D p1.73.k/ also depends on c.73.215) (5. Graphic representation of ADM.213) Continuously variable slope delta modulation (CVSDM) An alternative to the adaptation (5.k/ D 1.k/.k/b.k/ s .218) granular noise ^ s(k) 0 Tc t b(k) +1 +1 +1 1 +1 1 +1 1 +1 1 Figure 5.72 hold.k/] O s Decoder: 1.207) is given by the equation ( Þ1.k/ 1. Delta modulation 411 where pD ( po > 1 pg < 1 if c.208) We note that in this scheme 1.k C 1/ D c1 [O .212) (5.214) f q .
74. Disregarding the ﬁltering effect on noise. we obtain the PCM output signal sampled at the minimum rate 1=T0 . Linear PCM encoder via LDM. using a decimator ﬁlter. the oversampling factor F0 must be at most equal to 2b and. 1 − F0 Ä 2b For example. as illustrated in Figure 5. that is a lowpass ﬁlter followed by a downsampler. that brings a gain of 10 log10 F0 dB. It is sufﬁcient to accumulate fb.k/g with an accuracy of b bits. especially for Þ D 1. The main difﬁculty of this scheme is the sensitivity of fsq .222) Figure 5. the decoder is equivalent to the cascade of two leaky integrators.1 .6. this means 1=Tc D F0 =T0 ' 2 MHz.k/g to transmission errors.k O 1/ C c2 sq .z/ D 1 /.74. 5. in general. we observe that to generate a PCM signal fc PC M .2.z/ D 1 c1 z 1 c2 z 2 The function H . . in some cases secondorder predictors are used. for b D 8 and 1=T0 D 8 kHz.k 2/ (5.220) H . at the expense of worse performance.z/ can be split into the product of two ﬁrstorder terms. and D1 and D2 are suitable positive parameters with D2 × D1 . ADM with secondorder predictors With the aim of improving system performance.2 to generate a linear PCM encoded signal.1 p1 z p2 z 1 / If p1 and p2 are real with 0 < p1 . The value Þ controls the speed of adaptation.412 Chapter 5. p2 Ä 1. that employs the LDM implementation of Figure 5.k/g to obtain a PCM representation of the input s. Digital representation of waveforms where 0 < Þ Ä 1.k/g. where s . (5.t/.70c.4 PCM encoder via LDM We consider an alternative scheme to the three implementations of Section 5. The problem now is to determine the slope overload condition from the sequence fc.k/ D c1 sq . choosing Þ < 1 mitigates the effects of transmission errors.221) H . 1 (5.219) The transfer function of the synthesis ﬁlter is given by 1 (5.
. similarly to the scheme of Figure 5. given the general coding scheme of Figure 5. 6DM coding scheme. DPCM.76.7 Coding by modeling In the coding schemes investigated so far.75. which simpliﬁes the LDM integrator: therefore the decoder becomes a simple lowpass ﬁlter.74. and their variations. recalling that the spectrum of quantization noise in PCM and LDM is ﬂat.75a and the simpliﬁed scheme of Figure 5.6. the objective is to reproduce at the decoder a waveform that is as close as possible to the input signal. We now take a different approach and. Thus we get the general scheme of Figure 5. Moreover.5 Sigma delta modulation ( DM) With reference to the scheme of Figure 5.70c. Note that.75b: we note that the DACs are simple holders of binary signals. with respect to the scheme of Figure 5. it is sufﬁcient to employ a 6DM followed by a digital decimator ﬁlter with input the binary signal fb.5. to enhance the low frequency components of speech signals we can insert a preemphasis integrator before the LDM encoder. 5. It is interesting to observe the simplicity of the 6DM implementation. A differentiator then has to be inserted at the LDM decoder. the accumulator has been removed. Therefore it can be removed to a large extent by a simple lowpass ﬁlter. Coding by modeling 413 Figure 5. PCM.74. 6DM presents the advantage that the noise is colored and for the most part is found outside the passband of the desired signal.k/g.7. 5. One of the most frequent applications of 6DM is in linear PCM encoders where.
In (5. Multipulse LP (MELP). For further study we refer the reader to [3.223) is given by p (5. Three examples follow. The excitation signal is selected by a collection of possible waveforms stored in a table. The excitation signal consists of a train of undersampled impulses.76. is illustrated in Figure 5. We will now analyze in detail some coding schemes. At the encoder. The difference among the various coding schemes consists in the form of excitation. known also as the LPC vocoder.224) ¦ D JN where J N is the statistical power of the prediction error. is modeled by an AR. as we will assume that f f . however. .N / linear system H .414 Chapter 5.z/ D 1 ¦ N X i D1 (5.223) ci z i with input f f . The excitation signal consists of a certain number of impulses with suitable amplitude and lag.2. In particular. Codebook excited linear prediction (CELP). for example speech.223) the coefﬁcients fci g and ¦ are obtained by the prediction algorithms of Section 2. the standard deviation in (5.77. derived from the residual signal. 6]. and the LCP parameters are extracted. Vocoder or LPC The general scheme for the conventional LPC. Basic scheme of coding by modeling. the source fs.k/g.k/g.k/g has unit statistical power. the signal is classiﬁed as voiced or unvoiced. Regular pulse excited (RPE). together with the pitch period P for the voiced case. Digital representation of waveforms Figure 5.
78b. The choice of the best of the three subsequences (actually four are used in practice) is made by the analysisbysynthesis (ABS) approach. operating with blocks of 160 samples. The excitation is then ﬁltered by the AR ﬁlter to generate the reconstruction signal. RPE coding The RPE coding scheme. as shown in Figure 5. At the decoder. Vocoder or LPC scheme. .78a.5. In an early LPC scheme for military radio applications (LPC10). is a particular case of residual excited LP (RELP) coding in which the excitation is obtained by downsampling the prediction residual error by a factor of 3. where all the excitations are tried: the best is that which produces the output “closest” to the original signal.64.77. Coding by modeling 415 Figure 5. illustrated in Figure 5. the input signal sampled at 8 kHz is segmented into blocks of 180 samples.10) includes also a longterm predictor as the one of Figure 5. with a latency lower than 80 ms. The standard ETSI for GSM (06. For the analysis of the LPC parameters the covariance method is used. The bit rate is 13 kbit/s. the excitation sequence is then quantized using a 3bit adaptive nonuniform quantizer. the prediction residual error is not transmitted.7. for the voiced case a train of impulses with period P is produced. whereas for the unvoiced case white noise is produced. overall 54 bits per block are needed with a bit rate of 2400 bit/s.
. CELP coding As shown in Figure 5. trying to minimize the output of the weighting ﬁlter. RPE coding scheme.416 Chapter 5. also in this case the predictor includes a long term component. the excitations belong to a codebook obtained in a “random” way. The choice of the excitation (index of the codebook) is made by the ABS approach.79.8) of the residual signal. or by vector quantization (see Section 5.78. Digital representation of waveforms Figure 5.
80 exempliﬁes the encoder and decoder functions of a VQ scheme. : : : . using multidimensional signals opens the way to many techniques and applications that are not found in the scalar case [7. Block diagram of a vector quantizer. Q L g. The encoder computes the distortion associated with the representation of the input vector s by each Figure 5. 8].8 Vector quantization (VQ) Vector quantization (VQ) is introduced as a natural extension of the scalar quantization (SQ) concept.k/.80. a reproduction vector sq D Q[s] chosen from a ﬁnite set of L elements (code vectors). . called codebook. CELP coding scheme.8. Vector quantization (VQ) 417 Figure 5. so that a given distortion measure d. : : : . However. Q[s]/ is minimized. 5. The basic concept is that of associating with an input vector s D [s1 .s.5. The analysis procedure is less complex than that of the CELP scheme. with the difference that the minimization procedure is used to determine the position and amplitude of a speciﬁc number of impulses. generic sample of a vector random process s. Figure 5. s N ]T . Multipulse coding It is similar to CELP coding. A D fQ1 .79.
Qi /. : : : . dimension of the code vectors.k/ D [c1 . Digital representation of waveforms reproduction vector of A and decides for the vector Qi of the codebook A that minimizes it. s. i D 1. : : : . speech signal. associated with an observation window of a signal. We note that the information transmitted over the digital channel identiﬁes the code vector Qi : therefore it depends only on the codebook size L and not on N . ž Codebook A D fQi g. as indicated by (5. 8i 6D j (5.81. ž Quantizer rate Rq D log2 L (bit/vector) or (bit/symbol) (5. s. a vector quantizer is characterized by ž Source or input vector s D [s1 . are called Voronoi regions. s2 / : Q[s1 ] D Q[s2 ]g (5.s1 .1 Characterization of VQ Considering the general case of complexvalued signals.229) .227) every subset R` contains all input vectors associated by the quantization rule with the code vector Q` . c N . the decoder associates the vector Qi to the index i received. : : : . : : : .418 Chapter 5. : : : ..k N Tc /. Parameters determining VQ performance We deﬁne the following parameters.k N T .k/. L.227) The sets fR` g. s2 . Q` / ` (5.228) In other words.k/] 5.s. ž Quantization rule (minimum distortion) Q:C N N is a code vector.k/ D [s. R L g of the source space C N . L. ! A with Qi D Q[s] if i D arg min d.226) which associates input vector pairs having the same reproduction vector. or the N LPC coefﬁcients. L (5. : : : . An example of input vector s is obtained by considering N samples at a time of a N C 1/Tc /]T .s. identiﬁes a partition R D fR1 . whose elements are the sets R` D fs 2 C N : Q[s] D Q` g ` D 1. ` D 1. s. It can be easily demonstrated that the sets fR` g are nonoverlapping and cover the entire space C N : L [ `D1 R` D C N Ri \ R j D .1 (Partition of the source space) The equivalence relation Q D f.8. : : : . s N ]T 2 C N . where Qi 2 C ž Distortion measure d. An example of partition for N D 2 and L D 4 is illustrated in Figure 5.225) Deﬁnition 5.
ž Rate per dimension RI D ž Rate in bit/s Rb D RI log2 L D (bit/s) Tc N Tc (5.234) (5.235) E[d.s.233) If the input process s.232) ð A ! <C (5.s.231) Rq log2 L D (bit/sample) N N (5.s.k/g. A/ D E[d.k/ is stationary and the probability density function ps . ž Distortion d. Vector quantization (VQ) 419 R4 R1 1 0 Q4 C2 Q 11 00 1 Q 1 0 3 Q 11 00 2 R2 R3 Figure 5.230) where Tc denotes the time interval between two consecutive samples of a vector. In other words.236) . Partition of the source space C 2 in four subsets or Voronoi regions. Qi / j s 2 R` ]P[s 2 R` ] N d` P[s 2 R` ] D (5. in (5. Q[s]/] D L X `D1 L X `D1 (5.81.a/ is known. d:C N (5. Qi / The distortion is a nonnegative scalar function of a vector variable.5.R.8. we can compute the mean distortion as D.231) N Tc is the sampling period of the vector sequence fs.
.2 .n j ¹ #¼=¹ (5.N / S.N / M.1 .s Qi / H Rs . deﬁned in (1. A/ D lim K 1 X d.a/ da Z ps .s.N / e DV Q (5. Itakura–Saito distortion: d.n /Ł [Rs ]n. 2. and we use the average distortion (5.420 Chapter 5. Q i. Q` / ps .R.a/ da R` (5. Deﬁning Qi D [Q i.a. the metric deﬁned by (5.239) Q i. Qi / D jjs Qi jj¹ D jsn nD1 (5. Qi / D jjs 2.241) .k/.346). 1. : : : . n. Qi / D . m D 1.m . Digital representation of waveforms where N d` D Z R` d.sm Q i. Q[s. Q i.s.238) as an estimate of the expectation (5.238) In practice we always assume that the process fsg is stationary and ergodic.n j2 (5.240) The most common version is the squared distortion:11 d.242) where Rs is the autocorrelation matrix of the vector sŁ .s Qi / D N N XX nD1 mD1 Qi jj2 D 2 N X nD1 jsn Q i.243) 11 Although the same symbol is used.N ]T we give below two measures of distortion of particular interest.m .sn Q i. with elements [Rs ]n.241) is the square of the Euclidean distance (1. for a given rate R I we ﬁnd (see [9] and references therein) e DS Q D F.s.s. Comparison between VQ and scalar quantization e Deﬁning the mean distortion per dimension as D D D=N .234).237) If the source is also ergodic we obtain D.38). N .k/. Distortion as the `¹ norm to the ¼th power: " N X ¼ d. : : : .k/]/ K kD1 K !1 (5.m / (5.
247) jj ps .s.N / D jj ps . Vector quantization (VQ) 421 where ž F. 12 Extending (5. The expression of M.a/jj¼=¹ D Q ÄZ ÐÐÐ Z ps .2 Optimum quantization Our objective is to design a vector quantizer. Ri can be “shaped” very closely to a sphere. For N ! 1.1/ D ž M. choosing the code vectors of the codebook A and the partitioning R so that the mean distortion given by (5. : : : .R. deﬁned as12 S.1/ D 2³ e=12 D 1:4233 D 1:53 dB.a/jj1 Q (5. Qi / D min d.234) the solution is given by Ri D fs : d. Observing (5.N / D jj ps .8. 5.a/. L (5.N / is the memory gain.244) . Q L g ﬁxed.s. Ri contains all the points s “nearest” to Qi .a1 .5.240) to the continuous case we obtain jj ps .a/jj N =.N / increases as the correlation increases. deﬁned as M.N / does not depend on the variance of the random variables of s. Two necessary conditions arise. otherwise M.N C2/ (5.a/ is the probability density function of the input s considered with uncorQ related components.a/.248) As illustrated in Figure 5.a/jj N =. but only on the norm order N =.N C 2/ and shape ps .N / is the space ﬁlling gain.a/ and ps . which differ for the correlation Q among the various vector components.a/jj1=3 Q jj ps .a/.N C2/ Q jj ps .N C2/ Q (5. The asymptotic value for N ! 1 equals F. Rule A (Optimum partition).a/jj N =.a/ is the probability density function of the input s. : : : . Assuming the codebook A D fQ1 . obviously if the components of s are statistically independent.234) is minimized.245) where ps . : : : .a/jj1=3 Q jj ps . In the scalar case the partition regions must necessarily be intervals.N / D 1. S. ž S. a N / da1 : : : da N Q¹ ½¼=¹ (5. we want to ﬁnd the optimum partition R that minimizes D.246) where ps .N / depends on the two functions ps . A/. we Q obtain S.82.N / is the gain related to the shape of ps . Q` /g Q` 2A i D 1.8. we have M. In an N dimensional space.
Digital representation of waveforms Figure 5.251) ps . Example of partition for K D 2 and N D 8.249) In other words Qi coincides with the centroid of the region Ri .a/ da Generalized Lloyd algorithm The generalized Lloyd algorithm. choosing the squared distortion (5.241).422 Chapter 5.249) yields Z R Qi D Z i Ri a ps . Rule B (Optimum codebook). (5. given in Figure 5.83. As a particular case. 1. .237) becomes Z jjs Qi jj2 ps .a/ da 2 1 Ri N Z di D (5. The iteration index is denoted by j. we want to ﬁnd the optimum codebook A.a/ da Ri and (5. We choose an initial codebook A0 and a termination criterion based on a relative error ž between two successive iterations.s. Initialization. Assuming the partition is R given. Q j / j s 2 Ri ] Q j 2 Ri (5.a/ da (5. By minimizing (5. Qi / j s 2 Ri ] D min E[d.s. generates a sequence of suboptimum quantizers speciﬁed by fRi g and fQi g using the previous two rules.236) we obtain the solution Qi : E[d.82.250) ps .
2. If Dj 1 Dj Dj <ž (5. 4.83. otherwise we update the value of j.5. 1 ].8.236). 3. Using the rule A we determine the optimum partition R[A j ] using the codebook A j .252) we stop the procedure. We evaluate the distortion associated with the choice of A j and R[A j ] using (5. Vector quantization (VQ) 423 Figure 5. We go back to step 2. . Using rule B we evaluate the optimum codebook associated to the partition R[A j 6. Generalized Lloyd algorithm for designing a vector quantizer. 5.
Q` /g Q` 2A i D 1. whereas in the scalar quantization the partition of the real axis is completely speciﬁed by a set of .k/. as well as trying different initial codebooks. and some of the locally optimum codes may give rather poor performance. is no longer sufﬁcient.8. the calculation of the centroid is difﬁcult for the VQ. ž Also in the particular case (5.251).k/]/ K kD1 m D 1.k/. ž The algorithm assumes that ps . it is often advantageous to provide a good codebook to the algorithm to start with. In the scalar quantization it is possible.254) and the two rules to minimize D become: Rule A Ri D fs. : : : .s.s.3. as we also need to characterize the statistical dependence among the elements of the source vector.234) with (5. in many applications. The algorithm is clearly a generalization of the Lloyd algorithm given in Section 5. Digital representation of waveforms The solution found is at least locally optimum. : : : .a/.m/g The average distortion is now given by DD K 1 X d. and for the multidimensional case to ﬁnd the optimum solution becomes very hard.3 LBG algorithm An alternative approach led Linde. Buzo. For VQ with a large number of dimensions.a/ is known. The sequence used to design the VQ is called training sequence (TS) and is composed of K vectors fs. L (5. given that the number of locally optimum codes can be rather large. K (5. to develop an appropriate model of ps . Qi / D min d. because it requires evaluating a multiple integral on the region Ri . In fact. 5.424 Chapter 5.k/g of the TS nearest to Qi . Gaussian or Laplacian.k/ : d.k/. ž The computation of the input space partition is much harder for the VQ.2: the only difference is that the vector version begins with a codebook (alphabet) rather than with an initial partition of the input space. the identiﬁcation of the distribution type. Q[s.s.253) (5.L 1/ points. . but this becomes a more difﬁcult problem with the increase of the number of dimensions N : in fact. However.255) that is Ri is given by all the elements fs. nevertheless. for example. the implementation of this algorithm is difﬁcult for the following reasons.238) for K sufﬁciently large. and Gray [10] to consider some very long realizations of the input signal and to substitute (5. the partition becomes also harder to describe geometrically. in the twodimensional case the partition is speciﬁed by a set of straight lines.
Qi / D and (5.241) we have d.258) that is Qi coincides with the arithmetic mean of the TS vectors that are inside Ri .257) (5. ž The partition is determined without requiring the computation of expectations over C N. we arrive at the LBG algorithm. the ﬁrst L vectors of the TS can be used. Choice of the initial codebook With respect to the choice of the initial codebook. Using the structure of the Lloyd algorithm with the new cost function (5.k/ Q i. for the squared distortion (5.k/.k/ m i s. Iteratively the splitting procedure and optimization is repeated until the desired number of elements for the codebook is obtained.255) and (5. ž The computation of Qi in (5. ž It does not require any stationarity assumption. Q j / m i s. it is worthwhile pointing out some aspects of this new algorithm. . it is necessary to use L vectors that are sufﬁciently spaced in time from each other.8. each optimum code vector is changed to obtain two code vectors and the LBG algorithm is used for L D 4. At convergence. using the LBG algorithm. ž It converges to a minimum. Slightly changing the components of this code vector (splitting procedure). at this point.256) is still burdensome.5. A more effective alternative is that of taking as initial value the centroid of the TS and start with a codebook with a number of elements L D 1.k/2R i (5.k/.n j2 (5.254) and the two new rules (5. However. Before discussing the details.s.256) where m i is the number of elements of the TS that are inside Ri .256) simply becomes Rule B Qi D 1 X s.k/2R i N X nD1 jsn . Vector quantization (VQ) 425 Rule B Qi D arg min Q j 2C N 1 X d. we determine the optimum VQ for L D 2. if the data are highly correlated.256). and generally depends on the choice of the TS.s. which is not guaranteed to be a global minimum. we derive two code vectors and an initial alphabet with L D 2. however.
261) (5. Possible causes of this phenomenon are: ž TS too short: the training sequence must be sufﬁciently long.85. Selection of the training sequence A rather important problem associated with the use of a TS is that of empty cells. whereas its operations are depicted in Figure 5. i D 1. The splitting procedure generates 2L N dimensional vectors yielding the new codebook A jC1 D fA j g [ fAC g j where A j D fQi AC D fQi j Typically ε is the zero vector.260) (5.263) (5. : : : . we can limit the problem through the following splitting procedure. : : : . Let Ri be a region that contains m i < m min elements.84. L (5. in addition to the obvious solution of modifying this choice. Digital representation of waveforms Let A j D fQ1 . L . It is in fact possible that some regions Ri contain few or no elements of the TS: in this case the code vectors associated with these regions contribute little or nothing at all to the reduction of the total distortion. so that every region Ri contains at least 3040 vectors. (5. Q L g be the codebook at iteration jth. whose block diagram is shown in Figure 5.264) r Ms Ð1 N (5. ε D0 and 1 εC D 10 so that jjε C jj2 Ä 0:01 Ms 2 where Ms is the power of the TS. we obtain the LBG algorithm. ž poor choice of the initial alphabet: in this case. : : : .426 Chapter 5. We eliminate the code vector Qi from the codebook and apply the splitting procedure limited only to the region that gives .262) ε g εC g i D 1.259) Description of the LBG algorithm with splitting procedure Choosing ž > 0 (typically ž D 10 3 ) and an initial alphabet given by the splitting procedure applied to the average of the TS.
the largest contribution to the distortion. for a sequence different from the TS (outside TS) the distortion is 13 These rules were derived in the VQ of LPC vectors. Vector quantization (VQ) 427 Figure 5. then we compute the new partition and proceed in the usual way. and L is the number of code vectors. ž If K =L Ä 30.84. In this situation. the distortion computed for vectors of the TS is very small. LBG algorithm with splitting procedure. an appreciable difference between the distortion calculated with the TS and that calculated with a new sequence may exist. We give some practical rules. In the latter situation. the extreme case is obtained by setting K D L.13 that can be useful in the design of a vector quantizer. ž If K =L Ä 600. it may in fact happen that.5. . taken from [3] for LPC applications. for a very short TS. there is a possibility of empty regions. hence D D 0. where we recall K is the number of vectors of the TS.8. They can be considered valid in the case of strongly correlated vectors.
. computed over windows of duration equal to 20 ms of a speech signal sampled at 8 kHz. only if K is large enough. As a matter of fact. Taking L D 256 we have a rate Rb D 8 bit/20 ms equal to 400 bit/s. the 14 This situation is similar to that obtained by the LS method (see Section 3.2). Digital representation of waveforms Figure 5. Operations of the LBG algorithm with splitting procedure.428 Chapter 5.86.14 Finally we ﬁnd that the LBG algorithm.86. We consider. As illustrated in Figure 5. Figure 5. even though very simple. for example. requires numerous computations. Values of the distortion as a function of the number of vectors K in the inside and outside training sequences. as vector source the LPC coefﬁcients with N D 10.85. in general very high. does the TS adequately represent the input process and no substantial difference appears between the distortion measured with vectors inside or outside TS [10].
1.5. N L E V . N L E V D log2 L). Divide the training sequence into subsequences relative to every node of level n (n D 2. thus determining a full search.4 Variants of VQ Tree search VQ A random VQ.8. collect all vectors that are associated with the same code vector. which 5.87. ž a large computational complexity to evaluate the L distances for encoding. Figure 5. To determine the code vectors at different nodes. A variant of VQ that requires a lower computational complexity. determined according to the LBG algorithm. then we proceed along the branch whose node has a representative vector “closest” to s. 155000 vectors for the TS. at the expense of a larger memory. : : : . whereas in the memoryless VQ case the comparison of the input vector s must occur with all the elements of the codebook. Vector quantization (VQ) 429 LBG algorithm requires a minimum K D 600 Ð 256 roughly corresponds to three minutes of speech. is the tree search VQ.8. for a binary tree the procedure consists of the following steps. requires: ž a large memory to store the codebook. in the tree search VQ we proceed by levels: ﬁrst. 3. As illustrated in Figure 5. Apply the LGB algorithm to every subsequence to calculate the codebook of level n.87. Comparison between full search and tree search. the codebook contains 2 code vectors. 2. in other words. . 3. Calculate the optimum quantizer for the ﬁrst level by the LBG algorithm. we compare s with Q A1 and Q A2 .
Successively. Digital representation of waveforms Table 5.15 for a given value of Rq (bit/vector) in the cases of full search and tree search. Ð/ 1024 20 Number of vectors to memorize P Rq 2 Rq ' 2 Rq C1 i i D1 2 Number of vectors to memorize 1024 2046 As an example.89.Ð. Memory: L locations.Ð. A third stage could be used to quantize the error of the second stage and so on. Computation of d. Computations of d.Ð.Ð. The idea consists in dividing the encoding procedure into successive stages.430 Chapter 5. where the ﬁrst stage performs quantization with a codebook with a reduced number of elements. Let Rq D log2 L be the rate in bit/vector for both systems and assume that all the code vectors have the same dimension N D N1 D N2 . the second stage performs quantization of the error vector e D s Q[s]: the quantized error gives a more accurate representation of the input vector. ž Twostage: Rq D log2 L 1 C log2 L 2 . the memory requirements and the number of computations of d. as illustrated in Figure 5. however.88. ž Onestage: Rq D log2 L. the computational complexity of the encoding scheme for a tree search is considerably reduced. Memory: L 1 C L 2 locations.Ð. Ð/ are shown in Table 5. Ð/ for encoding: L. Although the performance is slightly lower. We compare the complexity of a onestage scheme with that of a twostage scheme. The advantage of a multistage approach in terms of cost of implementation is evident. Product code VQ The input vector is split into subvectors that are quantized independently. Computations of d. Ð/ full search tree search for Rq D 10 (bit/vector) full search tree search 2 Rq 2Rq Computation of d. illustrated in Figure 5. Multistage VQ The multistage VQ technique presents the advantage of reducing both the encoder computational complexity and the memory required. . it has lower performance than a onestage VQ. Ð/ for encoding: L 1 C L 2 . hence L 1 L 2 D L.15 Comparison between full search and tree search.
5. Product code VQ. Figure 5. .8.88.89. Vector quantization (VQ) 431 Figure 5. Multistage (twostage) VQ.
9 Other coding techniques We brieﬂy discuss two other coding techniques along with the perceptive aspects related to the hearing apparatus. For further details we refer the reader to [3. 6]. It presents the disadvantage that it does not consider the correlation that may exist between the various subvectors. assuming L D L 1 L 2 and N D N1 C N2 . or b) the input vector has too large a dimension to be encoded directly.g.432 Chapter 5.90. With reference to Figure 5. Block diagram of the ATC. Figure 5. we note that the rate per dimension for the VQ is given by log2 L 1 log2 L 2 log2 L D C N N N whereas for the product code VQ it is given by Rq D Rq D log2 L 1 log2 N2 C N1 N2 (5. prediction gain and LPC coefﬁcients.265) (5. e..89. where the quantization of the subvector n depends also on the quantization of previous subvectors.266) 5. Digital representation of waveforms This technique is useful if a) there are input vector components that can be encoded separately because of their different effects. that could bring about a greater coding efﬁciency. . A more general approach is the sequential search product code. [7].
discretevalued source signal can be encoded with a lower average bit rate by means of entropy coding [4]. i. a discretetime. but it operates in the time domain by using a ﬁlter bank (see Figure 5.10. using for example the DFT or the DCT (see Sections 3. We cite the Lempel–Ziv algorithm as one of the most common source coding algorithms.10 Source coding We brieﬂy mention an important topic.90 and includes a quantizer that adapts to the different inputs fS. Subband coding (SBC) The SBC exploits the same principle as the ATC. 5.91). which is used to “compress” digital information messages. to highly probable input patterns are assigned shorter code words and vice versa.4). In fact. . Source coding 433 Figure 5.5. Block diagram of the SBC.e. The basic scheme is illustrated in Figure 5.m/g. which assigns code words of variable lengths to possible input patterns.5. Adaptive transform coding (ATC) The ATC takes advantage of the nonuniform energy distribution of a signal in some transformed domain.5. namely source coding.3 and 3.91.
4 kbit/s CELP at 4.16 a partial list of the various standards to code audio and speech signals [11]. It is interesting to observe the various standards that adopt CELP coding.3 and 6. It is also interesting to compare the various standards to code video signals given in Table 5.16 for speech and audio.721 G. Standard 1 2 3 4 5 G.1).4 kbit/s SBC+ADPCM for wide band speech at 64.711 G.6 kbit/s version SBC at 192 kbit/s per audio channel (stereo) [generally 32 ł 448 kbit/s total] SBC at 128 kbit/s per audio channel [generally 32 ł 384 kbit/s total] SBC+MDCT+Huffman coding at 96 kbit/s per audio channel [generally 32 ł 320 kbit/s total] SBC+MDCT coding at 64 kbit/s per audio channel 6 7 8 9 10 G. 56 and 48 kbit/s . Digital representation of waveforms 5.723+G. 5+3 or 6+2 VSELP at 7.434 Chapter 5. Table 5.726 G. there is also a 5. listed in Table 5. AAC . The ﬁrst nine are for narrowband speech applications (see Table 5. 24 and 16 kbit/s (“embedded” means that a code also includes those of lower rate) LDCELP at 16 kbit/s (LD stands for low delay) CSACELP at 8 kbit/s CSACELP at 8 kbit/s with reduced complexity MPCMLQ at 5. for example.727 Description PCM at 64 kbit/s ADPCM at 32 kbit/s ADPCM at 24 and 40 kbit/s G.18 with those of Table 5.722 11 12 13 14 15 16 17 18 IS54 (TIA) FS1015 (LPC10E) FS1016 GSMFR MPEG1.17: we notice that most of them are for cellular radio applications. Bit allocation in the two bands is dynamic. Layer II MPEG1. 0 ł 4 kHz and 4 ł 8 kHz is used. Layer III MPEG2.729 G.728 G.721 embedded ADPCM at 40. 32. Layer I MPEG1.1 G. in each band there is a G.95 kbit/s (VSELP stands for vector sum excited linear prediction) LPC at 2.729 Annex A G. A SBC scheme having two bands.723 G.8 kbit/s RPELTP at 13 kbit/s (LTP stands for longterm prediction).721 encoder.11 Speech and audio standards We conclude this chapter by giving in Table 5.723.16 Main standards for audio and speech coding.
Application Target bit rate ISDN Video Telephone 64ł128 kbit/s ISDN Video Conferencing 128 kbit/s MPEG1 CDRom Video 1. [2] IEEE Signal Processing Magazine.18 Bit rates for video standards.17 Main standards based on CELP.4 kbit/s Table 5. W.5 Mbit/s MPEG2 TV (Broadcast Quality) 6 Mbit/s HDTV (Broadcast Quality) 24 Mbit/s TV (Studio Quality. (DoD) U. Uncompressed) 216 Mbit/s HDTV (Studio Quality. vol. Rabiner and R.2 kbit/s. Schafer.6 kbit/s. Englewood Cliffs. Jayant and P. encoding of speech and data 7. Codiﬁca numerica del segnale audio. Digital processing of speech signals.5.8 kbit/s 2. NJ: PrenticeHall. Sereno and P. coding of audio in multimedia systems 16 kbit/s 8 kbit/s. 14.S. (DoD) Abbreviations G. 1978.3 kbit/s. Body ITU ITU ITU TIA TIA TIA/ETSI ETSI U. Uncompressed) 1 Gbit/s Bibliography [1] L. enhanced full rate for GSM 5.4. half rate for GSM 4. 2. 1997. [4] N.723 G. 1996.729 IS54 IS95 US1 GSMHR FS 1016 MELP Bit rate 5.S. S. Noll.6 kbit/s. Reiss Romoli. 1984.728 G. Englewood Cliffs.95 kbit/s. Compressed) 140 Mbit/s TV (Studio Quality.8 and 9. NJ: PrenticeHall. Digital coding of waveforms. 4.27 and 6. R. full rate for North America cellular systems based on DAMPS 1. . Bibliography 435 Table 5. L’Aquila: Scuola Superiore G. coding for North America cellular systems based on CDMA 12. Compressed) 34 Mbit/s HDTV (Studio Quality.2. Sept. [3] D. Valocchi.
1991. Atal and J. vol. Sept. S. S. and A. on Communications. R. Remde.436 Chapter 5. M. ICASSP. Atal. Sept. Lookabaugh and R. Boston. Linde. “High–resolution quantization theory and the vector quantizer advantage”. MA: Kluwer Academic Publishers. 1. Gray. Gray. IEEE Trans. vol. “A new model of LPC excitation for producing naturalsounding speech at low bit rates”. 1984. 4–29. Vector quantization and signal compression. pp. Advances in speech coding. Jan. M. IEEE ASSP Magazine. 1997. D. Buzo. “Vector quantization”. Gray. Gray. Apr. 1980. Digital representation of waveforms [5] B. A. pp. “An algorithm for vector quantizer design”. 35. MA: Kluwer Academic Publishers. [9] T. 1989. vol. 614–617. pp. [10] Y. on Information Theory. [7] A. vol. Boston. V. Cuperman. 1020– 1033. . [11] IEEE Communication Magazine. 1982. [6] B. IEEE Trans. 84–95. M. Gersho and R. and R. in Proc. Gersho. 1992. eds. pp. [8] R. M. 28. 35.
and present a survey of the main modulation techniques. PSK.Chapter 6 Modulation theory The term modulation indicates the process of translating the information generated by a source into a signal that is suitable for transmission over a physical channel.2. in general. it may consist for example of a twistedpair cable. and sends it over the channel. or a combination of them.v. In the case of digital transmission. Using the vector representation of signals discussed in Section 1. based on the received signal. a radio link. the waveform is corrupted by realvalued additive white Gaussian noise w having zero mean and spectral density N0 =2. as discussed in Chapter 4. an optical ﬁber. and biorthogonal. With reference to the system illustrated in Figure 6. in this chapter we will introduce the optimum receiver.1 is modeled as a discrete r. The variable a0 in Figure 6. QAM. a coaxial cable. . M ½ 2 waveforms (Mary modulation). n D 1. the transmitter generates randomly one of M realvalued waveforms sn . an infrared link. referring to the detection theory. or.t/.1 Theory of optimum detection We consider ﬁrst the transmission of an isolated pulse. and noise. The task of the receiver is to detect which signal was transmitted. in any case the channel modiﬁes the transmitted waveform by introducing for example distortion. M. only in Section 6.12 we will give some simple results for a channel affected by ﬂat fading. The transmission rates achievable by the various modulation methods over a speciﬁc channel for a given target error probability are then compared with the Shannon bound. The mapping of a digital sequence to a signal is called digital modulation.. 6. We postpone the study of other effects to the next chapters. and the device that performs the mapping is called digital modulator. A modulator may employ a set of M D 2 waveforms to generate a signal (binary modulation). interference. the information is represented by a sequence of binary data (bits) generated by the source.g. and a comparison of the various methods is given in terms of spectral efﬁciency and required transmission bandwidth. which indicates the maximum rate that can be achieved for reliable transmission. The transmission medium determines the channel characteristics.1. e. orthogonal. The performance of each modulationdemodulation method is evaluated with reference to the bit error probability. : : : . or by a digital encoder of analog signals (see Chapter 5). In this chapter we assume that the channel introduces only additive white Gaussian noise (AWGN). PAM.
I (6. The theory O exposed in this section can be immediately extended to the case of complexvalued signals. : : : .t/ C w. we express the noise as w. M i D 1.t/. or symbol. : : : . with values in f1.2) (6. Mg. Let f i . is also called the system constellation.t/ The receiver.t/. : : : . signal is given by r. M (6.4) Recall that the set fsm g. Assuming that the waveform with index m is transmitted. and represents the index.t/g. 2. sm I ]T (6.t/g.t/ C w.t/ where w . or observed.t/ n D 1.7) .438 Chapter 6. : : : .t/ C w? . based on r.t/ dt m D 1. i D 1.5) wi i .t/ where smi D hsm .t/ Ł i . the received. I . sm . must decide which among the M hypotheses Hn : r.t/ (6. The basis of I functions may be incomplete for the representation of the noise signal w. be a complete basis for the M signals fsm . : : : . t 2 <. : : : . Model of the transmission system.t/ dt (6.t/ D and where wi D hw. let sm be the vector representation of sm .t/ D sn . that is a0 D m.2.6) w. : : : . In any case. m D 1.1) is the most probable.t/ D w .3) D sm .1. m D 1.t/ D sm . Z ii C1 1 ! sm D [sm1 . Modulation theory Figure 6. : : : . M. We represent timecontinuous signals using the vector notation introduced in Section 1. Z ii D 1 C1 I X i D1 (6. M. and correspondingly must select the detected value a0 [1].t/ Ł i . of the transmitted signal.
t/] Ł i .12) Ł i . : : : . i D 1. I .t2 /] ZZ Z Ž. fwi g.6.t/g. Mean: Z E[wi ] D as w has zero mean.9) Statistics of the random variables {wi } 1. it can be ignored because it is irrelevant for the detection.t/ w .t1 t2 / Ł i .t1 / j .t/g.t/ (6.t2 / dt1 dt2 E[w. are jointly Gaussian random variables. whereas w? is the error due to this representation.13) . As fwi g. then they are statistically independent with equal variance given by ¦ I2 D N0 2 (6.t/g.t1 / j . we can say that w? lies outside of the desired signal space and. : : : . i D 1. as they are linear transformations of a Gaussian process (see (6.t2 /] D and E[wi wŁ ] D j D ZZ E[w. Theory of optimum detection 439 In other words. I (6. i D 1. I . Hence the components fwi g are uncorrelated. Since w? . w I ]T D w (6. 2.8) is orthogonal to f i .t/ is white noise.t/ dt D0 i D 1.t/ j .t1 /w Ł . : : : . : : : . as we will state later using the theorem of irrelevance.t/ D w. we have E[w.t2 / dt1 dt2 (6. w is the component of w that lies in the space spanned by the basis f i .t1 /w Ł . : : : . are jointly Gaussian uncorrelated random variables with zero mean. The vector representation of the component of the noise signal that lies in the span of f i .t1 2 t2 / (6.1.11) N0 2 Ł i .t/g is given by w D [w1 .7)). I . t 2 <.10) N0 Ž. Correlation: as w. j D 1. I . I because of the orthogonality of the basis f i . : : : . 3. : : : .t/ dt N0 D 2 D N0 Ži 2 j i. i D 1. and hence also to w .
Recalling the total probability theorem. A particular case is represented by transformations that allow reconstruction of a signal using the basis identiﬁed by the desired signal.15) From the above results. under the hypothesis that waveform n is transmitted. : : : . Then we adopt the following decision rule: O choose Hn (and a0 D n) if r 2 Rn (6. Hn : r D sn C w n D 1. in general the notion of sufﬁcient statistics applies to any signal. . considering the basis feš j2³ f t .2). that is the Fourier transform of the noisy signal ﬁltered by an ideal ﬁlter with passband B.ρ j n/ dρ D M X nD1 D M XZ nD1 Rn 1 Given a desired signal corrupted by noise. that allows the optimum detection of the desired signal. M (6.380) for complexvalued signals. t 2 <.377) for realvalued signals. : : : . or a priori probability.17) The choice of M regions is made so that the probability of a correct decision is maximum. no information is lost in considering a set of sufﬁcient statistics instead of the received signal.14) the components of the vector r are called sufﬁcient statistics 1 to decide among the M hypotheses. For example. ii (6. we are able to reconstruct the noisy signal within the passband of the desired signal. for n 6D m). In other words.ρ j n/ D r exp jjρ sn jj2 (6. Let pn D P[a0 D n] be the transmission probability of the waveform n. we would get the same results using the formulation (1.18) pn P[r 2 Rn j a0 D n] pn prja0 . the probability density function of r. the probability of correct decision is given by P[C] D P[a0 D a0 ] O D M X nD1 P[a0 D n j a0 D n]P[a0 D n] O (6. f 2 Bg to represent a realvalued signal with passband B in the presence of additive noise. r I ]T with ri D hr.16) !I N0 N0 2³ 2 Decision criterion We subdivide the space < I of the received signal r into M nonoverlapping regions Rn SM ( nD1 Rn D < I and Rn \ Rm D . therefore the noisy signal ﬁltered by a ﬁlter with passband B is a sufﬁcient statistic. is given by:2 Ã Â 1 1 ρ 2 <I prja0 . Therefore we get the formulation equivalent to (6. 2 Here we use the formulation (1. or sequence of samples.440 Chapter 6. Modulation theory Sufﬁcient statistics Deﬁning r D [r1 .
ρ j n/ D pr . 2.24) has the largest probability of having been transmitted.ρ j n/.2. Theory of optimum detection 441 We deﬁne the indicator function of the set Rn as ² ρ 2 Rn In D 1 0 elsewhere Then (6. being the M regions nonoverlapping. n/ exist. n/ n (6. If two or more values of n that maximize f . −1 ] R2 D . a random choice is made to determine m. is achieved if for each value of ρ we select among M terms the term that yields the maximum value of pn prja0 . m D arg max f .² j n/. : : : . n D 1.ρ j n/ n (6.25) 3 arg means argument.x.−2 .22) P[a0 D n j r D ρ] pn (6.ρ/ the decision criterion becomes a0 D arg max P[a0 D n j r D ρ] O n a0 D m O if m D arg max pn prja0 . If we indicate with −1 .x.c/ denotes the phase of c. arg.18) becomes Z P[C] D M X < I nD1 (6. 1. M. be given as shown in Figure 6.−1 .23) (6. given that we observe ρ. for each value of ρ only one of the terms is different from zero. −3 ] (6. . For a complex number c.24) In other words.6. n/. it is easy to verify that R1 D . and hence of the integral. We give a simple example of application of the MAP criterion for I D 1 and M D 3.−3 .20) The integrand function consists of M terms but. Thus we have the following decision criterion. the signal detected by (6.x.19) In pn prja0 .3 Maximum a posteriori probability (MAP) criterion: ρ 2 Rm Using the Bayes’ rule prja0 . Therefore the maximum value of the integrand function for each value of ρ. −2 . Let the function pn pr ja0 . n/ is maximum for a given x.1.ρ j n/ dρ (6.21) denotes the value of m that coincides with the value of n for which the function f . and −3 the intersection points of the various functions as illustrated in Figure 6. −2 ] [ .2. are the a posteriori probabilities. n D 1. The probabilities P[a0 D n j r D ρ ]. C1/ R3 D . 3.x. for a function f .
An example is given in Figure 6. s j . 2. M.22) becomes ρ 2 Rm a0 D m O if m D arg max prja0 . The decision region associated with each vector sn is given by the intersection of two halfplanes as illustrated in Figure 6. 8n. we introduce a theorem that formalizes the distinction previously mentioned between relevant and irrelevant components of the received signal.2. Illustration of the MAP criterion.ρ j n/ n (6.ρ j n/ prja0 . Theorem of irrelevance With regard to the decision process.ρ/ D prja0 . The procedure then is repeated for every pair of vectors. In some texts the ML criterion is formulated via the deﬁnition of the likelihood ratios: Ln .442 Chapter 6. which is a monotonic function.27) In this case the ML criterion becomes a0 D m O if m D arg max Ln . we draw the straight line of points that are equidistant from si and s j : this straight line deﬁnes the boundary between Ri and R j . we obtain a0 D arg min jjρ O n sn jj2 (6.3. Considering a pair of vectors si .30) Hence the ML criterion coincides with the minimum distance criterion: “decide for the signal vector sm .16) we get Â a0 D arg max exp O n 1 jjρ N0 sn jj 2 Ã (6.2. Maximum likelihood (ML) criterion.29) Taking the logarithm.ρ j 1/ n D 1. n D 1.3 for the three signals of Example 1. are easily determined. : : : . the decision regions fRn g. observing (6. the criterion (6.26) The ML criterion leads to choosing that value of n for which the conditional probability that r D ρ is observed given a0 D n is maximum. Modulation theory p1 pra ( ρ 1) 0 p3 pra ( ρ 3) 0 p2 pra ( ρ 2) 0 τ 1 τ 2 τ 3 ρ Figure 6.ρ/ n (6. If the signals are equally likely a priori. . i.28) From (6. : : : . Moreover. which is closest to the received signal vector ρ”. pn D 1=M.26). M (6.e.2 on page 10.
Example 6.4. r2 ].32) (6.ρ 2 / We illustrate the theorem by the following examples.5) has two components w1 D w w2 D w? (6.ρ 2 j ρ 1 . Theory of optimum detection 443 φ2 A T 2 R3 s1 0 R1 A T 2 φ1 s3 s 2 R2 Figure 6.31) then the optimum receiver can disregard the component r2 and base its decision only on the component r1 . Then.34) .1.a0 .22) leads to the following result.a0 .r2 ja0 . Corollary 6. ρ 2 j n/ D pr2 jr1 .1 A sufﬁcient condition to disregard r2 is that pr2 jr1 .ρ 2 j ρ 1 . n/ does not depend on the particular value n assumed by a0 .ρ 1 .32) into (6. Let us assume that the signal vector r can be split into two parts.35) (6.1 The system (6.a0 . where the noise (6.6. n/ D pr2 .2.ρ 2 j ρ 1 . ρ 2 j n/ which.a0 . n/ pr1 ja0 .r2 ja0 . can be rewritten as pr1 .ρ 1 .1. under the hypothesis a0 D n.2) is represented using a larger basis. prja0 .ρ 2 j ρ 1 / (6. that is if pr2 jr1 . recalling the deﬁnition of conditional probability. Theorem 6. r D [r1 .ρ j n/ D pr1 . n/ D pr2 jr1 .2.3.33) (6.1 If pr2 jr1 . as illustrated in Figure 6.ρ 1 j n/ Substitution of (6.ρ 2 j ρ 1 . Construction of decision regions for the constellation of the Example 1.
the component r2 D w? can be disregarded by the optimum receiver.444 Chapter 6. by Corollary 6.1: the vector r2 is irrelevant. n/ D pr2 .5. Example 6.ρ 2 / (6.ρ 2 j ρ 1 . that is independent of the particular sn transmitted.ρ 2 j ρ 1 .1.ρ 2 that depends explicitly on n.a0 . from r2 D w2 C w1 and w1 D r1 sn . if r1 is known.5. n/ D pw2 . . Therefore we have pr2 jr1 . Example 6.ρ 2 j ρ 1 . then r2 depends only on the noise w2 .1.36) hence.1. Modulation theory s1 s 2 w1 r = w +s 1 1 n r =w 2 2 w2 s M Figure 6. however.1. Then pr2 jr1 . n/ D pw2 .3 As in the previous example. As r2 D r1 C w2 . Example 6.ρ 2 ρ 1 C sn / (6.ρ 2 ρ1/ (6.4. r2 cannot be disregarded by the optimum receiver: in fact. the noise vectors w1 and w2 in Figure 6. the noise vectors w1 and w2 are statistically independent. we get pr2 jr1 .1.6 are statistically independent.37) does not depend on n: therefore (6.33) is veriﬁed and r2 is irrelevant.2 In the system shown in Figure 6. s1 s s 2 w1 / D pw2 . Example 6.38) w1 w 2 r = w + r = w + w + sn 2 2 1 2 1 M r = w +s 1 1 n Figure 6. Under this condition.a0 .a0 . We note that the received signal vector r2 coincides with the noise vector w2 that is statistically independent of w1 and sn .2: the vector r2 is irrelevant.
8. if the noise vectors w1 and w2 are statistically independent. : : : .1.1 is a particular case of Example 6.a0 . where a correlation demodulator is substituted by a matched ﬁlter with impulse response iŁ . Theory of optimum detection 445 w1 s1 s s 2 w 2 r = w2 + w 1 2 r = w +s 1 1 n M Figure 6.a0 .4: the vector r2 is irrelevant. An equivalent implementation is based on the equivalence illustrated in Figure 6.− / iŁ . yields yi .t0 / D ri . ..4. As illustrated in Figure 6.ρ 2 / (6.7. M (6.34) the signal r2 can be neglected by the optimum receiver. and We note that the ﬁlter on branch i has impulse response given by rect.9. We note that Example 6. assuming that fsm . Implementations of the maximum likelihood criterion We give now two implementations of the ML criterion. observing (6.6.1.39) that does not depend on n.1.t/ D r.4 In Figure 6. there are two fundamental blocks: the ﬁrst determines the I components of the vector r.t/g and f i .ρ 2 j ρ 1 .t0 t/.3: the vector r2 is relevant w1 s1 s 2 w 2 r =w 2 2 r = r +r 2 1 r = w +s 1 1 n s M Figure 6. Example 6.ρ 2 j ρ 1 sn .t yields the output Z t yi .14). and the second computes the M distances Dn D jjr sn jj2 n D 1. Example 6. n/ D pw2 jw1 . t0 /.40) t0 =2/=t0 /.0.1.− / d− t t0 (6. Implementation type 1.1. Example 6.1. from (6.6.t/g have ﬁnite duration in the interval .7. it is: pr2 jr1 . In fact. n/ D pw2 .41) which sampled at t D t0 .
sn i] ½ (6. Implementation type 1 of the ML criterion. Modulation theory Figure 6. .) d τ tt 0 (a) r(t) ri t0 r(t) φ * (t t) 0 i ri (b) Figure 6.446 Chapter 6.10. Implementation type 2. from jjρ the ML criterion becomes Ä a0 D arg max Rehρ. φ *(t) i t0 t (.45) The implementation of (6.t/j2 dt (6. (a) Correlation demodulator and equivalent (b) matched ﬁlter (MF) demodulator.44) Ä D arg max Re n ÄZ C1 1 Ł ². sn i O n sn jj2 D jjρjj2 C jjsn jj2 2Re[hρ.t/ dt where E n is the energy of sn . Z En D 1 C1 jsn .43) ½ En 2 ½ (6. whereas the equivalent criterion (6.42) jjsn jj2 2 (6.43) is also given in Figure 6.44) is illustrated in Figure 6.8.39). Using (1.t/sn .8.9.
whose vector representation is illustrated in Figure 6.10.46) where P[C] is given by (6. Implementation type 2 of the ML criterion. in the applications we have I < M. assuming ² real.] t0 r(t) s* (t0 t) 2 Re[.18). Theory of optimum detection 447 t0 s* (t0 t) 1 Re[. For convenience we choose 2 parallel to the line joining s1 and s2 .47) We examine the case of two signals. Independently of the basis system. from (1. Error probability In general.49) .1.]  EM 2 UM Figure 6.11.t/j2 dt i D 1.48) d 2 D E 1 C E 2 2² E 1 E 2 where Z Ei D 1 C1 jsi . hence implementation type 1 is more convenient.41) the squared distance between the two signals is given by p (6. 2 (6. the error probability of the system is deﬁned as O Pe D P[E] D P[a0 6D a0 ] D 1 P[C] (6. we express the error probability as M X nD1 Pe D pn P[r 2 Rn j a0 D n] = (6. Using the total probability theorem. Typically.]  E1 2 E2 2 U1  U2 ^ a0=arg max Un n ^ a0 t0 s* (t0 t) M Re[.6.
448 Chapter 6. equation (6. As for all projections on an orthonormal basis. and Ł s1 .51) We assume that s2 is transmitted.52) Ã ½ Â d d DQ 2 2¦ I (6.t/ dt ²D p E1 E2 In the case of two equally likely signals.56) . it is Ã ½ Â Ä d d (6.47) becomes 1 Z C1 (6. the noise component w2 is Gaussian with zero mean and variance N0 2 Then the conditional error probability is given by Ä P[r 2 R2 j a0 D 2] D P w2 < = ¦ I2 D where Q.t/s2 . Given the received signal vector r as in Figure 6.51).11.a/ D Z (6. which means a0 D 2. Binary constellation and corresponding decision regions. we obtain Pe D Q Â d 2¦ I Ã (6.53) b2 1 (6.55) DQ P[r 2 R1 j a0 D 1] D P w2 > = 2 2¦ I C1 From (6. Likewise. we get a decision error if the noise w D r s2 has a projection on the line joining s1 and s2 that is smaller than d=2. Modulation theory Figure 6.11.50) Pe D 1 fP[r 2 R1 j a0 D 1] C P[r 2 R2 j a0 D 2]g = = 2 (6.A.54) p e 2 db 2³ a is the Gaussian complementary distribution function whose values are reported in Appendix 6.
58) Pe D Q @ 2N0 If E 1 D E 2 D E s .59) 6.t/ For E s D RT 0 s2 . Substitution of (6. we have Z ²D C1 1 Ł s1 .t/ D s.t/j2 dt.t/ dt Es D 1 (6. Theory of optimum detection 449 Observation 6.63) .1 ²/ N0 ! (6.57) We will now derive an alternative expression for Pe as a function of the modulator parameters.62) (6.61) The basis has only one element. we get s Pe D Q E s . For this reason it is useful to deﬁne the following signaltonoise ratio at the decision point Â D d 2¦ I Ã2 (6.t/s2 .t/ D p Es The vector representation of Figure 6.t/ . T / (6. Antipodal signals (ρ = −1) The signal set is composed of two antipodal signals: s1 .60) js.12 follows.t/ D s.13.48) and (6.t/ deﬁned in .t/ s. I D 1.1.6.1 The error probability depends only on the ratio between the distance of the signals of the constellation at the decision point and the standard deviation per dimension of the noise.56) yields 0s 1 p E 1 C E 2 2² E 1 E 2 A (6. (6.1 Examples of binary signalling We let M D 2 and E 1 D E 2 D E s . where p Á p Á s2 D s1 D Es Es The implementation type 1 of the optimum ML receiver is depicted in Figure 6.1.52) in (6. with s.0.
71).59) becomes s Pe D Q 2E s N0 ! (6.t/ D A cos.64) A modulation technique with antipodal signals is binary phase shift keying (2PSK or BPSK).t/ 0<t <T 0<t <T (6.66) Orthogonal signals (ρ = 0) We consider the two signals s1 .68) A basis is composed of the signals themselves r 2 s1 . then Es D E1 D E2 D A2 T 2 and ² ' 0 (6. ML receiver for binary antipodal signalling. deﬁned by (6. ^ =1 a r<0 0 . if f 0 D k=2T .t/ D A cos.13.2³ f 0 t/ s2 .t/ are “windowed” versions of sinusoidal signals. is shown in Figure 6. (6.2³ f 0 t/ D 1 .E s s1 0 d=2 Es Es φ Figure 6.t/ D 2 .t/ D A sin.t/ D p T Es and s2 . In this case s1 .60). where s.12.2³ f 0 t C '0 / s2 .t/ D p Es We note that 1 .65) (6. r(t) φ* (Tt) T r r>0 .69) r 2 sin .14. ^ =2 a 0 ^ a0 Figure 6. Vector representation of antipodal signals. . k integer.t/. or else f 0 × 1=T .2³ f 0 t/ 0<t <T (6.t/ cos.2³ f 0 t/ T 0<t <T (6. Modulation theory s2 .67) Observing (1.t/ D A cos.2³ f 0 t C '0 C ³ / D s1 .450 Chapter 6.t/ 0<t <T (6.70) and 2 . As ² D 1.
0 < t < T. As the two vectors s1 and s2 are p p orthogonal. As ² D 0.71) N0 .15. Plot of s.16.1. their distance is 2Es . for f0 D 2=T and '0 D ³=2.6.59) becomes s ! Es Pe D Q (6. The vector representation is given in Figure 6. Theory of optimum detection 451 A A/2 s(t) 0 −A/2 −A 0 t T Figure 6.14.2³ f0 t C '0 /. (6. The optimum ML receiver is depicted in Figure 6.t/ D A cos. φ2 Es s2 2E s s1 0 Es φ1 Figure 6. This distance is reduced by a factor of 2 as compared to the case of antipodal signals with the same value of Es . Vector representation of orthogonal signals.15.
t/ D A cos.17. Binary FSK We consider the two signals of Figure 6. Figure 6. ^ =1 a 1 2 0 ^ a0 T s* (Tt) 2 U 2 U <U . From the curves of probability of error versus E s =N0 plotted in Figure 6.2³. ^ =2 a 1 2 0 Figure 6.72) where '0 is an arbitrary phase. Modulation theory T s* (Tt) r(t) 1 U 1 U >U .2³.452 Chapter 6. ML receiver for binary orthogonal signalling. f 0 C f d /t C '0 / 0<t <T (6. f 0 f d /t C '0 / s2 .t/ D A cos.17.16.18. given by s1 . We examine in detail another binary orthogonal signalling scheme. . we note that for a given Pe we have a loss of 3 dB in E s =N0 for the orthogonal signalling scheme as compared to the antipodal scheme. Error probability as a function of Es =N0 for binary antipodal and orthogonal signalling.
we have s Pe D Q E s .2h/.1. we have hD Therefore ² D sinc.74) As FSK is a binary modulation. We have two “windowed” sinusoidal functions.18.2T / (6. and f d is the frequency deviation.76) .2T /. Binary FSK signals with f0 D 2=T.4 f d T / (6. Also in this case. one with frequency f 0 f d and the other with frequency f 0 C f d .t/s2 . fd D 2 fd T 1=. Theory of optimum detection 453 A s (t) 0 1 −A 0 t T A s (t) 0 2 −A 0 t T Figure 6.t/ dt A2 T 2 D sinc. if f 0 š f d × 1=T .1 ²/ N0 ! (6.75) Introducing the modulation index h as the ratio between the frequency deviation f d and the Nyquist frequency of the transmission system equal to 1=.6. f 0 is called the carrier frequency.73) s1 . fd D 0:3=T and '0 D 0. it holds Es D E1 D E2 ' and Z ²D C1 1 A2 T 2 (6.
and a loss of 2.sn . MSK also requires that the phase of the modulated signal be continuous (see Section 18. with squared distances Z C1 2 2 jsn . that is for f d D 1=.t/g.5 4 Figure 6. Modulation theory 1 0.4 0 0.77) and 2 2 dmin D min dnm n.8 0.3 dB with respect to antipodal signalling.4 ρ 0.79) 4 In fact. sn / < d. : : : ) that yield ² D 0. n D 1. 2.5). 6. 1:5.78) Upper bound We assume sm is transmitted.t/ sm . however. we get the minimum p value of ². be M equally likely signals.5 2 h 2.5 3 3.4 There are other values of h (1. sm / D dnm D 1 (6.19.7 dB in E s =N0 as compared to the case ² D 0. with the consequent requirement of larger channel bandwidth.19. 1:22E s =N0 /. ²min D 0:22.1. From the plot of ² as a function of h illustrated in Figure 6. .5 1 1.74) we have ² D 0 for h D 1=2. : : : . with a gain of 0. for h D 0:715 and Pe D Q. sm /g (6. An error event that leads to choosing sn is expressed as Enm D fr : d.6 0.r.2 0 −0.454 Chapter 6.2 Bounds on the probability of error Let fsn .t/j2 dt d . they imply larger f d .m (6. From (6. Correlation coefﬁcient ² as a function of the modulation index h.2 −0.r.4T /: in this case we speak of minimum shift keying (MSK). M.
M. For example.2¦ I //.81) On the other hand. applying the union bound.86) M 2¦ I In other words.85) Q Pe ½ M nD1 2¦ I We get a looser bound by introducing Nmin .1. let dmin. and Pe ½ Q. Limiting the equation in (6. from (6. then Q.53).87) d12 D 2 2 2 p Then dmin D A T =2 and Nmin D 3. an error occurs if sn is chosen. Limiting the evaluation of the error probability to error events that are associated to the nearest signals. as sm is transmitted.n be the distance of sn from the nearest signal.n D dmin . and a looser bound than (6.dnm =. : : : . we have Â Ã dmin Nmin Pe ½ Q (6.6.2¦ I // Ä Q.47).82) P[E j sm ] D P nD1.n (6.M 1/Q (6.80) In general. the error probability in the case of equally likely signals is Pe D P[E j sm ] 1 M (6.n6Dm nD1.t/g whose distance is dmin from the nearest signal: 2 Ä Nmin Ä M. we have Nmin D M.dmin =.dmin =. the number of signals fsn . n 6D m. . an error event E is reduced to considering only a signal at minimum distance. if there is one. n D 1.n6Dm Then an upper bound on Pe is given by Ã Â M M 1 X X dnm Q Pe Ä M mD1 nD1. Therefore.85) to such signals.83) is given by Â Ã dmin Pe Ä . the probability of the event Enm is given by Ã Â dnm P[Enm ] D Q 2¦ I M X mD1 (6. In the particular case of dmin. we obtain the following lower bound Ã Â M 1 X dmin. we have " # M M [ X Enm Ä P[Enm ] (6.2¦ I //. Theory of optimum detection 455 Using (6. for the constellation of Figure 6.83) Since dmin Ä dnm .n6Dm 2¦ I (6.84) 2¦ I Lower bound Given sn .3 with M D 3 we have r T Ap Ap d23 D T d13 D A T (6. given sm .
Note that the system of Figure 6. Some bits may be inserted in the message fb` g to generate an encoded message fcm g.20. However. In the model of Figure 6. which assume values in an Mary alphabet and are generated at instants kT .20.456 Chapter 6. it is associated with a waveform.t kT / (6. . Modulation theory Figure 6. Chapter 12. according to the scheme of Figure 6.8. which is assumed to introduce additive white Gaussian noise w.89) which is sent over the transmission channel.t/ be the signal at the output of the transmission channel.5 The value of the generic symbol ak is then modulated. In that case the notion of binary channel cannot be referred to the transmission of the sequence fcm g. for example.t/ with PSD N0 =2. where the information message consists of a sequence of information bits fb` g generated at instants `Tb . The bits of fcm g are then mapped to the symbols fak g. see. assuming that the transmitted waveforms do not give rise to intersymbol interference (ISI) at the demodulator 5 Note that there are systems in which encoder and modulator are jointly considered. Simpliﬁed model of a transmission system. with reference to Figure 6. Let sCh .2 Simpliﬁed model of a transmission system and deﬁnition of binary channel With reference to Figure 6.t Therefore the transmitter generates the signal s.1: ak ! sak . 6.t/ D C1 X kD 1 kT / (6.20. the transmission of a waveform is repeated every symbol period T .88) sak . according to rules that will be investigated in Chapters 11 and 12. that is.1 has been investigated assuming that an isolated waveform is transmitted. we discuss now some aspects of a communication system.
measured by the bit error probability .90) In the case of a binary symmetric channel (BSC).1 The transformation that maps cm into cm is called a binary channel.21. all signalling schemes that employ pulses with ﬁnite duration in the interval . The overall objective of the transmission system is to reproduce the sequence of information bits fb` g with a high degree of reliability. it is assumed that P[cm 6D cm j cm D Q 0] D P[cm 6D cm j cm D 1]. the symbol a0 transmitted at instant t D 0.3. m 2 . 1g Q (6. m N . We note that the aim of the channel encoder is to introduce redundancy in the sequence fcm g.6. For example.0. T / do not give rise to ISI. the bits fc` g are obtained by inverse bit mapping from the detected Q O message fak g.3. Simpliﬁed model of a transmission system and deﬁnition of binary channel 457 output6 we can still study the system assuming that an isolated symbol is transmitted. the following relation holds: Q Q P[cm 1 6D cm 1 . and by the bit error probability Q Pbit D PBC D P[cm 6D cm ] cm .91) In a memoryless binary symmetric channel the probability distribution of fcm g is obtained Q from that of fcm g and PBC according to the statistical model shown in Figure 6. sak . At the receiver. We say that the BSC is memoryless if. 6 Absence of ISI in this context means that the optimum reception of the waveform transmitted at instant kT . Memoryless binary symmetric channel. this is a particular case of the Nyquist criterion for the absence of ISI that will be discussed in Section 7.21. However. cm 2 f0. is not inﬂuenced by the presence of the waveforms associated with symbols transmitted at other instants. The information bits fb` g are then recovered by a decoding process. . cm N 6D cm N ] Q Q Q D P[cm 1 6D cm 1 ] P[cm 2 6D cm 2 ] : : : P[cm N 6D cm N ] Q (6. : : : .92) Figure 6. for every choice of N Q distinct instants m 1 . which is exploited by the decoder to detect and/or correct errors introduced by the binary channel. which is the transmission rate of the bits of the sequence fcm g. for example. O Deﬁnition 6.2. cm 2 6D cm 2 .t kT /.dec/ O Pbit D P[b` 6D b` ] (6. It is characterized by Q the bit rate 1=Tcod . : : : .
We assume the message fb` g is composed of binary i. ž E sCh : average energy of an isolated pulse (V2 s).dec/ Typically. It is equal to the time interval between two consecutive bits of the information message.458 Chapter 6. sCh . ž M: cardinality of the alphabet of symbols at the transmitter. It expresses the minimum value of PsCh such that the system achieves a given performance in terms of bit error probability. ž E b : average energy per information bit (V2 s/bit). ž Tb : bit period (s). ž 0: conventional signaltonoise ratio at the receiver input.dec/ Chapter 5) and Pbit ' 10 7 –10 11 for data messages.i. we consider the desired signal at the receiver input. Parameters of a transmission system We give several general deﬁnitions widely used to describe the various modulation systems that will be treated in the following sections. ž N0 =2: spectral density of additive white noise introduced by the channel (V2 /Hz). . ž 0 I : signaltonoise ratio per dimension.t/ D s. ž ¹: spectral efﬁciency of the system (bit/s/Hz). ž T : modulation interval or symbol period (s). ž Twi : effective receiver noise temperature (K). for an ideal AWGN channel sCh . ž PsCh : available power of the desired signal at the receiver input (W). ž S: sensitivity (W). symbols. ž 1=T : modulation rate or symbol rate (Baud). it is required Pbit ' 10 2 –10 3 for PCM or ADPCM coded speech (see .t/.d. ž R I : rate of the encodermodulator (bit/dim). As in practical systems the transmitted signal s is distorted by the transmission channel. ž Bmin : conventional minimum bandwidth of the modulated signal (Hz). ž I : number of dimensions of the signal space or of the signal constellation. in particular. ž L b : number of information message bits per modulation interval. Modulation theory . ž MsCh : statistical power of the desired signal at the receiver input (V2 ). ž E I : average energy per dimension (V2 s/dim). ž Rb D 1=Tb : bit rate of the system (bit/s).
Statistical power of the desired signal at the receiver input: MsCh D E sCh T (6.101) (6.96) log2 M I (6. Number of information bits per modulation interval : via the channel encoder (COD) and the bitmapper (BMAP). In general we have L b Ä log2 M (6. for an uncoded system. for continuous transmission (see Chapter 7). 5. ak .93) 2. Rate of the encodermodulator : RI D Lb I (6.98) 7.97). or. Symbol period : T D Tb L b 4. with abuse of language.99) E sCh I (6.99) becomes: Eb D E sCh log2 M (6. L b information bits of the message fb` g are mapped in an Mary symbol. for transmission of an isolated pulse E sCh is ﬁnite and we deﬁne MsCh through (6.2. Average energy per dimension: EI D 6. In this case we also have RI D 3. on the other hand. Average energy per information bit: Eb D For an uncoded system. (6.102) Bmin D . Conventional minimum bandwidth of the modulated signal : Bmin D 1 2T 1 T for baseband signals for passband signals (6. MsCh is ﬁnite and consequently we deﬁne E sCh D MsCh T . Simpliﬁed model of a transmission system and deﬁnition of binary channel 459 Relations among parameters 1.6.100) EI Es D Ch RI Lb (6.97) (6.94) where the equality holds for a system without coding.95) We note that.
8. we have ¹D RI I Bmin T (6. Spectral efﬁciency: ¹D Lb 1=Tb D Bmin Bmin T (6. Signaltonoise ratio per dimension: 0I D 2E sCh EI D N0 =2 N0 I (6.106) is the ratio between the energy per dimension of an isolated pulse E I and the noise variance per dimension ¦ I2 given by (6. In terms of R I .105) given by 0D PsCh kTwi Bmin (6. the deﬁnition of Bmin will be different and will include the factor 1=M. if Bmin doubles. 10. for the same value of N0 =2. In the next sections some examples of modulation systems without channel coding are illustrated. . 9.105) is useful to analyze the system.103) In practice ¹ measures how many bits per unit of time are sent over a channel with the conventional bandwidth Bmin .460 Chapter 6. Using (6. from (4. We note that.93).92) we obtain an alternative expression of (6.N0 =2/2Bmin N0 Bmin T (6. Link budget: if the receiver is matched to the transmission medium for the maximum transfer of power. from (6.108) is usually employed to evaluate the link budget. 11. Modulation theory For the orthogonal and biorthogonal signals of Section 6.108) We observe that (6. Conventional signaltonoise ratio at the receiver input: 0D MsCh E sCh D .13).104) Later we will see that.107) It is interesting to observe that in most modulation systems it turns out 0 I D 0. the general relation becomes 0 I D 2R I Eb N0 (6.99). the statistical power must also double to maintain a given ratio 0.7. and (6.105) In general 0 expresses the ratio between the statistical power of the desired signal at the receiver input and the statistical power of the noise measured with respect to the conventional bandwidth Bmin . for most uncoded systems. R I D ¹=2.
109) In other words.115) The transmitter is shown in Figure 6. : : : . t0 /.114) as illustrated in Figure 6.0.22 for M D 8.t/ D p Eh (6.111) Average energy of the system:7 Es D Basis function: h Tx .t/ D Þn h Tx . M may take values larger than 2. a transmitted isolated pulse is expressed as sn .M C 1/ 2 M X iD1 i2 D M. An example for M D 8 is illustrated in Table 6.2M C 1/ 6 M X iD1 i3 D Â M Ã M C1 2 2 . M (6.23. The map is a Gray encoder (see Appendix 6.3.t/ where Þn D 2n 1 M (6. Let h Tx be a realvalued ﬁniteenergy pulse with support .M C 1/.t/ .6. PAM signals are obtained by modulating in amplitude the pulse shape h Tx .e. : : : . M (6. is the ﬁrst example of multilevel baseband signalling. 7 A few useful formulae are: M X iD1 iD M. i.3 Pulse amplitude modulation (PAM) Pulse amplitude modulation. 2. Pulse amplitude modulation (PAM) 461 6.1.113) M 1 X M2 1 Eh En D M nD1 3 (6. The ﬁlter output yields the transmitted signal sa0 .B). also called amplitude shift keying (ASK).110) t 2< n D 1.t/j2 dt (6. The bit mapper is composed of a serialtoparallel (S/P) converter followed by a map that translates a sequence of log2 M bits into the corresponding value of a0 . The minimum distance is equal to p dmin D 2 E h D d (6.112) Vector representation: sn D Þn p Eh n D 1. Energy of sn : 2 E n D Þn E h Z Eh D C1 1 jh Tx . The symbol Þn is input to an interpolator ﬁlter with impulse response h Tx .
M.1 on page 559: 1 (6.23. In this case.116) where sn D Þn . Transmitter of a PAM system for an isolated pulse.Eh 010 0 s5 Eh 110 s6 3 Eh 111 s7 5 Eh s8 7 Eh 101 100 bitmapping Figure 6. Minimum bandwidth of the modulated signal. Table 6.d=2/.24 and consists of a matched ﬁlter to h Tx followed by a sampler. or signal constellation. Gray coding of symbols (M D 8) Threebit sequence 000 001 011 010 110 111 101 100 Þn 7 5 3 1 1 3 5 7 a0 1 2 3 4 5 6 7 8 The type 1 implementation of the ML receiver is shown in Figure 6.117) Bmin D 2T . : : : . from (6.15) r is given by r D sn C w (6. n D 1. and w is a realvalued Gaussian r. Figure 6.v. From the observation of r.22. equal to the Nyquist frequency. The O transmitted bits are then recovered by an inverse bit mapper. with zero mean and variance N0 =2. Modulation theory d=2 E h s1 s2 7 E h 000 5 E h 001 M=8 s3 3 E h 011 s4 . a threshold detector yields the detected symbol a0 . Vector representation.462 Chapter 6.1 Bit map for a 8PAM. see Deﬁnition 7. of an 8PAM system.
2.2T / (6.105) it follows 0D Es N0 =2 (6. we have P[E j s M ] D P[E j s1 ] D P[a0 6D 1 j a0 D 1] O Ä Â DP r> 1 M 2 Ã ½ d j a0 D 1 M 2 Ã ½ d j a0 D 1 (6. letting d D 2 considering the outer constellation symbols separately from the others.120) Ã ½ M d 2 p E h . Spectral efﬁciency: ¹D Signaltonoise ratio: . n D 1.M=2// d.119) Note that (6.1=T / log2 M D 2 log2 M (bit/s/Hz) 1=.2n M/ Eh D . : : : . The p thresholds are set at . Pulse amplitude modulation (PAM) 463 Figure 6. M 1. implementation type 1. and Â Ä d D P Þ1 C w > 1 2 D P .24.6. as I D 1.1 Ä Â d M/ C w > 1 2 ½ Ä d DP w> 2 Â DQ d 2¦ I Ã .3. Symbol error probability: from the total probability theorem.118) from (6. of a PAM system for an isolated pulse.n . ML receiver.119) expresses 0 as the ratio between the signal energy and the variance of the noise component: therefore. it follows that 0 I D 0.
121) (6. for equally likely symbols we have Ã Â Ã½ Ä Â d d 1 C .Þn 1/ 2 2 2 2 Ä ½ Ä ½ d d DP w< CP w> 2 2 Ã Â d D 2Q 2¦ I where ¦ I2 D N0 =2. if an error event occurs. the autocorrelation function of h Tx .125) Equation (6. two other baseband modulation schemes are described: pulse position modulation (PPM) and pulse duration modulation (PDM). Modulation theory and O n D 2.123) (6. M 1 P[E j sn ] D P[a0 6D n j a0 D n] ¦ ² ¦ Ä² ½ d d [ r > . Then. in other words.464 Chapter 6. : : : . it is very likely that one of the symbols at the minimum distance from the transmitted symbol is detected. substitution of (6.119) into (6. the bit error probability is given by Pbit ' Pe log2 M valid for 0 × 1 (6.Þn C 1/ j a0 D n D P r < .124) Assuming that Gray coding is adopted at the transmitter.Þn 1/ 2 2 ½ Ä ½ Ä d d d d C P Þn C w > .Þn C 1/ D P Þn C w < . for 0 sufﬁciently large.25 for different values of M.125) expresses the fact that. assuming absence of ISI at the decision point.3). .123) yields ! r Ã Â 3 1 Pe D 2 1 Q 0 M M2 1 (6.122) 6 M2 Es 1 N0 In terms of 0. 8 These results are valid also for continuous transmission with modulation period T .3.C. Curves of Pbit as a function of 0 are shown in Figure 6.t/ must be a Nyquist pulse (see Section 7. Thus with high probability only one bit of the log2 M bits associated with the transmitted symbol is incorrectly recovered. In the Appendix 6.M 2/2Q 2Q Pe D M 2¦ I 2¦ I Â Ã Â Ã 1 d D2 1 Q M 2¦ I In terms of E s we get8 Â Pe D 2 1 1 M Ã Q s ! (6.
. In this section we consider the case of an isolated pulse transmitted at instant k D 0. modulated by h Tx .2n 1/ C '0 .127) that is. Let h Tx be a realvalued ﬁniteenergy baseband pulse with support .126) then the generic transmitted pulse is given by sn . Bit error probability as a function of 0 for MPAM transmission. signals are obtained by choosing one of the M possible values of the phase of a sinusoidal function with frequency f 0 .128) ³ A more general deﬁnition of 'n is given by 'n D M .2n M 1/ n D 1.4. where '0 is a constant phase.25.t/ D h Tx . M.6. Consequently.2³ f 0 t C 'n / t 2< n D 1.t/ cos. Phaseshift keying (PSK) 465 10 −1 10 −2 M=16 10 −3 M=8 Pbit 10 −4 M=4 10 −5 M=2 10 −6 0 5 10 15 20 Γ=2E /N (dB) s 0 25 30 35 Figure 6.t/e j2³ f 0 t ] (6. that determines the transmitted signal phase at instant kT .t/ D Re[e j'n h Tx .4 Phaseshift keying (PSK) Phaseshift keying is an example of passband modulation. M (6.66) by assuming h . t0 /. : : : . 6.127) is given by sn .473).v.t/. : : : . M (6. n D 1. An alternative expression of (6. where w is the rectangular window of duration T deﬁned Tx T T 9 in (1.0.10 In the following sections we will denote by Âk the r. : : : . the values of Âk are given by 'n .t/ D w . Let9 'n D ³ . 10 We obtain (6.
2n M ÁiT 1/ n D 1.129) Energy of sn : if f 0 is greater than the bandwidth of h Tx . Modulation theory Moreover. q We note that the desired signal at the decision point.I h Tx .2³ f 0 t/ Eh s 2 h Tx .130) ³ 1/ (6.Q ³ . sin .131) (6.t/ sin.t/ D Þn.2n we have sn . The minimum distance is equal to p p ³ ³ D 2E h sin (6. using Parseval theorem we get En D Average energy of the system: Es D Basis functions: s 1 .2n sn D 2 M Á ³ 1/ .133) (6.466 Chapter 6.138) dmin D 2 E s sin M M .134) D 2 .t/ D 2 h Tx .136) Vector representation: r ³ Eh h cos .t/ cos.132) Þn. : : : .2³ f 0 t/ where Þn. M (6. Note that the signal constellation lies on a circle and the various vectors differ in the phase 'n . 2.2n M 1/ 1/ (6. setting Þn D e j'n D e j M .137) as illustrated in Figure 6. aside from the factor E h .2³ f 0 t/ (6.26 for M D 8.Q h Tx . s n .135) (6.I D Re[Þn ] D cos Þn. 2 coincides with Þn .t/ sin.2n M ³ D Im[Þn ] D sin .t/ M 1 X Eh En D M nD1 2 Eh 2 (6.t/ cos.2³ f 0 t/ Eh (6.
Signal constellation of an 8PSK system. we note that the basis functions (6. respectively. . and partially by a ﬁlter matched to h Tx .27. simple decision rules can be deﬁned. 8. by a Hilbert ﬁlter. 11 For the sake of notation uniformity with the following chapters.6.Q are input to interpolator ﬁlters h Tx . Phaseshift keying (PSK) 467 Figure 6. the projections of sn on the axis 1 (in phase) and on the axis 2 (quadrature) are also represented. sin. The transmitted signal (6.26 we note that the decision regions are angular sectors with phase 2³=M.4. The quadrature components Þn.2³ f 0 t/. 4.26.26.28.2³ f 0 t/. b2 . The bit mapper maps a sequence of log2 M bits to a constellation point with value Þn . together with the Gray coding of the various symbols represented by the bits b1 . The ﬁlter output signals are multiplied by the carrier signal.130) is obtained by adding the two components. cos. For M > 8 detection can be made by observing the phase vr of the received vector11 r D [r I . From the general scheme of Figure 6. For M D 2. From Figure 6. and by the carrier signal phaseshifted by ³=2.8. r Q ]T .136) are implemented partially by a correlator with a sinusoidal signal. b3 . for example. A PSK transmitter for M D 8 is shown in Figure 6. The type 1 implementation of the ML receiver is illustrated in Figure 6. respectively. the components r and r of r will be indicated 1 2 by r I and r Q .I and Þn. In Figure 6.
468 Chapter 6. Thresholds are set at . Minimum bandwidth of the modulated signal (passband signal): Bmin D Spectral efﬁciency: ¹D . Transmitter of an 8PSK system for an isolated pulse. . of an MPSK system for an isolated pulse. M. implementation type 1.28. Modulation theory Figure 6.2³ =M/n.1=T / log2 M D log2 M (bit/s/Hz) 1=T (6. Figure 6.27.140) 1 T (6. ML receiver. n D 1. : : : .139) We note that for M D L 2 we have the same spectral efﬁciency as PAM.
Decision region for an MPSK signal. where sn is given by (6.141) We note that 0 also expresses the ratio between the energy per dimension and the variance of the noise components. .6.29. ² Q j a0 D n/ d² I d² Q Rn (6. For a0 D n we get r D w C sn .143) Using polar coordinates.² I .105) we have 0D E s =2 Es D N0 ¦ I2 (6. (6. then. exploiting the symmetry of the signalling scheme we get Pe D P[E j sn ] D1 D1 D1 P[C j sn ] P[r 2 Rn j a0 D n] ZZ pr .142) where the angular sector Rn is illustrated in Figure 6.z/ dz (6. Symbol error probability: with equally likely signals.144) r θ w sm 2π M vr Figure 6. Phaseshift keying (PSK) 469 Signaltonoise ratio: from (6.16).137).29. observing (6. we get Z Pe D 1 ³ M ³ M pÂ . moreover 0 I D 0 if M > 2.142) becomes ² Ä ZZ Á2 Á2 ½¦ p p 1 1 ²I exp E s cos 'n C ² Q E s sin 'n Pe D 1 d² I d² Q N0 Rn ³ N 0 (6.4.
The information in the other half can be deduced by symmetry and is therefore redundant. substituting (6. for M ½ 4 we can use the approximation (6. hence dmin D 2 E s .150) Moreover.z/ D e E s =N0 ( 1C s 2³ " ³ E s . This result is due to the fact that a BPSK system does not efﬁciently use the available bandwidth 1=T : in fact only half of the band carries information. Then I D 1.149) 1 . where '0 is an arbitrary phase.146) in (6. We consider in detail two particular cases.145) to obtain s Es Es sin2 z pÂ .t/ cos. If E s =N0 × 1. Therefore 0 I D 20 p E s and (6.147) Assuming that Gray coding is adopted at the transmitter. and observing (6. The integral (6.144).141). and a basis is given by the function s 2 h Tx . .E s =N0 / cos2 z e 2 cos z 1 N0 s Q 2E s cos z N0 !#) (6.31.148) Curves of Pbit as a function of 0 D E s =N0 are shown in Figure 6. Binary PSK (BPSK) For M D 2 we get '1 D '0 and '2 D ³ C '0 .t/ D Eh The signal constellation.146) ³ N0 In turn.z/ ' cos ze N0 (6.30.363) in (6.2³ f 0 t C '0 / (6.470 Chapter 6. Modulation theory where pÂ .144) cannot be solved in closed form. we get s ! 2E s ³ Pe D 2Q sin N0 M p D 2Q ³Á 20 sin M (6. comprises the vectors s1 D p p s2 D E s .145) for ³ Ä z Ä ³ . it is ¹ D 1. illustrated in Figure 6. the bit error probability is given by Pbit D Pe log2 M valid for Es ×1 N0 (6.
Bit error probability as a function of 0 for MPSK transmission. 20/ The transmitter and the receiver for a BPSK system are shown in Figure 6. Figure 6. The inverse bit mapping to recover the bits of the information message is straightforward. From (6. Phaseshift keying (PSK) 471 10 −1 10 −2 M=32 10 −3 M=16 Pbit 10 −4 M=8 10 −5 M=4 M=2 10 −6 0 5 10 15 20 Γ=E /N (dB) s 0 25 30 35 Figure 6.32 and have a very simple implementation.6. and using (6.4. the evaluation of Pe yields Pe D Pbit s DQ 2E s N0 ! (6. the decision element implements the “sign” function to detect NRZ binary data.141). At the receiver. .151) p D Q.64).30.31. Signal constellation of a BPSK system. The bit mapper of the transmitter maps ‘0’ in ‘ 1’ and ‘1’ in ‘C1’ to generate NRZ binary data (see Appendix 7. obtained for antipodal signals.A).
wQ > D 1 Q (6.130).34. Quadrature PSK (QPSK) PSK for M D 4 is usually called quadrature PSK (QPSK).154) The QPSK transmitter is obtained by simpliﬁcation of the general scheme (6. Modulation theory Figure 6. as w I and w Q are statistically independent.472 Chapter 6. we get p Áh Pe D 1 P[C] D 1 P[C j s1 ] D 2Q 0 1 For 0 × 1. As the decision thresholds are set at .2.153) (6. Schemes of transmitter and receiver for a BPSK system with '0 D 0. the probability of correct decision is given by s !!2 p p ½ Ä Eh Eh Eh P[C j s1 ] D P w I > .134) E s D E h =2 and 0 D E s =N0 . The ML receiver for QPSK is illustrated in Figure 6. 3³=2/ . as illustrated in Figure 6.155) p Ái 0 1 2Q (6.152) 2 2 2N0 As from (6. ³=2.32. The binary bit maps are given in Table 6. ³. the following approximations are valid: p Á Pe ' 2Q 0 and Pbit ' Q p Á 0 (6.35.33.0. With reference to the vector representation of Figure 6.
Table 6. QPSK transmitter for an isolated pulse.Q ) p 1=p2 1= 2 .6. Phaseshift keying (PSK) 473 φ2 b 1 b2 01 11 s2 s1 E s = Eh /2 φ1 s3 00 s4 10 Figure 6. Figure 6.4. Binary bit map b1 (b2 ) 0 1 Þn. Signal constellation of a QPSK system.2 Binary bit map for a QPSK system.34.33.I (Þn.
156) . By the differential noncoherent method.5 Differential PSK (DPSK) We assume now that the receiver recovers the carrier signal. a receiver estimates 'a from the received signal. At the receiver the matched ﬁlter plus sampler becomes an integrator that is cleared before each integration over a symbol period of duration T . except for a phase offset of 'a . using a simple threshold detector with threshold set at zero. In other words ž for MPSK. (see Figure 6. Modulation theory Figure 6. the transmitter ﬁlter is a simple holder. a receiver detects the data using the difference between the phases of signals at successive sampling instants.:::. M M M ¦ (6. 6. We observe that. with Âk 2 ² . it consists of an integrateanddump. ML receiver for a QPSK system.2³ f 0 t 'a /.35. To prevent this problem there are two strategies.474 Chapter 6. and considers the original constellation for detection. using the signal O O r e j 'a . Consequently. decisions can be made independently on r I and r Q . In other words. In this case s n coincides with E s e j'a Þn . where Þn is given by (6. it is as if the constellation at the receiver were rotated by 'a . for h Tx . In particular. By the coherent method. the phase of the transmitted signal at instant kT is given by (6.t/. the reconstructed carrier is cos.126).33).28. where 'a is the estimate of 'a . with reference to the scheme p Figure 6. .129).2M 1/³ ³ 3³ .t/ D K wT .
M 1/ Âk 2 0.6. However. the phase associated with the transmitted signal at instant kT is equal to that transmitted at the previous instant . We note that the decision thresholds for Âk are now placed at .³=M/. 1 sin 1 C sin Pe 1 C Q 1 N0 M N0 M (6. k D k 1 C Âk M M (6. : : : . . Ð/ (see Appendix 6.12 the transmitted phase at instant kT is given by ¦ ² 2³ 2³ 0 0 . k k 1 D Âk (6.5. n D 1.k 1/T plus the increment Âk .161) 12 Note that we consider a differential noncoherent receiver with which is associated a differential symbol encoder at the transmitter (see (6. .159) are discussed in Chapter 18.369) can be used and we get "s r Âr Ã# Es ³ ³ 1 C sin 1 sin Pe ' 2Q N0 M M s ' 2Q Es ³ sin N0 M ! (6. the approximation (6.1 Error probability for an MDPSK system For E s =N0 × 1. 3] s s ! Es Es ³Á ³Á . Differential PSK (DPSK) 475 ž for MDPSK.5.A) it can be shown that the error probability of an isolated symbol is approximated by the following bound [2.160) s s ! ³Á ³Á Es Es Q1 1 C sin 1 sin .158) In any case.157)) or ((6. For a phase offset equal to 'a introduced by the channel. as we will see in the next section. the phase of the signal at the detection point becomes k D 0 k C 'a (6. which can assume one of M values.169)). three differential noncoherent receivers that determine an estimate of (6.2n 1/. if M is large. N0 M N0 M Moreover.:::. For phasemodulated signals.159) and the ambiguity of 'a is removed. M.157) that is. a differential encoder and a coherent receiver can be used.Ð. 6. using the deﬁnition of the Marcum function Q 1 .
3] Pe D 2Q 1 .ab/ e 0:5.3 dB for M D 4.1 N0 p 1=2/ s bD p Es .a. Modulation theory For Gray coding of the values of Âk in (6. 3] Pbit D Pe D 1 e 2 For M D 4.147) is given in Figure 6. for Pbit D 10 3 .36: we note that. DPSK presents a loss of only 1.165) I0 .a 2 Cb2 / Es N0 (6. Comparison between PSK and DPSK.163) (6. As a DPSK receiver is simpler as compared to a coherent PSK receiver. a comparison in terms of Pbit between DPSK (6.156). the bit error probability is given by Pbit D Pe log2 M (6. the exact formula is [2.216).164) and where the function I0 is deﬁned in (4. For M D 2.1 C 1=2/ N0 (6. for M D 2 DPSK is usually preferred to PSK.161) and PSK (6. and to 3 dB for M > 4.2 dB in 0 for M D 2.36.476 Chapter 6. in that it does not require recovery of the carrier phase.161). b/ where s aD Es .162) where Pe is given by (6. Using the previous results. that increases to 2. the exact formula of the error probability is [2. 10 −1 PSK DPSK −2 10 10 −3 Pbit 10 −4 M=2 M=4 M=8 M=16 M=32 10 −5 10 −6 5 10 15 20 Γ (dB) 25 30 35 Figure 6. .
3.166) N where ý denotes the modulo 2 sum. In particular. BPSK system without differential encoding. 1g. Binary case (M = 2. bk 2 f0. and13 ck D ck 1 if bk D 1. differentially encoded BPSK) Let bk be the value of the information bit at instant kT . 6.3 Bit map for a BPSK system. If fck g are the detected coded bits at the receiver. This drawback can be mitigated if the reference sample is constructed by using more than one previously received samples [4]. In this way we establish a gradual transition between differential phase demodulation and coherent demodulation. if the reference sample is constructed using the samples received in the two previous modulation intervals.2 Differential encoding and coherent demodulation If 'a is a multiple of 2³=M. therefore ck D ck 1 if bk D 0. and bk D 0 causes a phase repetition.5. c k O O 1/ D ck ý ck O O 1 (6. if the previously received sample is used as a reference. ³ g is associated with bk by the bit map of Table 6. N N N . at the receiver the phase difference can be formed between the phases of two consecutive coherently detected symbols.5. Differential PSK (DPSK) 477 Note that. The phase Âk 2 f0. instead of between the phases of two consecutive samples. the information bits are O recovered by O bk D c k ý .167) Table 6. For the bit map of Table 6.4 we have that bk D 1 causes a phase transition. especially for M ½ 4. In this case. because both the current sample and the reference sample are corrupted by noise. symbols are differentially encoded before modulation. bk Transmitted phase Âk (rad) 0 1 0 ³ 13 c denotes the one’s complement of c: 1 D 0 and 0 D 1. DPSK gives lower performance with respect to PSK. we encode the information bits as ý bk bk 2 f0.6. 1g. DPSK and PSK yield similar performance [4]. Differential encoder. Decoder. 1g k½0 (6. For any c ck D ck 1 1 2 f0.
Ch (6.2M 1/³ =Mg. In fact. .478 Chapter 6.M 1/2³=Mg in f k g. 3³ =M. 1. . : : : . At the receiver the information sequence is recovered by O O O (6.172) (6. Approximately. : : : .169) ý dk M where ý denotes the modulo M sum. M we have ck D ck M 1 1g.170) dk D c k ý . we observe that an error in fck g O causes two errors in fdk g. Â Ã Ä Â Ã½ O ck ý j ý O ck 1 ý j O D c k ý . in this case fck g becomes fck O .Ch Pbit. In this case (6. ck Transmitted phase 0 1 0 ³ k (rad) O We note that a phase ambiguity 'a D ³ does not alter the recovered sequence fbk g: in 0 D c ý 1g and we have O Ok fact.Ch the channel error probability.4 Bit map for a differentially encoded BPSK system.157). with dk 2 f0. This encoding and bitmapping scheme are equivalent to (6.Ch C 8Pe. a two step procedure is adopted: 14 If we indicate with P e.169).ck O O 1 ý 1/ D ck ý ck O O 1 O D bk (6. c k 1 / D dk O O (6. 1. To combine Gray encoding of values of ck with the differential encoding (6. is worse as compared to a system with absolute phase encoding. then the error probability after decoding is given by [2] Binary case Quaternary case Pbit D 2Pbit. Modulation theory Table 6. : : : .Ch ] 4 4Pe.M 1/g in the sequence fck g.Ch [1 Pe D 4Pe.14 which causes a negligible loss in terms of 0. Because ck 2 f0.173) 2 3 8Pe. O However.168) Multilevel case Let fdk g be a multilevel information sequence. O corresponding to a phase offset equal to f0. for small Pe . Pe increases by a factor 2. the phase asso ciated with the bit map is k 2 f³=M. 2³ =M. c k 1 / M It is easy to see that an offset equal to j 2 f0.171) M M M M Performance of a PSK system with differential encoding and coherent demodulation by the scheme of Figure 6. . does not O cause errors in fdk g.28. 1. : : : .ck ý 1/ ý . M 1g.Ch . : : : . up to values of the order of 0:1.
Three information bits 0 0 0 0 1 1 1 1 0 0 1 1 1 1 0 0 0 1 1 0 0 1 1 0 values of dk 0 1 2 3 4 5 6 7 1. . ck / 2 f0.1/ .5 Gray coding for M D 8.174) The bit map is given in Table 6.1/ .1 (Differential encoding 2B1Q) We consider a differential encoding scheme for a fourlevel system that makes the reception insensitive to a possible change of sign of the transmitted sequence.6. .1/ c k D dk ý c k 1 .0/ For M D 4 we give the law between the binary representation of dk D . .0/ O. 2.1/ O.1/ . For M D 4 this implies insensitivity to a phase rotation equal to ³ in a 4PSK signal or to a change of sign in a 4PAM signal.0/ ck ck Transmitted symbol ak (6.1/ . represent the values of dk with a Gray encoder using a combinatorial table.6 Bit map for the differential encoder 2B1Q. Differential PSK (DPSK) 479 Table 6. as illustrated for example in Table 6.6.5. 1g.1/ dk D c k ý c k 1 O.1/ O.dk .0/ O. The equations of the differential decoder are O.1/ dk D c k ý c k Table 6.5 for M D 8.0/ .1/ c k D dk ý c k (6.5.i dk / 2 f0.175) 0 0 1 1 0 1 0 1 3 1 1 3 . dk /.ck .169). determine the differentially encoded symbols according to (6.0/ . ck /.0/ . 1g: . Example 6.i .1/ . and the binary representation of ck D .
180) For a rectangular constellation M D L 2 .t/ sin.182) .177) indicates that the transmitted signal is obtained by modulating in amplitude two carriers in quadrature. M (6. n D 1. : : : . Þn Q . 3.I . M (6.2³ f 0 t/ Þn. .Q h Tx . respectively.L 3/.I and Þn.176) If we modulate a symbol of the constellation by a real baseband pulse h Tx with ﬁnite energy E h and support .I h Tx .6 AMPM or quadrature amplitude modulation (QAM) Quadrature amplitude modulation is another example of passband modulation.176) we also have sn . 3. Energy of sn : if f 0 is larger than the bandwidth of h Tx .t/ D Re[Þn h Tx .t/ cos. Consider choosing a bit mapper that associates to a sequence of log2 M bits a symbol from a constellation of cardinality M and elements given by the complex numbers Þn n D 1. t0 /. and Þn I .L Then Es D D hence Eh D Es 3 M 1 (6. 1. : : : . However. In fact QAM may be regarded as an extension of PSK.0. . equation (6.183) 2 2 .480 Chapter 6. we have E n D jÞn j2 Average energy of the system: Es D M 1 X En M nD1 Eh 2 (6. we obtain the isolated generic transmitted pulse given by sn .Q 2 [ .Q denote the real and imaginary part of Þn . hence the name amplitude modulationphase modulation (AMPM). 1.L 1/] (6. From (6. Modulation theory 6. 2.179) (6.178) suggests that the transmitted signals are obtained not only by varying the phase of the carrier but also the amplitude.177) where Þn. if the amplitudes jÞn j. : : : . M.178) The expression (6.t/ D Þn.L 3 M 3 1 1/ Eh Eh 2 1/.181) (6. : : : .t/e j2³ f 0 t ] (6.2³ f 0 t/ t 2< n D 1. are not all equal. : : : .
. in a QAM system s n coincides with Þn . : : : .2³ f 0 t/ 2 .37 for various values of M.184) Vector representation: r sn D Eh [Þn. according to the law 6 2 (6.I . except for the factor E h .6.185) is normalized to one.185) as illustrated in Figure 6. Signal constellations of MQAM.187) dmin D E s M 1 1 0 φ (via Q) 11 1 1 11 11 11 00 0 0 00200 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 00 0 0 00M=256 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 00 0 0 00 11 11 1 0 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 00 11 0 0 00 0M=128 00 1 1 11 11 0 0 00 00 11 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 1 0 11 1 1 11 11 00 00 0 0 00 00 M=64 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 0 0 00 11 11 11 1 1 11 00 00 00 0 0 00 M=32 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 0 11 11 0 11 11 1 0 1 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 0 11 11 0 0 00 M=16 00 00 1 0 11 11 03 1 00 00 1 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 0 00 00 0 0 00 M=40 00 00 1 0 00 00 01 1 11 11 1 00 00 0 1 11 11 11 1 1 11 00 00 00 0 0 00 11 11 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 1 11 11 11111111111111 00000000000000 11 1 1 11 0 00 00 00 0 0 00111 11 1 1 11 0 00 00 0 0 00 1 11 11 1 0 11 1 1 11 11 11 00 0 0 00 00 00 φ 1 (via I) 1 1 11 1 11 11 0 0 00 0300 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 1 1 11 11 0 0 0 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 1 1 11 11 0 0 0 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 11 1 1 11 11 11 00 0 0 00 00 00 1 1 11 1 11 11 0 0 00 0 00 00 1 1 11 11 0 0 00 00 1 0 Figure 6. M (6. AMPM or quadrature amplitude modulation (QAM) 481 Basis functions: basis functions for the signals deﬁned in (6. we need to increase the average energy of the system by about 3 dB. to maintain a given dmin .37. Þn. q We note that.186) dmin D 2E h Consequently.6.2³ f 0 t/ 1 .t/ D Eh (6.185) the minimum distance between p two symbols is equal to 2E h . hence p (6.t/ D Eh s 2 h Tx .177) are given by s 2 h Tx . that is doubling M. for every additional bit of information. 2 It is important to observe that for the signals in (6.t/ sin.t/ cos. The term p Eh =2 in (6.Q ]T 2 n D 1.
as the 16QAM constellation is rectangular.39.38. we need to compute the M distances from the points sn . We note that the signals that are multiplied by the two carriers are PAM signals: in this example they are 4PAM signals. M.189) ¹ D log2 M . In general.39. Transmitter of a 16QAM system for an isolated pulse. and choose the nearest to r. b1 b 2 b 3 b 4 s4 1000 s3 1100 3 s2 0100 s1 0000 b 3 b4 00 01 11 10 s8 1001 3 s12 1011 s7 1101 1 1 s11 1 1111 s6 0101 1 s10 0111 s5 0001 s 93 0011 s16 1010 s15 3 1110 s14 0110 s13 0010 b 1 b2 10 11 01 00 Figure 6. Modulation theory Figure 6. The bit map and the signal constellation of a 16QAM system are shown in Figure 6. The transmitter of an MQAM system is illustrated in Figure 6. however.38 for M D 16.40.188) T (6. r Q ]T .482 Chapter 6. Signal constellation of a 16QAM system. We note that. The following parameters of QAM systems are equal to those of PSK: 1 Bmin D (6. the decision regions are also rectangular and detection on the I and Q branches can be made independently by observing r I and r Q . : : : . The ML receiver for a 16QAM system is illustrated in Figure 6. given r D [r I . n D 1.
10. ML receiver for a 16QAM system.2¦ I /.6.6.192) P[C j sn ] D 1 Q 2¦ I Ã½ Ä Â Ã½ Ä Â d d 1 2Q n D 2. 14.194) 2¦ I The probability of error is then given by Ã Â Ã Â Ã Â d d d 2 2:25Q ' 3Q Pe D 1 P[C] D 3Q 2¦ I 2¦ I 2¦ I where the last approximation is valid for large values of d=.193) P[C j sn ] D 1 Q 2¦ I 2¦ I Ä Â Ã½2 d P[C j sn ] D 1 2Q n D 6. 7. 11 (6. we have 0I D 0 (6. and 0D Moreover.195) .40. We ﬁrst evaluate the probability p correct decision for a 16QAM signal. rectangular constellation. 9. 8. 12. 15 (6. 5.190) Symbol error probability for M D L 2 .191) Es N0 (6. 3. AMPM or quadrature amplitude modulation (QAM) 483 Figure 6. (6. 13. 16 (6. We need to consider the following of cases (d D 2E h ): Ã½2 Ä Â d n D 1. 4.
. on average we need an increase of the energy of the system of 3 dB. Modulation theory In general.190).187).197) The bit error probability is approximated as Pbit ' Pe log2 M (6. (6. 10 −1 10 −2 M=256 10 −3 M=64 Pbit 10 −4 M=16 10 −5 M=4 10 −6 0 5 10 15 20 Γ=Es/No (dB) 25 30 35 Figure 6. to achieve a given Pbit . we get Â Pe ' 4 1 1 p M Ã Q Â d 2¦ I Ã (6. for a rectangular constellation with M elements.41.41. if we increase by one the number of bits per symbol. and (6. we need to increase 0 by 6 dB: in other words. We arrived at the same result using the notion of dmin in (6.186). We note that.484 Chapter 6.183). if M is increased by a factor 4. Bit error probability as a function of 0 for MQAM transmission with rectangular constellation.196) Another expression can be found in terms of 0 using (6. Â Pe ' 4 1 1 p M Ã Q s .198) Curves of Pbit as a function of 0 are shown in Figure 6.M 3 1/ ! 0 (6.
where only the argument of the Q function in the expression of Pbit is considered.95 12.92 15. while having the same spectral efﬁciency as PSK.92 1/ Ã (dB) 2 sin2 . 10 −1 10 −2 PSK M=32 QAM M=256 PSK QAM M=64 10 −3 M=16 PSK M=8 QAM M=16 PSK QAM M=4 Pbit 10 −4 10 −5 10 −6 0 5 10 15 20 Γ=Es/No (dB) 25 30 35 Figure 6. Â M 10 log10 4 8 16 32 64 128 256 3=. In general.65 4. QAM yields a lower Pbit .7.20 7. for given M ½ 4 and 0. For given Pbit and M.00 1. for given M.02 9.42.42.7 Gain of QAM with respect to PSK in terms of 0.M 0.6.³=M/ . Comparison between PSK and QAM systems in terms of Pbit as a function of 0. Table 6. the gain of QAM with respect to PSK is given in terms of 0 in Table 6.6. AMPM or quadrature amplitude modulation (QAM) 485 Comparison between PSK and QAM A comparison between the performance of the two modulation systems is shown in Figure 6.
: : : . Figure 6.7 6. : : : . : : : . 0.t/ D A sin. 1.199) j 0 A basis for these signals is simply given by the functions sn .t/ D p Es The vector representations of sets of orthogonal signals for M D 2 and M D 3 are illustrated in Figure 6.202) D or else f1 × 1 T (6. (b) M D 3. : : : .2³ f n t C '/ where the conditions 1 2T 1 fn C f` D k (k integer) T guarantee orthogonality among the signals.t/ dt D E s Ži j i. Vector representations of sets of orthogonal signals for (a) M D 2 and (b) M D 3. j D 1. hence Z t0 hsi . M (6. fn fn 1 0<t <T n D 1.0. sn . where p s n D E s [ 0. 0. Modulation theory 6. . s j i D si . Example 6.t/s Ł . t0 / and energy E s .486 Chapter 6. belongs to a set of M orthogonal signals with support .7.201) 1 2 n M p We note that the distance between any two signals is equal to dmin D 2E s .t/ n D 1. 0. : : : . 0 ]T (6.7.1 Modulation methods using orthogonal and biorthogonal signals Modulation with orthogonal signals The isolated generic transmitted pulse.43.1 (Multilevel FSK) 1.203) φ2 Es s 2 s1 Es φ1 φ2 Es s 2 s1 Es φ3 Es s3 φ1 (a) M D 2. M (6. Coherent sn . We will now consider a few examples of orthogonal signalling schemes.200) n .43. M (6.
determined as discussed in Appendix 6.2 (Binary modulation with orthogonal signals) In case we use only two orthogonal signals out of L available signals to realize a binary modulation.2³ f n t C 'n / where the conditions fn fn 1 0<t <T n D 1. from (6.v.207) M 2T (6.D. : : : .t/ D A sin. : : : . Modulation methods using orthogonal and biorthogonal signals 487 2. In the case of coherent demodulation. L j D 0. we have 0I D 0 (6.206) 1 T For noncoherent demodulation. L 1 (6. 0 D E s =. for Bmin D L=.204) D (6.211) (6.2T /. ¹ D .208) 2 log2 M M (6. j g n D 1. ¹D Example 6.3 (Code division modulation) Let f pn.7. Example 6.6.N0 M/. In (6.204) the uniform r.7.209) 2E s N0 M (6.105) 0D As I D M. we use the deﬁnition Bmin D Correspondingly. and 0 I D 20. we have Bmin D M=T .210) be the Walsh code of length L D 2m . M (6.205) 1 1 (k integer) or else f 1 × fn C f` D k T T guarantee orthogonality among the signals. 'n is introduced as each signal has an arbitrary phase.log2 M/=M. We note that in both cases the bandwidth required by the passband modulation system is proportional to M. : : : . the spectral efﬁciency is 2 L The system is not efﬁcient in exploiting the available bandwidth.7. . Noncoherent sn .103) we have ¹D and from (6.
7.105). j wTc . 2T / depends on the behavior in .t/g n D 1. : : : . Redundancy is introduced because the behavior of sn . from (6.215) from (6. Moreover.T.216) (6.220) are chosen as two Walsh signals of length L and duration L Tc D 2T . (6.219) Example 6.t j Tc / n D 1. Choosing M D L. and 0 I D 0. and (6. L.t/ in the interval .t/g n D 1. and we adopt the deﬁnition Bmin D M 2T (6.71.214) coincide with two Walsh signals of length L and duration L Tc D T . : : : . the elements of the set fsn . 0 D . n D 1.106) we obtain.212) The plots of sn .213) as the modulation is of the baseband type.7.103).2E s /=.488 Chapter 6. Now. RI D ¹D 0D 0I D 1 D 0:5 2 1 2 D . Modulation theory Choosing M Ä L. Setting Bmin D L 2T (6. for L D 8 are shown in Figure 6.93). Example 6.5 (Binary orthogonal modulation with coding) We have M D 2 as in the previous example. we note that also in this case the required bandwidth is proportional to 1=Tc D M=T . the signal sn is given by sn .N0 M/.218) (6. M 0 < t < L Tc D T (6. (6. respectively. 2 (6.217) (6.93) RI D log2 M M Correspondingly.L=2T /T L 2E s N0 L Es N0 (6.2 log2 M/=M. 2 (6. ¹ D . however. In this case fsn .t/ D L 1 X jD0 pn.4 (Binary code division modulation) We analyze in detail the previous example for M D 2 and L > M.
Bmin D L 4T 0:5 D 0:25 RI D 2 0:25 2 2D .223) (6.v. Probability of error The ML receiver is given by the general scheme of Figure 6. the number of information bits is equal to log2 M D 1: consequently. sn i] D Re 0 Assuming the signal sm is transmitted.Un mUn /.t/sn . with respect to the binary code division modulation presented above.225) (6. n D 1. n i] is the nth noise component. as T is the modulation interval we have L b D 1=2. M (6. Then.6. we have ÄZ Un D E s Žn m C Re D E s Žn m t0 0 Ł w. note that we also have I D 2.229) m (6.7.222) The other parameters are given by ¹D 0D and 0I D Es N0 (6.v. 2T /.228) .t/ dt n D 1. With reference to the interval .0. E[.200). : : : . where I D M and i is proportional to si according to (6. the r. M. the decision variables are given by ½ ÄZ t0 Ł r. Then fUn g. are Gaussian r. bandwidth and rate are halved.t/ dt ½ (6.221) (6.s fUn g are statistically independent with variance E s N0 =2.U` mU` /] D (6.L=4T /T L 4E s N0 L (6.0.t/sn . As the various signals have equal energy. Modulation methods using orthogonal and biorthogonal signals 489 the interval . T /.226) Un D Re[hr.s with mean mUn D E[Un ] D E s Žn and crosscovariance N0 E s Ž` n 2 Hence.224) We note that this case can be regarded as an example of a repetition code where the same symbol is repeated twice.8.227) C p E s wn where wn D Re[hw. : : : .
also called character. Um > Um 1 .230) With the change of variables ÞDp it follows Z C1 a E s N0 =2 1 2 Þ þDp r b E s N0 =2 (6. Therefore we have M=2 Pbit D Pe M 1 (6. Um > UmC1 . . for a given Pbit 0 decreases as M increases. : : : . page 294]. For each bit of the transmitted character. conditioned on sm .232) is independent of sm : consequently P[C j sm ] is the same for each sm . respectively. 15 The computation of the integral (6.233) (6.M 1/ wrong characters only M=2 yield a wrong bit.44 and Figure 6. Therefore for equally likely signals we get P[C j sm ] D P[C] D P[C j sm ] The error probability is given by Pe D 1 P[C] Z C1 1 2 Þ r 2E s N0 !2 2E s N0 !2 (6.235) 1 ' Pe 2 for M sufﬁciently large.a E s /2 2 E s . is equal to P[C j sm ] D P[Um > U1 .232) p e 2³ 1 We note that (6. as indicated in [5.b M / db1 : : : db M da 1 1 1 1 Dp 2³.a/ ÐÐÐ pU1 .N0 =2/ db #M 1 da (6. Modulation theory The probability of correct decision.234) was carried out using the Hermite polynomial series expansion. Curves of Pbit as a function of 0 D 2E s =. in Figure 6. : : : .Þ/] M 1 dÞ (6.45. The drawback is an increase of the required bandwidth with increasing M.234) 1 M 1 D1 [1 Q. This error event happens with probability Pe .E s N0 =2/ Z a e 1 b2 1 2 E s .Þ/] dÞ p e 2³ 1 Let M be a power of 2: with each signal sm we associate a binary representation. with log2 M bits.231) 1 [1 Q.E s N0 =2/ Z C1 e 1 1 . among the possible . Then a signal error occurs if a character different from the transmitted character is detected.490 Chapter 6.N0 =2/ " 1 p 2³. Um > U M ] ½ ÄZ a Z C1 Z a D pUm .b1 / : : : pU M .15 We note that.N0 M/ and E b =N0 are given. in contrast with QAM modulation.
7. 10 −1 10 −2 10 −3 Pbit 10 −4 10 −5 M=128 M=32 M=16 M=8 M=4 M=2 0 10 −6 −5 5 E / N (dB) b 0 10 15 20 Figure 6.45. . Modulation methods using orthogonal and biorthogonal signals 491 10 −1 10 −2 M=128 M=32 M=16 M=8 M=4 M=2 10 −3 Pbit 10 −4 10 −5 10 −10 −6 −5 0 5 Γ=2Es/(N0M) (dB) 10 15 20 Figure 6.6.44. Bit error probability as a function of Eb =N0 for transmission with M orthogonal signals. Bit error probability as a function of 0 for transmission with M orthogonal signals.
a useful approximation of Pbit is given by s ! Es M Q Pbit Ä 2 N0 (6. Exploiting the bound (6. for various (6. Comparison between the exact error probability and the limit (6. .236) Figure 6. Limit of the probability of error for M increasing to inﬁnity We give in Table 6.46 shows a comparison between the error probability obtained by exact computation and the bound (6.238) 1:59 dB is the minimum value of E b =N0 that is necessary to reach an error probability that can be made arbitrarily small for M ! 1 (see Section 6.46.236) for two values of M. In fact we can show that Pbit only if the following condition is satisﬁed Eb > N0 otherwise Pbit ! 1.84).8 the values of E b =N0 needed to achieve Pbit D 10 values of M. Modulation theory Figure 6.236) for transmission with M orthogonal signals. Therefore M!1 6.10).237) !0 M!1 1:59 dB (6.492 Chapter 6.
212) and their antipodal signals. 4PSK is a biorthogonal signalling scheme.243) .240) (6.0 5. For biorthogonal signalling with M signals. Modulation methods using orthogonal and biorthogonal signals 493 Table 6.3 7.6. M 23 24 25 26 210 215 220 : : : 1 E b =N0 (dB) 9. We give the parameters of the system in the two cases of noncoherent and coherent demodulation.5 7.2 Modulation with biorthogonal signals The elements of a set of M biorthogonal signals are M=2 orthogonal signals and their antipodal signals: for example.9 : : : 1:59 6.5 3. A further example of biorthogonal signalling with 2M signals is given by a signalling scheme using the M orthogonal signals in (6.4 8.239) (6.241) ¹D2 0D and. the required bandwidth is proportional to M=2.8 Values of Eb =N0 required to obtain Pbit D 10 6 for various values of M. as I D M=2. 2E s N0 M 0 I D 20 Baseband signalling or passband signalling with coherent demodulation: Bmin D M 1 2 2T (6.4 4.7.242) (6. Passband signalling with noncoherent demodulation: Bmin D M 1 2 T log2 M M (6.7.
we proceed as in the previous case. or matched ﬁlters. A bound to (6.249) The bit error probability can be approximated as Pbit ' 1 Pe 2 (6. : : : . : : : .251) for two values of M.248) [1 2Q. respectively. To compute the probability of correct decision.48. subsequently it selects si or si depending on the sign of Ui .M 2/Q N0 N0 where the ﬁrst term arises from the comparison with .247) The optimum receiver selects the output with the largest absolute value. jUm j > jU1 j. . jUm j 1 p e 2³ 1 2 Þ r 2E s N0 !2 (6. : : : .47 and in Figure 6. which provide the decision variables fUn g n D 1.N0 M/ and E b =N0 are plotted. jUm j > jU M=2 j] Z D 0 C1 1 j. Figure 6. and the second arises from the comparison with an antipodal signal. for various values of M.251) Pe Ä .244) (6.49 shows a comparison between the error probability obtained by exact computation and the bound (6. Assuming that sm is taken as one of the signals of the basis. in Figure 6.Þ/] M=2 1 dÞ The symbol error probability is given by Pe D 1 P[C j sm ] (6.245) 4E s N0 M 0I D 0 (6. Modulation theory ¹D4 0D and log2 M M (6. then P[C j sm ] D P[Um > 0.250) Curves of Pbit as a function of 0 D 4E s =. jUm j > jUm > jUmC1 j.249) for transmission with M biorthogonal signals is given by s s ! ! Es 2E s CQ (6.494 Chapter 6.M 2/ orthogonal signals. M 2 (6.246) Probability of error The receiver consists of M=2 correlators. jUi j.
Bit error probability as a function of Eb =N0 for transmission with M biorthogonal signals.6. . Modulation methods using orthogonal and biorthogonal signals 495 10 −1 10 −2 10 −3 Pbit 10 −4 10 −5 M=128 M=32 M=16 M=8 M=4 M=2 −5 10 −10 −6 0 Γ=4Es/(N0M) (dB) 5 10 15 Figure 6. 10 −1 10 −2 M=128 M=32 M=16 M=8 M=4 M=2 10 −3 Pbit 10 −4 10 −5 10 −6 −2 0 2 4 6 E / N (dB) b 0 8 10 12 14 Figure 6. Bit error probability as a function of 0 for transmission with M biorthogonal signals.47.7.48.
1] T cn. we derive the structure of the optimum receiver. we have L b D log2 M D n 0 . we have Bmin D 2T . : : : .t Q jT/ n D 1.0 . Every sequence of n 0 binary coefﬁcients cn D [cn.n 0 1/T / (6. it follows 2n 0 . and wT .8 Binary sequences and coding We consider a baseband signalling scheme where the transmitted signal is given by sn . wT . 1g (6.n 0 is allowed. Then E w is the energy of the pulse sn evaluated 1 on a generic subperiod T . : : : . cn. Comparison between the exact error probability and the limit (6.49. : : : .t/ D p rect t T =2 is the normalized rectangular window of T duration T (see (1. Uncoded sequences.t Q Q . C1g. j ž f 1.456)) with unit energy. As RI D log2 M D1 I (6. j ž f 1. Moreover. Interpreting the n 0 pulses cn.254) For a modulation interval Ts .252) wT .255) .t/.t/ D p Ew n0 1 X jD0 T 1 Q where cn. M 0 < t < n 0 T D Ts (6. hence M D I D n 0 .251) for transmission with M biorthogonal signals. Modulation theory Figure 6. j wT . 6.496 Chapter 6.253) as elements of an orthonormal basis.
2¦ I /2 (6.8. as for example those with index j D 0.266) (6. through appropriate binary functions. Binary sequences and coding 497 Es D n0 Ew Es D Ew I EI D Ew Eb D RI EI D 0I D 0D 2E w EI 2E b D D N0 =2 N0 N0 2E w Es D D 0I 1 N0 Ts N0 2T (6. Because the number of elements of the basis is always I D n 0 .57) u D 2 dmin d2 2E w 2E b D min D D 2N0 N0 N0 .252) corresponding to M D 2k0 binary sequences cn with n 0 components. : : : .259) (6.258) is used. assuming that only k0 components in (6. 1.254).268) . also the remaining n 0 k0 components.256) (6.263) (6. 1g arbitrarily: these components determine.260) Moreover. Coded sequences. we have RI D k0 log2 M D I n0 (6.265) (6. k0 1. can assume values in f 1. The error probability is determined by the ratio (6.262) (6. We consider a set of signals (6.261) where in the last step equation (6.6.267) Es D n0 Ew n0 Ew D Ew I EI n0 Eb D Ew D RI k0 EI D 0I D 0D 2E w EI k0 2E b D D N0 =2 N0 n 0 N0 Es D 0I 1 Ts N0 2T H Indicating with dmin the minimum number of positions in which two vectors cn differ.264) (6.252) is p equal to dmin D 4E w .257) (6. the minimum distance between two elements of the set of signals (6. we ﬁnd 2 H dmin D 4E w dmin (6.258) (6.
as the elements of the orthonormal basis (6. and H consequently a lower bit error probability. 2 3 2 3 2 3 2 3 1 1 C1 C1 6 17 6 17 6 C1 7 6 C1 7 7 7 7 7 c1 D 6 c2 D 6 c3 D 6 (6.269) c0 D 6 4 15 4 C1 5 4 15 4 C1 5 1 C1 1 C1 H 2 For this signalling system. alternative coding methods will be examined in Chapter 12. H We will discuss in Chapter 11 the design of codes that yield a large dmin for given values of the parameters n 0 and k0 . where the projections of the received signal r.50.253) are obtained by shifting the pulse wT .50. ML receiver for the signal set (6. the optimum receiver can be simpliﬁed Q as illustrated in Figure 6. the signaltonoise ratio at the decision point is given by c D H 4dmin E w d H R I 2E b D min 2N0 N0 (6.t/ onto the Figure 6. Modulation theory H In the case of uncoded sequences dmin D 1. and therefore dmin D 8E w . . An example of coding is given by the choice of the following vectors (code sequences or code words) for n 0 D 4 and k0 D 2.252). Using (6. we have dmin D 2. A drawback of these systems is represented by the reduction of the transmission bit rate Rb for a given modulation interval Ts .t/.265). Optimum receiver With reference to the implementation of Figure 6.270) We note that for a given value of E b =N0 the coded system presents a larger .8. if dmin R I > 1.498 Chapter 6.
253) are obtained sequentially.10). This procedure is usually called softinput decoding. cn o 1 ]T . c1 . the one that differs in the Q smallest number of positions with respect to the sequence c.50 is obtained by ﬁrst detecting the single components ci ž f 1. Passband PAM is considered as single sideband (SSB) modulation or double sideband (DSB) modulation (see Appendix 7. the receiver can be simpliﬁed by computing the Euclidean distance component by component.9 summarizes some important results derived in the previous sections.ri / O i D 0. cn 0 1 ]T is formed.21. In the latter case Bmin is equal to 1=T .6. : : : . 6.51. hence 0 D E s =N0 . that represents the minimum theoretical value of 0 I . Comparison between coherent modulation methods 499 Figure 6. in correspondence of a given R I . the binary vector c D [c0 . Q ci D ci D sgn. however the PAMCDSB technique has a Bmin that is double as compared to PAM or PAMCSSB methods. In the binary case under examination. for which Pbit can be made arbitrarily small by using channel coding without constraints in complexity and latency (see Section 6. 2k0 . In some receivers for coded systems. The vector components r D [r0 . Then we Q choose among the possible code sequences cn . We note .9 Comparison between coherent modulation methods Table 6.51.271) The resulting channel model (memoryless binary symmetric) is that of Figure 6. rn o 1 ]T are then used to compute the Euclidean distances with each of the possible code sequences.51. For a given value of the symbol error probability. a simpliﬁcation of the scheme of Figure 6. for a given noise level.252). components of the basis (6. For the uncoded system. We note that. This scheme is usually called hard input decoding and is clearly suboptimum as compared to the scheme with soft input. : : : . n D 1. according to O O the ML criterion. : : : . 1g according to the scheme Q Q Q of Figure 6. ML receiver for the signal set (6.252) under the assumption of uncoded sequences. PAMCSSB and PAMCDSB methods require the same statistical power to achieve a certain Pe . : : : . r1 . or equivalently the detected code sequence c D [c0 . The scheme of Figure 6.C).9. The result will be compared with the Shannon limit given by 0 I D 22R I 1. PAM.50 yields the detected signal of the type O O (6. : : : . we now derive 0 I as a function of R I for some multilevel modulations. n 0 1 (6. Successively. as illustrated in Figure 6.
M > 2/ Â 1 Â 1 Â 1 1 M 1 M Ã Q 2 log2 M log2 M log2 M 1 log2 M 2 1 1 1 log2 M 2 1 log2 M M 2 log2 M M 1 p M log2 M 1 2 log2 M Q Âr 2 sin2 2Q Es N0 ! orthogonal (BB) .500 Chapter 6.1=Tb /=Bmin (bit/s/Hz) 2 Encoder. Modulation theory Table 6.M M 2T M 4T 2 log2 M M log2 M M 2E s N0 M 4E s N0 M biorthogonal (BB) . and spectral efﬁciency.272) .107).9 Comparison of various modulation methods in terms of performance. z 0 / D 10 6 implies z 0 ' 22.104). related to 0 I through (6. p for all systems.2 3 1/ (6. MPAM. Pe D 10 6 .273) 1 0 D z0 (6. related to R I through (6. we have the following results. bandwidth. considering only the argument of the Q function in Table 6. Modulation Approximated symbol error probability Pe p Á 0 Âr 2Q Ã 2Q Ã 4Q p Á 20 ³Á 0 M r 1/Q r 2/Q M 0 2 Ã 1 T Âr 3 M2 6 1 3 1 M2 r M 1 0 0 Ã Ã ! 0 Minimum bandwidth Bmin (Hz) 1 2T 1 2T 1 T 1 T Spectral efﬁciency ¹ D . 1. As Q. A ﬁrst comparison is made by assuming the same symbol error probability. From 3 M2 and 0I D 0 we obtain the following relation 0I D z 0 2R I .Signalmodulator tonoise rate R I ratio 0 (bit/dim) 1 2E s N0 2E s N0 Es N0 Es N0 binary antipodal (BB) MPAM MPAM C SSB MPAM C DSB MQAM . as a function of ¹.9.M ! ! r M M 0 CQ 0 4 2 4 that an equivalent approach often adopted in the literature is to give E b =N0 .274) R I D log2 M (6.M D L 2 / BPSK o 2PSK QPSK o 4PSK MPSK .
orthogonal and biorthogonal modulation operate with R I < 1. and corresponding very small values of 0 I . Both R I and ¹ are doubled.277) RI D 1 2 1 0 D z0 (6. 5. the behavior of R I as a function of 0 I for Pbit D 10 6 is illustrated in Figure 6. An exact comparison is now made for a given bit error probability. moreover. The symbol error probability is approximately the same as that of orthogonal modulation for half the number of signals. MPSK. whereas for PAM the same efﬁciency is reached for MPAM D p 2 R I D MQAM .278) holds for M ½ 4. Biorthogonal modulation.6. the gap can be reduced by channel coding. As will be discussed in Section 6. for large R I . It turns out 0I D z 0 4R I 2 20 (6. for a given value of R I .M 1/Q ! (6. PAM and QAM allow a lower 0 I with respect to PSK. and is obtained by approximating sin.2 3 1/ (6. Orthogonal modulation.9. whereas PSK requires a much larger value of 0 I .52. MQAM. Using the Pbit curves previously obtained.276) We note that for QAM a certain R I is obtained with a number of symbols equal to MQAM D 22R I . PAM and QAM require the same value of 0 I .³=M/ with ³=M. We note that. 3. 4. .279) M 0 2 we note that the multiplicative constant in front of the Q function cannot be ignored: therefore a closedform analytical expression for 0 I as a function of R I for a given Pe cannot be found. Comparison between coherent modulation methods 501 2. From 3 M and 0I D 0 we obtain 0I D z 0 2R I .278) Equation (6. We also note that. We observe that the required 0 I is much larger than the minimum value obtained by the Shannon limit.275) log2 M (6.10. and ³ 2 with 10. Using the approximation r Pe ' .
234)). biorthogonal modulation (see (6. Tradeoffs for QAM systems There are various tradeoffs that are possible among the parameters of a modulation method. We note that to modify ¹ a modulator with a different constellation must be adopted. for different modulation methods and bit error probability equal to Pbit D 10 6 . in this region. but requires half the bandwidth. and PSK are bandwidth efﬁcient modulation methods as they cover the region for R I > 1.52.52. given ¹ (and the bandwidth).502 Chapter 6. Orthogonal and biorthogonal modulation are not very efﬁcient in bandwidth (R I < 1). Modulation theory Shannon limit Figure 6. or equivalently ¹ > 2. However. We assume that 1=Tb is ﬁxed. ﬁxed 0.41 for MQAM. where the parameter is ¹ D log2 M D . Comparison of modulation methods PAM. as illustrated in Figure 6. we obtain 0 as a function of ¹.1=Tb /=Bmin . but require much lower values of 0. the tradeoff is between Pbit and 0. in this region. The bandwidth is traded off with the power. As illustrated in Figure 6.52. from which the required bandwidth is also obtained. or Bmin < 1=. The parameter in the ﬁgure denotes the number of symbols M of the constellation. by increasing the bandwidth it is possible to decrease 0. We consider for example Figure 6. a slight decrease in 0 may determine a large increase of the bandwidth. that is 0. higher values of 0 are required to increase ¹.2Tb /. For a given Pbit . by increasing the number of levels: we note that. QAM. The Pbit of orthogonal or biorthogonal modulation is almost independent of M . we get ¹ as a function of Pbit . 0I required for a given rate RI . ﬁnally.249)) has the same performance as orthogonal modulation (see (6.
the capacity can be expressed in bits per dimension as CD 1 2 log2 . the redundancy of the alphabet can be used to encode sequences of information bits: in this case we speak of coded systems (see Example 6.93).1 C 0/ (bit/s) (6.282) . Limits imposed by information theory 503 and depends mainly on the energy E s of the signal and on the spectral density N0 =2 of the noise. the choice of a modulation scheme is based on the channel characteristics and on the cost of the implementation: until recently.103).93) of the encodermodulator rate. noncoherent receivers were preferred in radio mobile systems because of their simplicity. we have the cardinality of alphabet A is equal to M D 2 R I D 8. B D B d f . we deﬁne the maximum spectral efﬁciency as ¹max D C[b=s] D log2 . R I D L b =I . Some speciﬁc examples will be illustrated in Chapter 12. such that L b < log2 M.105) by choosing Bmin D B. where I is the number of signal space dimensions.t/ is a Gaussian random process with zero mean and constant power spectral density in the passband B. We recall the deﬁnition (1.10 Limits imposed by information theory We consider the transmission of signals with a given power over an AWGN channel having noise power spectral density equal to N0 =2. Channel capacity is deﬁned as the maximum of the average mutual information between the input and output signals of the channel [6.280) and (6. Equation (6.10.95). Let us consider for example a monodimensional transmission system (PAM) with an alphabet of cardinality A for a given rate R I .1 has rate R I D 3 (bit/dim).7. which belong to an I dimensional space.5). for example. For transmission over an ideal AWGN channel. For example. 6. as L b D 3 and I D 1.281) With reference to a message composed of a sequence of symbols. From (6. The mapping of sequences of information bits into sequences of coded output symbols may be described by a ﬁnite state sequential machine.1 C 0/ (bit/s/Hz) B (6. with bandwidth given by (1.1 C 0 I / (bit/dim) (6.280) is a limit derived by Shannon assuming the transmitted signal s.140). We recall the deﬁnition (6.280) where 0 is obtained from (6. Using (6. In addition to the required power and bandwidth. the encodermodulator for the 8PAM system with bit map deﬁned in Table 6.6. channel capacity is given in bits per second by C[b=s] D B log2 . 7]. even though the performance is inferior to that of coherent receivers (see Chapter 18) [2]. that is M > 2 R I from (6. Let us take a PAM system with R I D 3 and M D 16: redundancy may be introduced in the sequence of transmitted symbols.135) of the passband B associated with the frequency response R of a channel.
Á Þn /2 pr ja0 . where the Shannon limit given by (6. 6].53. such coding does not exist if R I > C. but it does not give any indication about the practical realization of channel coding. the computation of the maximum of C with respect to the probability distribution of the input signal can be omitted.283) (6. within which we can develop systems that allow reliable transmission of information. and also lower limited and approximated for large values of 0 I by a logarithmic function as follows: 0I − 1 : C Ä 0I × 1 : C ½ 1 2 1 2 log2 . p M 1 5 dÁ 4X nD1 pi pr ja0 . in terms of encodermodulator rate or.285) C D max pn p r ja0 . equivalently.2 (Shannon’s theorem) For any rate R I < C.504 Chapter 6. in terms of transmission bit rate (see (6. We note that Shannon’s theorem indicates the limits.282).0 I / (6.Á j Þi / i D1 where pn indicates the probability of transmission of the symbol a0 D Þn .284) Extension of the capacity formula for an AWGN channel to multiinput multioutput (MIMO) systems can be found in [9. The capacity C as well as the signaltonoise ratio given by (6.:::.e/ 0 I log2 . where 0 I is given by (6.Á j Þn / log2 6 M 7 p1 .286) 2¦ I2 With the further hypothesis that only codes with equally likely symbols are of practical interest. The capacity of a realvalued AWGN channel having as input an MPAM signal is given in bits per dimension by [11] 3 2 M X Z C1 pr ja0 .Á j Þn / / exp (6. The capacity can be upper limited and approximated for small values of 0 I by a linear function.287) Q is illustrated in Figure 6. 10].Þn C ¾ Þi /2 ¾ 2 2¦ I2 log Q D log2 M C e exp d¾ p 2 M nD1 1 2¦ I2 2³¦ I i D1 (6. By the hypothesis of white Gaussian noise. there exists channel coding that allows transmission of information with an arbitrarily small probability of error. we have ( ) .106). Capacity of a system using amplitude modulation Let us consider an MPAM system with M ½ 2.Á j Þn / (6. Theorem 6. The channel capacity is therefore given by " !# ¾2 M Z M X 1 1 X C1 . Modulation theory obtained assuming a Gaussian distribution of the transmitted symbol sequence.124) for which a symbol error probability .280)). We give without proof the following fundamental theorem [8.
We see from Figure 6. the additional achievable gain is negligible.282) is essentially equivalent to the capacity given by (6. at an error probability of 10 6 . for example.6. Capacity of an ideal AWGN channel for Gaussian and MPAM input signals.53 dB) is due to the choice of a uniform distribution rather than Gaussian for the set of input symbols. choosing 4PAM modulation. If the number of symbols in the alphabet A doubles.53 that for small values of 0 I the choice of a binary alphabet is almost optimum: in fact for 0 I < 1 (0 dB) the capacity given by (6. where we have a symbol error probability equal to 10 6 for 0 I D 13:5 dB. For large values of 0 I . The asymptotic loss of ³ e=6 (1. the capacity of multilevel systems asymptotically approximates a straight line that is parallel to the capacity of the AWGN channel.53. Let us consider. If the number of symbols is further increased. [From Forney and Ungerboeck (1998). We note that the curves saturate as information cannot be transmitted with a rate larger than R I D log2 M. we obtain in practice the entire gain that would be expected from the expansion of the input alphabet. and an arbitrarily small error probability can be obtained for 0 I D 5 dB. To achieve the Shannon limit it is not sufﬁcient to use coding techniques with equally likely input symbols.10. Therefore we conclude that. the uncoded transmission of 1 bit of information per modulation interval by a 2PAM system. we see that the coded transmission of 1 bit of information per modulation interval with rate R I D 1 is possible. by doubling the number of symbols with respect to an uncoded system.287) with a binary alphabet of input symbols. c 1998 IEEE. no matter how sophisticated they are: to bridge the gap . are also indicated [12]. Limits imposed by information theory 505 Figure 6.] equal to 10 6 is obtained for uncoded transmission. This indicates that a coded 4PAM system may achieve a gain of about 8:5 dB in signaltonoise ratio over an uncoded 2PAM system.
Moreover. as we will see in Chapter 13.290) M M2 1 We note that Pe is function only of M and 0 I . using (6. shaping and equalization.53 dB.506 Chapter 6.22C suggests the deﬁnition of the normalized signaltonoise ratio 0I D 0I 22R I 1 1/ D 1.289) bits of information are mapped into each transmitted symbol. if M is large.292) We note that the relation between Pe and 0 I is almost independent of M. if R I < C. then 0 I > 1. the gap that separates the system from capacity. Therefore the value of 0 I indicates how far from the Shannon limit a system operates. Coding techniques for small 0 I and large 0 I are therefore quite different: for low 0 I . Pe can therefore be expressed as Ã Âq Ã Ã Âq Â 1 N I ' 2Q Q Pe D 2 1 30 30 I M (6. We now consider two cases. .291) For large M. To reach capacity. to reach the capacity in channels with limited bandwidth. The average symbol error probability is given by (6.288) we obtain 0I D 0I M2 1 (6. High signaltonoise ratios.124). R I is equal to the capacity of the channel C and 0 I D 1 (0 dB). We note from Figure 6.289) and (6.288) for a given R I given by (6. coding must be extended with shaping techniques.53 that for high values of 0 I it is possible to ﬁnd coding methods that allow reliable transmission of several bits per dimension. R I D log2 M (6. ! r Ã Â 3 1 Pe D 2 1 Q 0I (6. the binary codes are almost optimum and the shaping of the constellation is not necessary. techniques are required that combine coding. This relation (6.54 between uncoded systems and the Shannon limit given by 0 I D 1. or. in other words. as it must be in practice. For an uncoded MPAM system. This relation is used in the comparison illustrated in Figure 6. For a scheme that achieves the capacity.93).282) can be expressed as 0 I =. for high 0 I instead constellations with more than two elements must be used. moreover. shaping techniques are required [13] that produce a distribution of the input symbols similar to a Gaussian distribution. Modulation theory of 1. Coding strategies depending on the signaltonoise ratio The formula of the capacity (6.
Bit error probability as a function of Eb =N0 for an uncoded 2PAM system.8). For systems with limited power and unlimited bandwidth. ž if R I D 1=2. For low values of 0 I the capacity is less than 1 and can be approximated by binary transmission systems: consequently we refer to coding methods that employ more binary symbols to obtain the reliable transmission of 1 bit (see Section 6. by using an orthogonal modulation with T ! 0 (see Example 6.3=2/ 0 I . then by increasing the bandwidth.6. c 1998 IEEE.7. then E b =N0 ³ . For low 0 I . [From Forney and Ungerboeck (1998).54.] Low signaltonoise ratios.293) .ln 2/ 0 I . usually E b =N0 is adopted as a ﬁgure of merit. Limits imposed by information theory 507 Figure 6. (6.10.107)): 22R I 1 Eb D 0I N0 2R I We note the following particular cases: ž if R I − 1.3). for example. and symbol error probability as a function of 0 I for an uncoded MPAM system. then E b =N0 D 0 I . or equivalently the number of dimensions M of input signals. For low values of 0 I it is customary to introduce the following ratio (see (6. both 0 I and R I tend to zero. if the bandwidth can be extended without limit for a given power. ž if R I D 1. then E b =N0 D .
as E b =N0 D . a reference uncoded MPAM system operates at about 9 dB from the Shannon limit: in other words. at this probability of error. the Shannon limit can be achieved by a code having a gain of about 9 dB.3=2/ 0 I . if. If the modulation rate of the coded system remains unchanged.296) N0 Coding gain Deﬁnition 6.508 Chapter 6.295) (6. 1:59 dB/ (6. then a binary code with rate R I D 1=2 can yield a coding gain up to about 10. in principle. the reference uncoded 2PAM system operates at about 12. Let us consider as reference systems a 2PAM system and an MPAM system with M × 1.295) afﬁrms that even though an inﬁnitely large bandwidth is used.5 dB from the ultimate Shannon limit. . we obtain the Shannon limit in terms of E b =N0 for a given rate R I as Eb 22R I 1 > 2R I N0 This lower limit monotonically decreases with R I . For Pbit D 10 6 . that is required to obtain a given probability of error relative to a reference uncoded system.294) in other words. the bandwidth can be extended only by a factor 2 with respect to an uncoded system.294) we ﬁnd that the Shannon limit in terms of E b =N0 is higher. ž if the bandwidth is limited.54 also shows the symbol error probability for an uncoded MPAM system as a function of 0 I for large M. respectively. from (6. for example. reliable transmission can be achieved only if E b =N0 > 1:59 dB. For Pe D 10 6 . we typically refer to 0 or 0 I . for small and large values of 0 I .2 The coding gain of a coded modulation scheme is equal to the reduction in the value of E b =N0 . Figure 6. instead. Thus a coding gain up to 12. In particular.293) and the Shannon limit 0 I > 1. ž if R I D 1. or in the value of 0 or 0 I (see (11.9)). assuming a limited bandwidth system.5 dB is possible. Modulation theory From (6. the symbol error probability or bit error probability for an uncoded 2PAM system can be expressed in two equivalent ways: s ! Ã Âq 2E b Pbit ³ Q 30 I D Q (6.8 dB. we examine again the three cases: ž if R I tends to zero. equation (6. if R I D 1=2 the limit becomes E b =N0 > 1 (0 dB). Figure 6. the ultimate Shannon limit is given by Eb > ln 2 N0 .54 illustrates the bit error probability for an uncoded 2PAM system as a function of both E b =N0 and 0 I . if the bandwidth can be sufﬁciently extended to allow the use of binary codes with R I − 1.
bb/ sn . ³ / (6.'1 / D A cos.298) 0 0 2 At the receiver.bb/Ł D Re[sn .297) every signal sn has a bandwidth smaller than f 0 .t/ e j2³ f 0 t ] Let us consider transmission over an AWGN channel of one of the signals n D 1.11 Optimum receivers for signals with random phase .11.t.299) (6.1 (Noncoherent binary FSK) The received signals are expressed as (see also (6. 6.t/ dt D (6. Receivers. in [ ³. 2.'2 / D A cos. M (6.6.bb/ sn . assuming a certain class of coding and decoding techniques.204)): s1 . for a given modulation and class of codes [2]. except. '2 2 U [ ³.bb/ where sn is the complex envelope of sn . with support .t/j dt En D sn .t. which do not rely on the knowledge of the carrier phase. relative to the carrier frequency f 0 . ³ /.t/ D Re[sn .v. however. we observe the signal r.t/ where .297) .t/e j' e j2³ f 0 t ] .'/ C w. .301) . : : : .302) 0<t <T 0<t <T (6. We give three examples of signalling schemes that employ noncoherent receivers. : : : .t. t0 /. 2.t/e j' (6. M e j2³ f 0 t In other words. Example 6.300) ] n D 1. for a phase ' that we assume to be a uniform r. Optimum receivers for signals with random phase 509 Cutoff rate It is useful to introduce the notion of cutoff rate R0 associated with a channel.t/ D sn .'/ D Re[sn . Therefore for a given channel we can determine the minimum signaltonoise ratio . are called noncoherent receivers.11.0.E b =N0 /0 is 2 about 2 dB above the signaltonoise ratio at which capacity is achieved.2³ f 1 t C '1 / s2 . at the receiver we assume the carrier is known. Typically. If in (6. then the energy of sn is given by Z t0 Z t0 1 . We sometimes refer to R0 as a practical upper bound of the transmission bit rate.bb/ 2 2 jsn . for codes with rate Rc D 1 (see Chapter 11).2³ f 2 t C '2 / where r AD 2E s T '1 .t.E b =N0 /0 below which reliable transmission is not possible.bb/ .
308) 0<t <T (6. and E s is the average energy of a pulse. The minimum value of f d is given by .510 Chapter 6. p/ dt Ã (6.t/ sn .11.3 (DSB modulated signalling with random phase) .bb/ We consider an Mary baseband signalling scheme.t.'/ D sn . '1 / and s2 .7.ρ j n.310) . 2³.307) Example 6. s1 . The received signals are expressed as .2³ f 0 t C '/ n D 1.t. that is modulated in the passband by the double sideband technique (see Example 1.t/g. that is for known '.303) where f d is the frequency deviation with respect to the carrier f 0 . p/ dt N0 0 1 N0 Z 0 t0 2 sn .3 on page 58).304) which is twice the value we ﬁnd for the coherent demodulation case (6.11. the ML criterion to detect the transmitted signal has been previously developed starting from (6. '2 / are orthogonal.'/ D 0 where A D p 4E s =T . fsn .t.N0 =2// Â Z t0 2 D K exp r.2³ f 0 t C '/ and s2 .t. : : : .t.'/ D A cos. : : : .t. M (6.305) 1 (k1 integer) T or else f0 × 1 T (6. We recall that if f 1 C f 2 D k1 and if 2 f d T D k (k integer) then s1 .309) ML criterion Given ' D p. n D 1. (6. Example 6. Modulation theory and f1 D f0 fd f2 D f0 C fd (6.t/ cos. The conditional probability density function of the vector r is given by 1 1 jjρ sn jj2 prja0 . p/ D p e N0 I .t.26). M. f d /min D 1 2T (6. for example.2 (Onoff keying) Onoff keying (OOK) is a binary modulation scheme where.bb/ sn .' .203).306) (6.
313) The dependency on the r. the maximum likelihood criterion yields the decision rule a0 D arg max Ln [ p] O n (6.27): Ã Â Â Ã Z t0 2 En exp Ln [ p] D exp r.315) Introducing the polar notation L n D jL n je j arg L n . pC2³ f 0 t/ dt ½Ã dp (6.316) De En N0 p 2 En jL j cos. ' is removed by taking the expectation of Ln [ p] with respect to ':16 Z ³ Ln D Ln [ p] p' .317) 2.6.216): 1. M (6.311) we deﬁne the following likelihood function. to that deﬁned in (6.bb/Ł r.314) becomes Ln D e En N0 1 2³ 1 2³ Z Z ³ ³ ³ ³ p 2 En Re[L n e e N0 jp] dp (6. : : : .t/ e j . 1 I0 .t/ sn .t/ sn . p/ d p ³ De En N0 1 2³ Z ³ ³ Â exp 2 Re N0 ÄZ 0 t0 .t.300). Optimum receivers for signals with random phase 511 Using the result Z 0 t0 2 sn .'/ dt D E n (6. .bb/ r.t. p arg L n / e N0 n dp We recall the following properties of the Bessel functions (4. We deﬁne 1 Ln D p En Z 0 t0 iŁ h .x/ D 2³ Z ³ ³ e x cos. but not equal. p/ dt n D 1.v. 16 Averaging with respect to the phase ' cannot be considered for PSK and QAM systems.312) N0 N0 0 Given ' D p. I0 .314) using (6. p Â/ dp 8Â (6. where information is also carried by the phase of the signal.t/ sn .t/ e j2³ f 0 t dt (6.x/ is a monotonic increasing function for x > 0. (6.11. which is equivalent.
the desired value jL n j coincides with the absolute value of the output signal of the phasesplitter at instant t0 .bb/ f 0 .t/ D Re[yn .318) Taking the logarithm we obtain the loglikelihood function Ã Â p 2 En En `n D ln I0 jL n j N0 N0 (6. .t/jtDt0 (6.a/ jL n j D jyn .bb/ .199) we have .55.t/ D jsn .bb/ A simpliﬁcation arises if the various signals sn have a bandwidth B much lower than . and then determines the squared magnitude.202).56.321) From (6.316) becomes Ln D e En N0 Ã Â p 2 En I0 jL n j N0 n D 1.bb/ yn .196) and (1.t/j cos. where '0 is a constant. Polar notation is adopted for the complex envelope: . the ML decision criterion can be expressed as a0 D arg max jL n j O n (6. Note that the available signal is . M (6. Alternatively.bb/ . the scheme ﬁrst determines the real and the imaginary parts of L n starting .322) The cascade of the phasesplitter and the “modulo” transformation is called the envelope detector of the signal yn .2³ f 0 t C arg yn .t/ e j'0 .322) and (1.323) Moreover. at the matched ﬁlter output the following relation holds . recalling (1.512 Chapter 6. : : : . rather than sn . if yn is the complex envelope of yn .t/je j8n . from (6.57.bb/ jL n j D jyn . For the generic branch. Modulation theory Then (6.55 for the case of all E n equal. and considering that both ln and I0 are monotonic functions.202)).t/ (see (1.324) . the generic branch of the scheme in Figure 6.t/e j2³ f 0 t ] .315)). In this case. composed of the I branch and the Q branch.320) Implementation of a noncoherent ML receiver The scheme that implements the criterion (6.bb/ from sn . .t/ (6.t/: this.bb/ sn .315). does not modify the magnitude of L n .320) is illustrated in Figure 6.bb/ sn . the bold line denotes a complexvalued signal.319) If the signals have all the same energy. however. can be implemented by a complexvalued passband ﬁlter (see (6. the matched ﬁlter can be real valued if it is followed by a phasesplitter: in this case the receiver is illustrated in Figure 6. As shown in Figure 6.t// (6.t/jtDt0 (6.bb/ .bb/ D jyn .
11. r(t) t0 s(bb)* (t0 t)e j(2π f0 t+ ϕ 0) n . 2 L n  2 Figure 6.6. we can use one of the schemes of Figure 6. Noncoherent ML receiver of the type squarelaw detector. to determine the amplitude jyn . . Optimum receivers for signals with random phase 513 Figure 6.4 (Noncoherent binary FSK) We show in Figure 6.11.1.55.58.11. if f 0 × B.56. Implementation of a branch of the scheme of Figure 6.t/j.bb/ Now.55 by a complexvalued passband matched ﬁlter. .59 two alternative schemes of the ML receiver for the modulation system considered in Example 6. Example 6.
57. we have N0 UTh D p I0 1 . (a) Ideal implementation of an envelope detector. Example 6.60 the receiver for the modulation system of Example 6.e E 1 =N0 / 2 E1 where E1 D A2 T 2 (6.514 Chapter 6. using passband matched ﬁlters.318).11.58. recalling (6. and (b) two simpler approximate implementations.2 where. Noncoherent ML receiver of the type envelope detector. Modulation theory Figure 6.11.5 (Onoff keying) We illustrate in Figure 6.325) . (a) (b) Figure 6.326) (6.
Envelope detector receiver for an onoff keying system. Two ML receivers for a noncoherent 2FSK system.59.6. r(t) T w (t)cos(2 π f t+ ϕ ) T 0 0 envelope detector ^ U U>UTh .60. Optimum receivers for signals with random phase 515 Figure 6.11. a0 =1 ^ U<UTh . . a =2 0 ^ a 0 Figure 6.
t/ cos.t/ sin.c 2 C w2.2³ f 2 t C '0 / dt Z 0 T Z 0 T w.t. Error probability for a noncoherent binary FSK system We now derive the error probability of the system of Example 6.61 the receiver for a baseband Mary signalling scheme that is DSB modulated with random phase.305). we show in Figure 6.t/ and Pbit D P[U1 < U2 j s1 ] Equivalently.t/ sin. we have ÂZ U2 D 0 T (6.333) w.2³ f 1 t C '0 / dt Z 0 T . '2 /. '1 / ? s2 .2³ f 2 t C '0 / dt Ã2 (6.t.327) (6.3.516 Chapter 6. Depending upon the signalling type.c D w1.328) p U1 and V2 D p U2 (6. Modulation theory Example 6.t/ sin.2³ f 2 t C '0 / dt Ã2 C ÂZ 0 T r. We assume that s1 is transmitted: then r.6 (DSB modulated signalling with random phase ) With reference to the Example 6.329) (6.c D w2.2³ f 1 t C '0 / dt (6.s w2.332) w.t/ cos.s D Z 0 T w.t/ C w.2³ f 2 t C '0 / dt (6. further simpliﬁcations can be done by extracting functions that are common to the different branches. recalling assumption (6.t/ D s1 . that is s1 .s D If we deﬁne w1.t/ cos.330) r.11.11.4. if we deﬁne V1 D we have Pbit D P[V1 < V2 j s1 ] Now.11.331) D where 2 w2.
61. . Two receivers for a DSB modulation system with Mary signalling and random phase.11. Optimum receivers for signals with random phase 517 Figure 6.6.
t/ is a white Gaussian random process with zero mean.c ] D E[w2.337) whereas V1 has a Rice probability density function v1 e pV1 .þx/ d x D eþ =2Þ Þ (6.c and w1.334) Â D AT cos. Modulation theory we have ÂZ U1 D 0 T r.s Ã2 Ã2 (6. Therefore V2 .s .c and w2.2³ f 1 t C '0 / dt '1 / C w1.s ] D where from (6.s are two jointly Gaussian r.N0 T =4/ Â I0 Ã v1 .c ] D E[w2.'0 2 AT sin.340) D 1 e 2 17 To compute the following integrals we recall the WeberSonine formula: Z C1 0 2 1 2 x e Þ.v1 C.v1 / dv1 Ã pV2 .v1 / D N0 T =4 2 .302) we also have N0 T 2 2 t2 / cos.'0 2 r AT Es T D (6.335) 2 2 As w. w2.2³ f 1 t C '0 / dt '1 / C w1.AT =2/ 1.2³ f 2 t1 C '0 / sin.s with E[w2.2³ f 2 t2 C '0 / dt1 dt2 D 0 E[w2.s ] D Z 0 T Z 0 T N0 Ž.v2 / dv2 0 Z D 0 C1 ÂZ v2 0 1 Es 2 N0 pV1 .s ] D 0 2 2 E[w2. with statistical power 2.339) .t/ sin.t1 2 (6.v 2/ (6.336) Similar considerations hold for w1.t/ cos.v2 / dv2 (6.v1 / N0 T =4 (6. has a Rayleigh probability density v2 e pV2 .v.N0 T =4/ 1.518 Chapter 6.c w2.AT =2/2 / 2.c Ã2 Ã2 C Â C ÂZ 0 T r.338) Consequently equation (6.330) assumes the expression17 Z C1 Pbit D P[V1 < v2 j V2 D v2 ] pV2 .x =2/ I0 .N0 T =4/.v2 / D N0 T =4 2 v2 2.
)BPSK (ρ =−1) 10 −5 DBPSK 10 −6 5 6 7 8 9 10 Γ=E /N (dB) s 0 11 12 13 14 15 Figure 6. The differential receiver for DBPSK directly gives the original uncoded bits. from (6. where f d satisﬁes the constraint (6.340) we have 1 E 1 2 Ns 0 e (6.62.6.e. for the same Pbit .305).11. thus from (6. such as DBPSK.62. .301). Performance comparison of binary systems The received signals are given by (6. is illustrated in Figure 6. Optimum receivers for signals with random phase 519 It can be shown that this result is not limited to FSK systems and is valid for any pair of noncoherent orthogonal signals with energy E s . and from (6.163) we have FSK(NC): Pbit D DBPSK: Pbit D 1 e 2 Es N0 (6. with coherent (CO) and noncoherent (NC) detection.343) N0 10 −1 10 −2 10 −3 FSK (ρ =0) FSK NC Pbit CO −4 10 (d.342) indicates that DBPSK is better than FSK by about 3 dB in 0. In particular.62.341) 2 A comparison with a noncoherent binary system with differentially encoded bits.75) it follows s ! Es FSK(CO): Pbit D Q (6.341) and (6.342) A comparison between (6. Bit error probability as a function of 0 for BPSK and binary FSK systems. The correlation coefﬁcient between the two signals is equal to zero. The performance of a binary FSK system and that of a differentially encoded BPSK system with coherent detection are compared in Figure 6.
Q are uncorrelated Gaussian r.344) We observe that the difference between coherent FSK and noncoherent FSK is less than 2 dB for Pe Ä 10 3 . in addition to a random phase.s. where ' 2 U .t/ n 2 f1.349).6.343) and (6. In polar notation g1 D jg1 je j' .a/ da (6.350) . 6.v. FSK systems with M > 2 are not widely used.345) where g1 D g1. from (6. is a function of jg1 j and therefore it is a random variable.v.I and g1. and becomes less than 1 dB for Pe Ä 10 5 .e. we regard the expressions of Pe obtained in the previous sections as functions of 0.I C jg1.346) where E s is the average energy of the transmitted signal. Modulation theory Taking into account differential decoding.a/ p0 . ³. We deﬁne the average signaltonoise ratio 0avg D Es E[jg1 j2 ] N0 1 0avg (6. see (6.64) we have18 s ! 2E s (d.347) Then the probability density of 0 is that of a chisquare r. that yields Z C1 Pe D Pe . also a random attenuation (see Section 4. we obtain:19 18 For a more accurate evaluation of the probability of error see footnote 14 on page 478.bb/ r.: p0 . (6. 19 For the computation of the integral in (6.)BPSK: Pbit ' 2Q N0 (6. (6. : : : . Therefore we consider Pe as the conditional error probability for a given value of jg1 j. To evaluate the mean error probability we apply the total probability theorem.12 Binary modulation systems in the presence of ﬂat fading We assume now that the channel introduces.215).Q is a Rayleigh r.342).348) To compute the performance of a signalling scheme in the presence of ﬂat fading. The received signal is expressed as .520 Chapter 6.349) we recall the following result: Z C1 Q 0 x p Ð1 þ 1 Þx dx D e þ 2 s 1 þ 2 þC Þ ! (6. Mg 0<t <T (6..t/g1 e j2³ f 0 t ] C w. the signaltonoise ratio at the receiver input. with zero mean and equal variance. As jg1 j determines the signal level at the input of the receiver.t/ D Re[sn .349) 0 Limiting ourselves to binary signalling schemes and substituting Pe given by (6. ³ / and pjg1 j is given by (4. 0D Es jg1 j2 N0 (6.v.344) in (6.341). We also note that because of the large bandwidth required.a/ D e a=0avg 1.5).206).a/ (6. that is g1.
we refer the reader to [14]. or at least highly uncorrelated. Orthogonal binary FSK with coherent detection s ! 0avg 1 Pbit D 1 2 2 C 0avg 2. . DBPSK Pbit D 1 2.1 C 0avg / (6. for the same N0 . For a systematic method to determine the performance of systems in the presence of a channel affected by multipath fading. Orthogonal binary FSK with noncoherent detection Pbit D 4.352) We note that both the above expressions are in practice a lower limit to Pbit . rather than exponentially as in the AWGN channel case: therefore a large transmission power is needed to obtain good system performance. that is exploiting channels that are independent.12. which is very hard to obtain under fading conditions. There are various diversity techniques. Frequency diversity: the same signal is transmitted using several carriers. noncoherent DPSK and FSK systems are valid alternatives.351) (6. 3. and to the references therein. as it is assumed that an estimate of the phase ' is available. separated from each other in frequency by an interval that is larger than the coherence bandwidth of the channel. We note that to achieve a certain Pbit . it is required a substantially larger E s as compared to the case of transmission over an AWGN channel. Binary modulation systems in the presence of ﬂat fading 521 1. To mitigate this problem it is useful to resort to the concept of diversity. 1. The basic idea consists in providing the receiver with several replicas of the signal via independent channels. Diversity In the previous section it became apparent that the probability of error for transmission over channels with Rayleigh fading varies inversely proportional to the signaltonoise ratio. for communication.63 and compared with the case of transmission over an AWGN channel. In case the uncertainty on the phase is relevant.353) The various expressions of Pbit as a function of 0avg are plotted in Figure 6.6.354) 1 2 C 0avg (6. so that the probability is small that the attenuation due to fading is high for all the channels. Differentially encoded BPSK with coherent detection s 0avg Pbit D 1 1 C 0avg (6.
16.1 Transmission methods Transmission methods between two users A transmission link between two users of a communication network may be classiﬁed as a) Full duplex. Combinations of the previous techniques: for the many techniques of combining available. and binary FSK systems.18 and to the bibliography [15. we can select the antenna that provides the signal with higher power. 4. Space diversity: multiple reﬂections from ground and surrounding buildings can make the power of the received signal change rapidly. 2. selection. Polarization diversity: several channels are obtained for transmission by orthogonal polarization.13 6. 17].13.522 Chapter 6.63. 3. both linear (equal gain. when two users A and B can send information to each other simultaneously. 6. for a ﬂat Rayleigh fading channel. Time diversity: the same signal is transmitted over different time slots. 5. we refer to Section 8. not necessarily by using the same transmission channels in the two directions. spaced by an interval that is larger than the coherence time of the channel. DBPSK. Modulation theory Figure 6. maximal ratio) and nonlinear (square law ). by setting two or more antennas close to each other. . Bit error probability as a function of 0avg for BPSK.
64b).1. N users may share the channel using one of the following methods. Examples of FDD systems are the GSM.20 1. If the duration of one slot is small with respect to that of the message. each slot is identiﬁed by an index i. when only A can send information to B. thus allowing fullduplex transmission. each in turn subdivided into N S adjacent subsets called slots.13. . We note that fullduplex transmission over a single band is possible also over radio channels. which uses a radio channel (see Section 17. CSMA/CD.64a)..71).2). N S 1 (see Figure 6.6. Subdivision of a sequence of modulation intervals into adjacent subsets called frames. which uses a twisted pair cable.13. Transmission methods 523 b) Half duplex. i D 0. that is the link is unidirectional. which uses a radio channel (see Section 17. and the pingpong BRISDN. which uses a twisted pair cable (see page 1146). an alternative approach is represented by random access techniques.2 Channel sharing: deterministic access methods We distinguish three cases for channel access by N users: 1.g. a) Frequencydivision duplexing (FDD): in this case the two users are assigned different transmission bands using the same transmission medium. Subdivision of the channel passband into N B separate subbands that may be used for transmission (see Figure 6. e. the receiver eliminates echo signals by echo cancellation techniques. Frequency division multiple access (FDMA): to each user is assigned one of the N B subbands. 2. b) Timedivision duplexing (TDD): in this case the two users are assigned different slots in a time frame (see Section 6. examples are the HDSL (see Section 17. Signalling by N0 orthogonal signals (see for example Figure 6. Three methods In the following we give three examples of transmission methods which are used in practice. c) Fullduplex systems over a single band: in this case the two users transmit simultaneously in two directions using the same transmission band. 6. and in general highspeed transmission systems over twistedpair cables for LAN applications (see Section 17. from A to B or from B to A. we speak of fullduplex TDD systems. when two users A and B can send information in only one direction at a time.2).1. collision resolution protocols [18] (see also Chapter 17). 3. Within a frame. : : : . ALOHA.13.A. The two directions of transmission are separated by a hybrid.A. as each user knows exactly at which point in time the channel resources are reserved for transmission. but in practice alternative methods are still preferred because of the complexity required by echo cancellation.2).6). alternatively. Examples of TDD systems are the DECT. and the VDSL. c) Simplex.1). 20 The access methods discussed in this section are deterministic.
there is then only one bit for the synchronization of the whole frame. Modulation theory Figure 6. In this case the frame is such that one bit per channel is employed for signalling: this bit is “robbed” from the least important bit of the 8bit PCM sample. As the duration of a frame is of 125 µs. The frame structure must contain information bits to identify the beginning of a frame (channel ch0) by 8 known framing bits. for binary modulation. 8 bits are employed for signalling between central ofﬁces (channel ch16).048 Mbit/s. we note. We give an example of implementation of the TDMA principle. thus making it a 7bit code word per sample. that is obtained by multiplexing 30 PCM coded speech signals (or channels) at 64 kbit/s. is called T1 carrier system and has a bit rate of 1. For example. and Japan the base group. at 2. and (b) TDMA.544 Mbit/s. The entire frame is formed of 24 Ð 8 C 1 D 193 bits. equivalent to 32Ð8 D 256 bits. Code division multiple access (CDMA): to each user is assigned a modulation scheme that employs one of the N0 orthogonal signals. Illustration of (a) FDMA. As shown in Figure 6. Canada. . 2.1 (Timedivision multiplexing) Timedivision multiplexing (TDM) is the interleaving of several digital messages into one digital message with a higher bit rate. each 8bit sample of each channel is inserted into a preassigned slot of a frame composed of 32 slots. analog to E1. as an example we illustrate the generation of the European base group. The remaining 30 channels are for the transmission of signals. the overall digital message has a bit rate of 256 bit/125 µs = 2. whose elements identify the modulation intervals.71.524 Chapter 6. Time division multiple access (TDMA): to each user is assigned one of the N S time sequences (slots). called E1. that of the 256 bits of the frame only 30 Ð 8 D 240 bits carry information related to signals. within a modulation interval each user then transmits the assigned orthogonal signal or the antipodal signal. however. preserving the orthogonality between modulated signals of the various users. obtained by multiplexing 24 PCM speech coded signals at 64 kbit/s.65. 3.64. In the United States.13. Example 6.048 Mbit/s. equal to the interval between two PCM samples of the same channel. for the case N0 D 8. to each user may be assigned one orthogonal signal of those given in Figure 6.
1965. 1999. M. . Simon. [2] S. Principles of digital transmission with wireless applications. New York: McGrawHill. Bibliography 525 Figure 6. 1990. Abramovitz and I. Jacobs. 38. Sept. A. IEEE Trans. New York: John Wiley & Sons. Handbook of mathematical functions. M. Stegun. 1995.65.6.048 Mbit/s. G. K. M.. pp. Biglieri. Benedetto and E. Divsalar. 1391–1403. eds. [5] M. Bibliography [1] J. New York: Kluwer Academic Publishers. 3rd ed. New York: Dover Publications. TDM in the European base group at 2. 1965. Digital communications. Shahshahani. and M. vol. “The performance of trelliscoded MDPSK with multiple symbol detection”. [3] J. on Communications. [4] D. Proakis. Principles of communication engineering. Wozencraft and I.
vol. [15] M. 2000. IEEE Trans. Shannon. Jan. 359–366.. 379–427 (Part I) and 623–656 (Part II). vol. IEEE Trans. F. Nov. [8] C. Modulation theory [6] R. [13] G. pp. 38. “Channel coding with multilevel/phase signals”. Trans. Europ. 1996. M. [12] G. on Information Theory. Englewood Cliffs. [11] G. and noise. pp. J. MA: Kluwer Academic Publishers. 6. Stein. Commun. modulation. vol. pp. .. 4th ed. E. Foschini and M. D. [9] G. Ziemer and W.S. IEEE Trans. L. “On limits of wireless communications in a fading environment when using multiple antennas”. New York: John Wiley & Sons. 1996. J. Cover and J. Mar. on Telecomm. 1991. 44. D. Thomas. 48. Piscataway: IEEE Press. G. “A mathematical theory of communication”. on Communications. R. 10. “Capacity of multi–antenna Gaussian channels”. Information theory and reliable communication.526 Chapter 6. Mar. and G. 1948. S. pp. 1992. 281–300. (N.). NJ: PrenticeHall. [19] W. June 1998.. Alouini. Bennett. Jr. 55–67. New York: McGrawHill. Wireless communications: principles and practice. on Information Theory. pp. Elements of information theory. 1995. 500–501. “Trellis shaping”. Gallager. H. [16] T. 27. pp. K. 311–335. on Information Theory. Ungerboeck. Gans.. Ungerboeck. Tranter. vol. vol. 1998. Rappaport. 1993. and S. vol. Forney. Abramson. Jr. ed. Telatar. 2384–2415. pp. on Information Theory. Oct. 1968. New York: John Wiley & Sons.–Dec. vol. Principles of mobile communication. vol. E. Norwell. [7] T. Forney. [18] Multiple access communications: foundations for emerging technologies. “Modulation and coding for linear Gaussian channels”. 1982. Wireless Person. IEEE Trans. [17] G. July 1970. [14] M. Simon and M. IEEE Trans. McGee. 1999. Principles of communications: systems. 585–595. W. “Another recursive method of computing the Qfunction”. 1966. Schwartz. Bell System Technical Journal. New York: John Wiley & Sons. 28. pp. 16. [10] E. “Exponentialtype bounds on the generalized Marcum Qfunction with application to error probability analysis over fading channels”. Stuber. [20] R. Communication systems and techniques.
b m/2 2¦ 2 (6.357) Q.b/ db (6.360) (6.a/ D pw . Gaussian distribution function and Marcum function 527 Appendix 6.359) Z 1C2 1 p a 2 pw .a/ (6.a/ D 1 8. deﬁned as Z C1 1 2 p e b =2 db (6.a/D 1 C erf p 2 2 Â Ã 1 a Q.358) b2 Z 2 C1 p a 2 1 p e 2³ db which are related to 8 and Q by the following equations Ä Â Ã½ a 1 8.A.356) 2³ 1 1 It is often convenient to use the complementary Gaussian distribution function.a/D erfc p 2 2 (6.a/ D D1 and the complementary error function erfc .A Gaussian distribution function and Marcum function 6.361) .b/ D p 2³ ¦ 1 .355) We deﬁne normalized Gaussian distribution (m D 0 and ¦ 2 D 1) as the function Z a Z a 1 2 8.6.A.a/ D 2³ a Two other functions that are widely used are the error function erf .1 The Q function The probability density function of a Gaussian variable w with mean m and variance ¦ 2 is given by e pw .b/ db D p e b =2 db (6.a/ D 1 erf .
3:5930. 6:8714. 1:3008. 1:0724.10 the values assumed by the complementary Gaussian distribution are given for values of the argument between 0 and 8.10 Complementary gaussian distribution. 3:0106. 1:0780. 2:5551. 1:3346. 03/ 03/ 03/ 03/ 04/ 04/ 04/ 04/ 04/ 04/ 04/ 05/ 05/ 05/ 05/ 05/ 06/ 06/ 06/ 06/ 06/ 07/ 07/ 07/ 07/ 08/ 08/ a 5:4 5:5 5:6 5:7 5:8 5:9 6:0 6:1 6:2 6:3 6:4 6:5 6:6 6:7 6:8 6:9 7:0 7:1 7:2 7:3 7:4 7:5 7:6 7:7 7:8 7:9 8:0 Q. 1:8175. 3:3693. Â Ã Â 2Ã 1 a 1 1 exp bound1 : Q 1 . 3:0954. 8:5399. 1:8990. 7:2348. 9:6800. 1:6983. 4:7918. 5:7901. a 0:0 0:1 0:2 0:3 0:4 0:5 0:6 0:7 0:8 0:9 1:0 1:1 1:2 1:3 1:4 1:5 1:6 1:7 1:8 1:9 2:0 2:1 2:2 2:3 2:4 2:5 2:6 Q. 2:1125. 8:0757. 01/ means 5:0000 ð 10 1 .a/ D exp 2 2 The Q function and the above bounds are illustrated in Figure 6. 2:8665. Modulation theory Table 6. 4:4565. 1:8658. 6:8033. 5:9904. 6:2378. 4:8096. 5:4125. 3:8209. 2:2750. 7:9333. . 2:8717.362) 2 2 a 2³a Â 2Ã a 1 bound2 : Q 2 . 1:5866. 1:4807. 1:7864. 1:0421.528 Chapter 6. 5:2310. 4:8342. 1:1507. 1:4882. 8:1975. 2:6001. 01/Ł 01/ 01/ 01/ 01/ 01/ 01/ 01/ 01/ 01/ 01/ 01/ 01/ 02/ 02/ 02/ 02/ 02/ 02/ 02/ 02/ 02/ 02/ 02/ 03/ 03/ 03/ a 2:7 2:8 2:9 3:0 3:1 3:2 3:3 3:4 3:5 3:6 3:7 3:8 3:9 4:0 4:1 4:2 4:3 4:4 4:5 4:6 4:7 4:8 4:9 5:0 5:1 5:2 5:3 Q. 3:0854.a/ D p (6.363) 2 2³a Â 2Ã a 1 (6. 3:1671. 1:2798. 1:8406. 3:3157. 4:6017.66. 4:2074. 3:1909. 9:6760.a/ D p exp (6. 2:7425. 08/ 08/ 08/ 09/ 09/ 09/ 10/ 10/ 10/ 10/ 11/ 11/ 11/ 11/ 12/ 12/ 12/ 13/ 13/ 13/ 14/ 14/ 14/ 15/ 15/ 15/ 16/ Ł Writing 5:0000. 2:3263. 1:5911. 4:6612. 2:8232. 3:3977. 9:9644.a/ 5:0000. 1:4388. 3:4458. 1:3945. 7:7688. We present below some bounds of the Q function. 6:2097. 5:4799. 1:0718. 6:8092. 2:0658. 1:3567. 9:8659. 6:6807.364) bound3 : Q 3 . In Table 6. 2:0558.a/ 3:4670. 5:3034. 6:2210. 1:3499. 1:3903. 2:1186.a/ 3:3320. 2:4196. 4:0160.
Gaussian distribution function and Marcum function 529 Figure 6.368) b2 2 (6. The Q function and relative bounds.a.a. 0/D1 Moreover.357). two particular cases follow: Q 1 . A useful approximation valid for b × 1.66. 6. From (6.6. b/ D b (6. a × 1. is given by a/ (6.b a/2 2 b>a>0 (6.0. b/ We also give the Simon bound [14] e . b × b 1 C Q 1 .b Ä Q 1 . a/ ' 2Q. for b × 1 and b × b a the following approximation holds Q 1 . b/De Q 1 .369) (6. b/ ' Q.b.a.A. b/ Ä e .b a/ a > 0.a.365).a.365) where I0 is the modiﬁed Bessel function of the ﬁrst type and order zero.2 The Marcum function x 2 Ca 2 2 I0 .367) where the Q function is given by (6.366) (6.bCa/2 2 Q 1 . deﬁned in (4.ax/ dx We deﬁne the ﬁrstorder Marcum function as Z C1 xe Q 1 .216).A.370) .
370) the upper bound is very tight.a b/2 2 e . . and the lower bound for a given value of b becomes looser as a increases. In (6.530 Chapter 6.371) the lower bound is very tight. Modulation theory and 1 " 1 e 2 .a.371) We observe that in (6. A recursive method for computing the Marcum function is given in [19].aCb/2 2 # Ä Q 1 . b/ a>b½0 (6.
By induction it is just as easy to prove that two adjacent words in each list differ by one bit at most.375) and appending a 1 in front. the ﬁrst 4 words are obtained by repeating the list (6.372): 0 0 0 1 (6.B Gray coding In this appendix we give the procedure to construct a list of 2n binary words of n bits.372) and appending a 1 in front: 1 1 1 0 The ﬁnal result is the following list of words: 0 0 1 1 0 1 1 0 (6.374) (6. the ﬁnal result is the list of 8 words 0 0 0 0 1 1 1 1 0 0 1 1 1 1 0 0 0 1 1 0 0 1 1 0 (6. Gray coding 531 Appendix 6. where adjacent words differ in only one bit.372) The list for n D 2 is constructed by considering ﬁrst the list of .376) It is easy to extend this procedure to any value of n.375) and appending a 0 in front of the words of the list. .1=2/22 D 2 words that are obtained by appending a 0 in front of the words of the list (6.6. Inverting then the order of the list (6. The case for n D 1 is immediate.373) The remaining two words are obtained by inverting the order of the words in (6.375) Iterating the procedure for n D 3. We have two words with two possible values 0 1 (6.B.
67.69. Modulation theory Appendix 6. that is a multiple of a minimum time duration equal to T =M. In PDM. two other baseband pulse modulation techniques are pulse position modulation (PPM) and pulse duration modulation (PDM).380) (6. In practice the received pulse. : : : . PPM consists of a set of pulses whose shift.t/ D g0 Â Ã t n n2A (6.378) 1g (6. Fundamental pulse shape of PPM. which are disturbed by noise.532 Chapter 6. has a rise time t R different from g (t) 0 0 T/M T t Figure 6. : : : .t/ D g0 t T n M Ã n2A (6.68.67. M The transmitted isolated pulse is Â sn . .379) where g0 is given in Figure 6.377) For M D 4 the set of waveforms is represented in Figure 6. 2. We consider the fundamental pulse shape of Figure 6.C Baseband PPM and PDM In addition to the widely known PAM.67 and an alphabet given by A D f0. instead. the input symbol determines the duration of the transmitted pulse. 2. with amplitude equal to A. For an alphabet A D f1. 1. The set of PDM waveforms for M D 4 is illustrated in Figure 6. Signaltonoise ratio In both PPM and PDM the information lies in the position of the fronts of the transmitted pulses: therefore demodulation consists in ﬁnding the fronts of the pulses. If the channel bandwidth were inﬁnite. depends on the value of the transmitted symbol. 3. Mg the transmitted isolated pulse is given by sn . one could receive perfectly rectangular pulses. with respect to a given time reference.
C. n =1 T/4 t n=2 2T/4 t n=3 3T/4 t n =4 0 T t Figure 6. .68.6. Baseband PPM and PDM 533 n =1 T/4 t n =2 2T/4 t n =3 3T/4 t n =4 0 T t Figure 6. PDM waveforms for M D 4.69. PPM waveforms for M D 4.
382) We consider the following approximated expression that links the rise time to the bandwidth of the pulse tR ' 1 2B (6.383) Substitution of the above result in (6.ti /.2T /. The detection of the front of a pulse is obtained by establishing the instant ti in which the received signal.ti / is related to the noise w. and noise disturbs the reception of the pulse. that is 0D 2E sCh N0 (6. as illustrated in Figure 6. as the channel has a ﬁnite bandwidth B. The error ". and rise time t R of the received pulse: w.382) yields E[" 2 ] ' N0 4A2 B (6. pulse plus noise. zero.70.384) On the other hand.534 Chapter 6. crosses a given threshold. the meansquare error is given by E[" ] D 2 Â tR A Ã2 E[w ] D 2 Â tR A Ã2 N0 B (6. assuming the average duration of the pulses is −0 . PPM and PDM demodulation in the presence of noise. Modulation theory Figure 6.ti / D tR A (6.381) Assuming the noise stationary with PSD N0 =2 over the channel passband with bandwidth B.70.385) .105) with Bmin D 1=. amplitude A.ti / ". the signaltonoise ratio is given by (6.
387)) is illustrated.384) yields E[" 2 ] D −0 1 2B 0 (6. Baseband PPM and PDM 535 where E sCh D −0 A2 Finally. substitution of (6.C.386) For a more indepth analysis we refer to [20].385) in (6. where a tradeoff between the channel bandwidth and the signaltonoise ratio at the decision point (see (6.387) (6. .6.
with values f 1.389) 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 8Tc 8Tc 8Tc 8Tc 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 8Tc 8Tc 8Tc 8Tc Figure 6. with binary elements from the set f0. We consider 2m ð 2m Hadamard matrices Am .536 Chapter 6. 1g.71. Modulation theory Appendix 6.388) (6. 1g. of length 2m . Eight orthogonal signals obtained from the Walsh code of length 8.D Walsh codes We illustrate a procedure to obtain orthogonal binary sequences. . we have A0D[0] Ä ½ 0 0 A1D 0 1 (6. For the ﬁrst orders.
71 shows the 8 signals obtained with the Walsh code of length 8: the signals are obtained by interpolating the Walsh code sequences by a ﬁlter having impulse response wTc .392) N where Am denotes the matrix that is obtained by taking the 1’s complement of the elements of Am .6.391) Ä AmC1 D Am Am N Am Am ½ (6.393) . Walsh codes 2 3 0 17 7 15 0 0 1 1 0 0 1 1 0 0 0 0 0 1 1 1 1 0 1 0 1 1 0 1 0 0 0 1 1 1 1 0 0 0 1 1 0 1 0 0 1 3 7 7 7 7 7 7 7 7 7 7 5 537 0 60 A2D6 40 0 2 0 60 6 60 6 60 A3D6 60 6 60 6 40 0 In general the construction is recursive 0 1 0 1 0 1 0 1 0 1 0 1 0 0 1 1 0 0 1 1 0 0 1 1 (6. From the construction of Hadamard matrices. Figure 6.t/ D rect t Tc =2 Tc (6.D. A Walsh code of length 2m is obtained by taking the rows (or columns) of the Hadamard matrix Am and by mapping 0 into 1.390) (6. it is easily seen that two words of a Walsh code are orthogonal.
.
which is associated with the message fb` g.1. or in other words the sequence of symbols fak g is not obtained by applying channel coding.2) Tb Transmitter Bit mapper.1) Usually fb` g is a sequence of i. 3g.i. To select the values of ak we consider pairs of input bits and map them into quaternary symbols as indicated in Table 7. 7. symbols. Note that bits are mapped into symbols without introducing redundancy. 1. and 3) line coding. We will also consider the effects of errors introduced by the digital transmission on a PCM encoded message (see Chapter 5) [3]. so that errors can be detected and/or corrected at the receiver (see Chapters 11 and 12). Their objectives are respectively: 1) “compress” the digital message by lowering the bit rate without losing the original signal information (see Chapter 5). that are emitted every Tb seconds: fb` g D f: : : . b0 . b 1 .Chapter 7 Transmission over dispersive channels In this chapter we will reconsider amplitude modulation (PAM and QAM. 2) increase the “reliability” of the transmission by inserting redundancy in the transmitted message. The signals at various points of a ternary PAM system are shown in Figure 7. 2]. 2) channel coding. .1 This situation will be maintained throughout the chapter.2) 1 Rb D (bit/s) (7. ak 2 A D f 3. not explicitly indicated in Figure 7. We assume that a source.1 Baseband digital transmission (PAM systems) Let us brieﬂy examine the fundamental blocks of the baseband digital transmission system illustrated in Figure 7. is equal to (see Section 6. 1g. taking into account the possibility that the transmission channel may distort the transmitted signal [1.1. Source. 1 We distinguish three types of coding: 1) source or entropy coding. b2 . and 3) “shape” the spectrum of the transmitted signal (see Appendix 7. therefore we speak of uncoded transmission.1.A). see Chapter 6) for continuous transmission. generates a message fb` g composed of a sequence of binary symbols b` 2 f0. The system bit rate. The bit mapper uses a onetoone map to match a multilevel symbol to an input bit pattern. b1 . symbols fak g from a quaternary alphabet.2. 1. Let us consider. : : : g (7. for example.d.
if the values of ak belong to an alphabet A with M elements. 1=T is the modulation rate or symbol rate of the system and is measured in Baud: it indicates the number of symbols per second that are transmitted. Signals at various points of a ternary PAM transmission system with alphabet A D f 1.3) T log2 M Tb . 1g. Transmission over dispersive channels Figure 7. In general. 0. then 1 1 1 D (Baud) (7. Figure 7.540 Chapter 7. For uncoded quaternary transmission the symbol period or modulation interval T is given by T D 2Tb .1.2. Block diagram of a baseband digital transmission system.
M 1/.t kT / (7.109). Mg.t/ then we have sCh .7. 2.7) ak qCh .t/ that is input to the transmission channel is given by s.t/ D C1 X kD 1 ak h T x . that is ak 2 A D f . that is ak 2 f1.8) The transmission channel introduces an effective noise w. 1.9) .6) (7. : : : .t/ D gCh Ł s. 1. .t/ D h T x Ł gCh . Now the values of ak are associated with fÞn g.t/ C w. Therefore the desired signal at the output of the transmission channel still has a PAM structure. : : : . Modulator. M.M 1/g.t/ (7.t/ we deﬁne qCh . : : : .1 Example of quaternary bit map.t kT / (7.t/ D sCh . From the relation sCh . For a PAM system. see (6. Therefore the signal at the input of the receive ﬁlter is given by: r.3 we considered an alphabet whose elements were indices.t/ D C1 X kD 1 (7.t kT / (7. : : : . the modulator associates the symbol ak with the amplitude of a given pulse h T x : ak ! ak h T x . n D 1.4) Therefore the modulated signal s.5) Transmission channel The transmission channel is assumed to be linear and time invariant. Baseband digital transmission (PAM systems) 541 Table 7.1. with impulse response gCh . b2k 0 1 1 0 b2kC1 0 0 1 1 ak 3 1 1 3 We note that in Section 6.
in the absence of noise. 1g is given in Figure 7.t0 C kT / D ak q R .t/ then s R . the detected binary information O O message fb` g is obtained. using an inverse bit mapper. Then the desired signal is given by: s R . 2. .12) where h 0 D q R .t/ where w R .542 Chapter 7. 0.t/ D In the presence of noise.t0 / D ak h 0 (7.3. Transmission over dispersive channels Receiver The receiver consists of three functional blocks: 1. From the sequence frk g we detect the transmitted sequence fak g. Sampler.t/ D qCh Ł g Rc .t/ D s R . From the sequence fak g.t/ Let the overall impulse response of the system be q R . the sample index k. r R . The simplest structure is the instantaneous nonlinear threshold detector: ak D Q[rk ] O (7. 2 To simplify the notation.t kT / (7.t/ C w R .13) C1 X kD 1 (7. and its choice is fundamental for system performance.t/ D w Ł g Rc .14) (7. An example of quantizer characteristic for A D f 1. here appears as a subscript.t/. then the various pulses do not overlap and.10) (7. 3.15) O where Q[rk ] is the quantizer characteristic with rk 2 < and ak 2 A. This block is assumed linear and time invariant with impulse response g Rc . The parameter t0 is called timing phase.11) ak q R . Threshold detector. If the duration of q R . Ampliﬁerequalizer ﬁlter. sampling at instants t0 CkT yields:2 rk D r R .t/ D g Rc Ł sCh .t0 / is the amplitude of the overall impulse response at the sampling instant t0 .t/ D h T x Ł gCh Ł g Rc . alphabet of ak . associated with the instant t0 C kT .t/ is conﬁned to a modulation interval.
f / (7. Baseband digital transmission (PAM systems) 543 ^ = Q[r ] ak k h0 2 0 −1 1 h0 2 rk Figure 7.9. s R . sCh .4 it is important to verify that by ﬁltering s. In other words. then the bandwidth B of the transmitted signal is equal to that of h T x .398)): þ2 þ þ þ1 N Ps . From the spectral analysis (see Example 1. a PAM signal s.t/. ak T Figure 7. f / is periodic of period 1=T .16) where q.t/. Characteristic of a threshold detector for ternary symbols with alphabet A D f 1. t 2 <. The PAM signal as output of an interpolator ﬁlter. As Pa . From Figure 7. We recall that the receiver structure described above was optimized in Chapter 6 for an ideal AWGN channel.t/ may be regarded as a signal generated by an interpolator ﬁlter with impulse response q.t/ we obtain a signal that is still PAM. The spectral density of a ﬁltered PAM signal is obtained by multiplying Pa .t kT / (7.t/ D C1 X kD 1 ak q.1.3. have the following structure: s. 1g. f /þ Pa .4. as shown in Figure 7. f / in (7.7. and s. q s(t) . 0.4.t/.t/ is the impulse response of a suitable ﬁlter. with a pulse given by the convolution of the ﬁlter impulse responses. and amplitude h0 of the overall impulse response.t/.17) by the squared magnitude of the ﬁlter frequency response. Power spectral density of a PAM signal The PAM signals found in various points of the system. f / D þ Q.9 on page 69) we know that s is a cyclostationary process with average power spectral density given by (see (1.17) þ þT where Pa is the spectral density of the message and Q is the Fourier transform of q.
544 Chapter 7. see (7. The mean value and the statistical power of the sequence are given by X X Þp. Transmission over dispersive channels Occasionally the passband version of PAM that is obtained by DSB modulation is also proposed.18) (7. that describes the decomposition of the PSD of a message into ordinary and impulse functions.Þ/.i. f / is nonzero at frequencies multiple of 1=T .22) ` T We note that spectral lines occur in the PSD of s if ma 6D 0 and Q.5. f /þ ¦a T D jQ. We refer the reader to Appendix 7.c/ N s .20) P þT þ T and þ2 þ C1 X Â þ þ . f / D þ 1 Q. symbols with values from the alphabet A D fÞ1 . Following Example 1. Þ2 .1 (PSD of an i. The bit mapper uses a map to associate a complexvalued symbol ak to an input bit pattern. the spectrum of the transmitted signal is shaped by introducing correlation among transmitted symbols by a line encoder. In some applications.Þ/ Ma D jÞj2 p.19) 2 Consequently ¦a D Ma jma j2 .7. Passband PAM. and by assigning equal probabilities to antipodal values.2 Passband digital transmission (QAM systems) We consider the QAM transmission system illustrated in Figure 7.d. f /j2 ¦a (7. Example 7.1.C.d/ N s .A for a description of the more common line codes. Þ M g and p. be the probability distribution of each symbol. Typically the presence of spectral lines is not desirable for a transmission scheme. together with the associated parameters. f /þ jma j2 P Ž f þ þT `D 1 C1 þ þ m þ2 X þ Â ` Ãþ2 Â þ þ aþ þ Ž f þQ Dþ þ þ T `D 1 T þ ` T Ã (7. f / D þ 1 Q.21) Ã (7. Transmitter Bit mapper. We obtain ma D 0 by choosing an alphabet with symmetric values with respect to zero. the decomposition of the PSD of s is given by þ þ2 2 þ þ 2 .6 are given two examples of constellations and corresponding . is discussed in Appendix 7. : : : .d.1 on page 51.Þ/ ma D Þ2A Þ2A (7.i. 7. Þ 2 A. symbol sequence) Let fak g be a sequence of i. In Figure 7.17): in this case the sequence has memory.
7. Two constellations and corresponding bit map.t/ D D C1 X kD 1 C1 X kD 1 ak h T x .I D Re [ak ] and ak.I 1 ak.Q D Im [ak ].I h T x .Q (7.bb/ .t kT / kT / C j C1 X kD 1 (7.Q (11) (01) (10) (00) 3 (1011) 1 (1111) 1 (0111) 3 a (0011) k.2. Passband digital transmission (QAM systems) 545 Figure 7. Modulator. binary representations. 4PSK (or QPSK) symbols are taken from an alphabet with four elements. Figure 7.t ak.Q h T x .24) ak.Q (1000) (1100) (0100) (0000) 3 (1001) (1101) (0101) (0001) 1 ak.t kT / . however.I (1010) (1110) (0110) (0010) 3 (a) QPSK.5.I C jak. the baseband modulated signal is complexvalued: s . Block diagram of a passband digital transmission system.23) and ak. ak. Similarly each element in a 16QAM constellation is uniquely identiﬁed by four bits. (b) 16QAM. each identiﬁed by two bits. where ak D ak. Typically the pulse h T x is realvalued.6.
the carrier frequency (radian frequency) and phase. respectively.C/ .t/ D 2Refs .bb/ to s is illustrated in Figure 7. We deﬁne s . Fourier transforms of baseband signal and modulated signal.27) . s .395).25) then the realvalued transmitted signal is given by: s. f 2 f 0 /e j'0 (7.26) The transformation in the frequency domain from s . Power spectral density of a QAM signal From the analysis leading to (1.t/g F ! S.7.C/ .C/ .bb/ is a cyclostationary process of period T with an average PSD (see (1.bb/ .7.!0 tC'0 / 2 F ! S .bb/ . f / D þ HT x .C/Ł . f / D S .t/e j .t/ D 1 s . f/ (7.398)): þ þ2 þ1 þ N Ps . f /þ Pa .bb/ . f / þT þ (7. Transmission over dispersive channels S (bb)(f) B= 1 (1+ρ ) 2T 0 B S (+) (f) f B 0 S (f) f0 B f0 f 0 +B f f0 B f 0 f 0+B 0 f0 B f0 f 0 +B f Figure 7. Let f 0 (!0 D 2³ f 0 ) and '0 be.546 Chapter 7. f / C S . f / D 1 S .C/ .
!0 t C '0 / (7. We ﬁrst consider the situation where the condition rs .t. QAM transmitter: complexvalued representation.bb/ is in general a complexvalued signal.28) needs clariﬁcation.t/ D Re[s . and in particular the case where rs .bb/ .bb/Ł . circularly symmetric symbols (see (1. for 1=T − f 0 it happens that the autocorrelation terms approximate the same terms found in the previous case.8.t/e j . an implementation based on this representation requires a processor capable of complex arithmetic. 2.bb/Ł . we get that s is a cyclostationary random process with an average PSD given by3 N N Ps .29) is shown in Figure 7. 3 The result (7.24).bb/Ł .30) Figure 7.25) and (7.29) The blockdiagram representation of (7.bb/ and rs . f 4 N f 0 / C Ps . From (7.bb/ s .2.304)) that relates rs to rs . in the equation similar to (1. and the crosscorrelation terms become negligible. In this situation the crosscorrelations.bb/ s . t − / D 0 is satisﬁed. t − / 6D 0. such that T p is an integer multiple of both T and 1= f 0 .bb/ .304). as the crosscorrelations are zero.!0 t C '0 / C j sin.27) and (7. then s is cyclostationary in t of period equal to T p . Taking the average correlation in a period T . t − / in Fourier series (in the variable t). and expanding rs .28) We note that the bandwidth B of the transmitted signal is equal to twice the bandwidth of h T x . as for example in the case of QAM with i. using (7. t − / is a periodic function in t of period T .bb/ s . From the equation (similar to (1. Taking the average correlation over the period T p .8.C).t. As s . f f 0 /] (7.!0 tC'0 / D cos.bb/ . it turns out s. . If a real value T p exists.!0 tC'0 / kD 1 # kT / (7. as for example in the case of PAMDSB (see Appendix 7.bb/Ł . we ﬁnd that the process s is cyclostationary in t of period T . We now consider the situation where rs .t D Re e j . do not vanish. Passband digital transmission (QAM systems) 547 Moreover.bb/Ł .bb/ s .d. starting from a relation similar to (1.407)).t.7. f / D 1 [Ps .!0 tC'0 / ] " C1 X ak h T x .26).t. As e j .i.304). Three equivalent representations of the modulator 1.bb/ s .28) follow. the results (7.
f /G Rc . f / D SCh . which follows the scheme of Figure 6.31) is shown in Figure 7. f / (7. f / (7.548 Chapter 7.t kT / " D Re C1 X kD 1 kD 1 C1 X kD 1 jak je j .!0 t C '0 C Âk /h T x . f / D S M0 . Coherent receiver In the absence of noise the general scheme of a coherent receiver is shown in Figure 7. .I h T x .D.t kT / kT / (7. where the information bits select only the value of the carrier phase.t/e j . the received signal is translated to baseband by a frequency shift.34) then it is ﬁltered by a lowpass ﬁlter (LPF). s M0 .t/ F ! S R .t/ D cos.31) The blockdiagram representation of (7. (7.127).t If jak j is a constant we obtain the PSK signal (6. f C f 0 /e j'1 (7.9.t kT / (7. 3.!0 t C '0 / C1 X kD 1 ak.t/ D s Ł gCh .31) is discussed in Appendix 7.!0 t C '0 / C1 X kD 1 ak. f / D S.29) becomes: s.29) takes the form: " # C1 X s.t/ F ! SCh .33) First.35) Figure 7.!0 tC'0 / jak je jÂk h T x .!0 tC'1 / F ! S M0 .9.t/ D sCh . QAM receiver: complexvalued representation. The received signal is given by: sCh .38).5 (see also Figure 6. The implementation of a QAM transmitter based on (7.!0 tC'0 CÂk / # h T x .t/ D s M0 Ł g Rc .32) D jak j cos. s R .t/ D Re e j . f /GCh .40.t kT / sin.Q h T x . Transmission over dispersive channels (7. Using the polar notation ak D jak je jÂk . g Rc .
10 illustrates these transformations.t/ D . Figure 7. This is the same as assuming as reference carrier e j .bb/ and r .2³ f 0 tC'0 / . Frequency responses of the channel and of signals at various points of the receiver.11 are deﬁned apart from the term e j'0 . then the receiver in Figure 7. . We note that in the above analysis.7. therefore the signals s .3 Baseband equivalent model of a QAM system Recalling the relations of Figure 1. then s R . the receive carrier phase '1 may be different from the transmit carrier phase '0 . 7.1=2/sCh .11:4 by assuming that the transmit and receive carriers have the same We note that the term e j'0 has been moved to the receiver.5.30.t/. if g Rc is a nondistorting ideal ﬁlter with unit gain.10. 4 . we illustrate the baseband equivalent scheme with reference to Figure 7.9 is simpliﬁed into that of Figure 7. In the particular case where g Rc is a realvalued ﬁlter.3.bb/ of Figure 7.bb/ We note that. as the channel may introduce a phase offset. Baseband equivalent model of a QAM system 549 G (f) Ch f0 S (f) Ch f −2f0 1 f0 −B f0 G (f) Rc f0 +B f SMo(f) −2f0 SR (f) f −2f0 f Figure 7.
7.bb/ .37) To simplify the analysis.'1 '0 / (7.'1 '0 / appears in Figure 7.1. f / D e j .'1 '0 / p 2 2 . Transmission over dispersive channels Figure 7.3.bb/ . we can study QAM systems by the same method that we have developed for PAM systems.bb/ D ¦w 2 (7. f /. f / < GC . that is w I ? w Q . Pw I w Q . frequency. hence 2 2 2 2 ¦w I D ¦w Q D 1 ¦w.11. We will include the factor e j . Therefore the scheme of Figure 7. f / D 2Pw . for the study of a QAM system we will adopt the PAM model of Figure 7.550 Chapter 7. and the factor 1= 2 in the impulse response g Rc . f / D Pw. f C f 0 / for f ½ f 0 .bb/ gCh . we recall here that if for f > 0 the spectral density of w. Consequently the additive noise has a spectral density equal to . f C f 0 / f ½ f0 1 (7. is an even function around the frequency f 0 .11. We note that p the factor . f / D 0.1=2/e j .1 holds also for QAM in the presence of additive noise: the only difference is that in the case of a QAM system the noise is complexvalued with orthogonal inphase and quadrature components.4 for an analysis of passband signals. each having spectral density Pw .'1 '0 / = 2 p in the impulse response of the transmission channel gCh .12 is a reference scheme for both PAM and QAM.1=2/Pw.t/ have a power spectral density that is given by ( 2Pw . where 8 for PAM > GCh . f / D 2 0 elsewhere Moreover.1 Signal analysis We refer to Section 1. f C f 0 /1.38) > : GCh . Baseband equivalent model of a QAM transmission system. assuming that all signals and ﬁlters are in general complex. 7.t/ D w I .t/ (7. f / D Pw Q . f C f 0 / for f ½ f 0 .39) . Pw .bb/ . Hence the scheme of Figure 7.36) Pw I . then the real and imaginary parts of w. f C f 0 / for QAM p 2 We note that for QAM we have gC .t/ C jw Q .t/ D e j .
f / D and PwC . to simplify the notation. sC . fak g. wC .t/.7. f / D N0 (V2 /Hz) In the model of PAM systems. the symbols of the sequence fak g assume real values. In PAM systems. f / D Pw0Q . additive Gaussian noise. 1. With reference to the scheme of Figure 7. sequence of symbols with values from a complexvalued alphabet A. only the component w0I is considered.42) 4. Baseband equivalent model of a QAM system 551 Figure 7.t/ is in fact s .t/ D h T x Ł gC .43) (7. Baseband equivalent model of PAM and QAM transmission systems.t/ D 1 p g Rc . this condition can be omitted.44) We point out that for QAM s.t/ D C1 X kD 1 C1 X kD 1 ak h T x .t/ D 3.3.41) ak qC . complexvalued. Signal at the channel output. In the case of white noise it is:6 Pw0I . the ﬁlter g Rc will be indicated in many passband schemes simply as g Rc . We summarize the deﬁnitions of the various signals in QAM systems.t/ C jw0Q .t/ D w0I .9. with spectral density PwC .t/ (7.t kT / (7. .t/ .12. the relation between the impulse responses of the receive ﬁlters is given by g Rc .43) should include the condition f > f 0 . 5 6 N0 (V2 /Hz) 2 (7. Circularlysymmetric. 2. Modulated signal.40) In the following.t kT / qC .t/ 2 (7.5 s. Sequence of input symbols. Because the bandwidth of g Rc is smaller than f 0 .bb/ . In fact (7.
we get qCh D qC .105).t/ and w R .t/ where s R . r R . sCh .48) C1 X kD 1 (7. we have 0D We express now E sCh 2 E[sCh .552 Chapter 7. For an i.i.50) (7.53) qCh D can be (7.t/ D sC .49) (7. that we recall here.t/ D with q R .46) ak q R . Signal at the decision point at instant t0 C kT . Because for PAM. Signal at the output of the complexvalued ampliﬁerequalizer ﬁlter g Rc .t/ D s R .t/ 6.399) we have E sCh D Ma E qCh and.51) PAM systems.38).t/ C w R .d. we obtain Ma E qCh N0 =2 where Ma is the statistical power of the data and E qCh is the energy of the pulse h T x Ł gCh .1 and 7. input symbol sequence. Transmission over dispersive channels 5. and assuming the noise w white with Pw . fak g O (7. rC . Received or observed signal. g Rc is a realvalued ﬁlter.47) Signaltonoise ratio The performance of a system is expressed as a function of the signaltonoise ratio 0 deﬁned in (6. using (1.t kT / (7.N0 =2/2Bmin N0 Bmin N0 .45) (7. In general.53) expressed as Ma E qC 0D N0 =2 0D (7. having minimum bandwidth Bmin . Sequence of detected symbols.Bmin T / in the cases of PAM and QAM systems.54) (7.t/ C wC .52) .t/ D wC Ł g Rc . for Bmin D 1=.t/ D qC Ł g Rc . observing (7. then (7.5.2T /.t/] MsCh E sCh D D . 7.t/ In PAM systems. with reference to the schemes of Figures 7. yk D r R . (7. f / D N0 =2. for a channel output signal.t0 C kT / 8.
Baseband equivalent model of a QAM system 553 QAM systems.Re[sC . or a QAM system may be used. h T x .51) becomes 0D We note that (7. f 2 ). with wide spectrum. .t/] D 1 E[jsCh . the channel is bandlimited with a ﬁnite bandwidth f 2 f 1 .3.t/ with longer duration and smaller bandwidth.14.t/]/2 ] and T E[. for transmission over radio. 7 The term Ma E qC =2 represents the energy of both Re[sC .38). 7. Transmitter The choice of the transmit pulse is quite important because it determines the bandwidth of the system (see (7. coincides with (7.t/]/2 ].295). f / is as represented in Figure 7. f 1 may be of the order of a few hundred Hertz.13. equal to N0 =2. h T x .3.2 Characterization of system elements We consider some characteristics of the signals in the scheme of Figure 7.t/j2 ] D E[jsC . and the PSD of the noise components. f 1 may be in the range of MHz or GHz.t/]. Then (7. assuming that sC .53) of PAM systems.55) Hence.t/] and Im[sC .Im[sC . Two choices are shown in Figure 7.t/ D rect T . In any case. For transmission over cables.bb/ 2 E[sCh . From (1. as Bmin D 1=T . Therefore it is represented by a ﬁlter having impulse response gCh .56) represents the ratio between the energy per component of sC . expressed as7 0D Ma E qC =2 N0 =2 (7.17) and (7.t/ D wT .56). assuming as carrier frequency f 0 the center frequency of the passband ( f 1 . Consequently. whereas for radio links. instead.56). 2. As described in Chapter 4.t/j2 ] 2 (7.407)). we obtain .57) Ma E qC N0 (7. (7. the majority of channels are characterized by frequency responses having a null at DC. a PAM signal needs to be translated in frequency (PAMDSB or PAMSSB). where Â Ã t T 2 1. PAM (possibly using a line code) as well as QAM transmission systems may be considered over cables.t/ is circularly symmetric (see (1.38). using (7. observing also (7. Transmission channel The transmission channel is modelled as a time invariant linear system. where the passband goes from f 1 to f 2 .12. Therefore the shape of the frequency response GCh . given by T E[.28)).7.
t/. An example of frequency response of a radio channel is given in Figure 4.13. f / (7.60) are too stringent. for channels encountered in practice conditions (7. According to (1. f /j D G0 for j f j < B (7. known as Heaviside conditions for the absence of distortion. .79).554 Chapter 7. that is: sC .59) 2. f / D 2³ f t0 for j f j < B (7. In short. a channel presents ideal characteristics.t/ may be very different from s.59) and (7.60) Under these conditions.t t0 / (7. the phase response is proportional to f for j f j < B. f /je j arg GC .32: the overall effect is that the signal sC . Transmission over dispersive channels Figure 7. s is reproduced at the output of the channel without distortion.144). we adopt the polar notation for GC : GC . f / D jGC . channels introduce both “amplitude distortion” and “phase distortion”. Two examples of transmit pulse hTx . for PAM and QAM transmission systems we will refer instead to the Nyquist criterion (7. With reference to the general model of Figure 7. arg GC .t/ D G0 s. jGC . if the following two properties are satisﬁed: 1.58) Let B be the bandwidth of s.12. the magnitude response is a constant for j f j < B.t/.61) In practice.
f / D G M . linear distortion and additive noise.16. f /C. and a data detector.12. Frequency response of the transmission channel.62) where G M . In practice. Baseband equivalent model of a QAM system 555 Figure 7. if the frequency response of the receive ﬁlter G Rc .3.15. consisting of a ﬁlter g Rc followed by a sampler with sampling rate 1=T .7.14.e j2³ f T /. f / is a generic function. as illustrated in Figure 7. . in the system of Figure 7. In general.15 yk should be equal to ak . f / contains a factor C. such that the following factorization holds: G Rc . the only disturbances considered here. where the sampler is followed by a discretetime ﬁlter. periodic of period 1=T . Ideally.e j2³ f T / (7. then the ﬁltersampler block before the data detector of Figure 7.12 can be represented as in Figure 7.t/ and yk is the same. It is easy to prove that in the two systems the relation between rC . Receiver We return to the receiver structure of Figure 7. may determine a signiﬁcant deviation of yk from the desired symbol ak .
in the presence of noise and linear distortion.3 Intersymbol interference Discretetime equivalent system From (7.k D w R .17.Q 0 0 y y −1 −1 −2 −3 −2 −4 −3 −3 −2 −1 y k.t0 C kT / D and w R.556 Chapter 7. 7.k D s R .I 0 1 2 3 4 5 (a) QPSK. from (7.49).I 0 1 2 3 −5 −5 −4 −3 −2 −1 y k. One of the simplest data detectors is the threshold detector.65) (7.k C w R:k (7. that associates with each value of yk a possible value of ak in the constellation.63) . at the decision point the generic sample is expressed as yk D s R. Values of yk D yk.3. (b) 16QAM.I C jyk.Q . the decision regions for a QPSK system and a 16QAM system are illustrated in Figure 7. Using the rule of deciding for the symbol closest to the sample yk .64) C1 X iD 1 ai q R . The last element in the receiver is the data detector.16.k i/T / (7.15.46) we deﬁne s R.t0 C . 3 5 4 2 3 2 1 1 k.Q k.t0 C kT / Then. Receiver structure with analog and discretetime ﬁlters. Figure 7. Transmission over dispersive channels Figure 7.
t0 C t/ and deﬁning h i D h.k (7. (b) 16QAM. which behaves as a disturbance with respect to the desired term ak h 0 .k D where ik D C1 X i D 1.69) represents the intersymbol interference (ISI).66) (7. the more valid . Moreover (7. i 6Dk C1 X iD 1 (7. The coefﬁcients fh i gi 6D0 are called interferers.3.65) becomes yk D ak h 0 C ik C w R.Q 3 1 yk. For the analysis. even in the absence of noise.70) We observe that.I 3 1 3 yk. timeshifted by t0 .67) ai h k i D ak h 0 C ik (7.17. Decision regions for a QPSK system and a 16QAM system. Baseband equivalent model of a QAM system 557 yk. it is often convenient to approximate ik as noise with a Gaussian distribution: the more numerous and similar in amplitude are the interferers.Q 3 1 1 yk. Introducing the version of q R .I (a) QPSK.7. Figure 7.i T / D q R . the detection of ak from yk by a threshold detector takes place in the presence of the term ik .t0 C i T / it follows that s R. as h.68) ai h k i D ÐÐÐ C h 1 akC1 C h 1 ak 1 C h 2 ak 2 C ÐÐÐ (7.t/ D q R .
the ﬁrst two moments of ik are easily determined. we derive the discretetime equivalent scheme.78) 8 See Observation 1.77) In the case of PAM (QAM) transmission over a channel with white noise.I m[w R. Discretetime equivalent scheme.k D ¦w R D PwC . is this approximation.65). called overall discretetime equivalent impulse response of the system. Concerning the additive noise fw R. with fs R.18).k is equal to that of w R and is given by Z C1 2 2 ¦w R. f / D C1 X `D 1 (7. f /jG Rc . (7.k g is given by Pw R.71) i D 1. i 6D0 Variance of ik : jh i j2 (7.d.74) Pw R Â f ` 1 T In any case. the variance per dimension of the noise is given by PAM QAM 2 ¦ I2 D E[w 2 ] D ¦w R R. f /jG Rc .558 Chapter 7. that relates the signal at the decision point to the data transmitted over a discretetime channel with impulse response given by the sequence fh i g. symbols. where PwC .i. the variance of w R. f / D N0 =2 (N0 ). Transmission over dispersive channels Figure 7.k 2 ¦ I2 D E[.76) (7. i 6D0 2 2 ¦i D ¦a C1 X i D 1. In the case of i.75) yields a variance per dimension equal to ¦ I2 D N0 E g Rc 2 (7.Re[w R.73) Ã (7.68). with period T (see Figure 7. f / D PwC . .8 being Pw R .18. C1 X Mean value of ik : m i D ma hi (7.k . with period T.75) In particular.6 on page 62.k g given by (7. of a QAM system.72) From (7.k g. f /j2 the PSD of fw R.k ]/2 ] D 1 ¦w R 2 (7. f /j2 d f 1 (7.k ]/2 ] D E[.
19a.t/ that satisﬁes the conditions (7. The solution is the Nyquist criterion for the absence of distortion in digital transmission. We observe that (7. C1 X iD 1 hi e j2³ f i T D Â C1 1 X H f T `D 1 ` T Ã (7. is called Nyquist frequency. which coincides with half of the modulation frequency. Nyquist pulses The problem we wish to address consists in ﬁnding the conditions on the various ﬁlters of the system.2T /.90). hence the condition for the absence of ISI is formulated in the frequency domain for the generic pulse h as: Nyquist criterion in the frequency domain C1 X `D 1 H Â f ` T Ã DT (7. A pulse h.68).79) and ISI vanishes.78) holds for PAM as well as for QAM. From (7. so that.82) Deﬁnition 7.1 The frequency 1=. f / is the Fourier transform of h. to obtain yk D ak it must be: Nyquist criterion in the time domain ( h0 D 1 hi D 0 i 6D 0 (7. Baseband equivalent model of a QAM system 559 where E g Rc is the energy of the receive ﬁlter.t/.3.81) From (7.81) we deduce an important fact: the Nyquist pulse with minimum bandwidth is given by: h. They can be derived using the Fourier transform of the sequence fh i g (1.7.79) have their equivalent in the frequency domain. are illustrated in Figure 7. for three values of the parameter ². in the absence of noise.80) where H.80) is equal to 1. f / D Th 0 rect f 1=T (7.79) is said to be a Nyquist pulse with modulation interval T . From the conditions (7.79) the lefthand side of (7. yk is a replica of ak . A family of Nyquist pulses widely used in telecommunications is composed of the raised cosine pulses whose time and frequency plots. The conditions (7.t/ D h 0 sinc t T F ! H. .
560 Chapter 7. f / D T rcos Â f .² 1=T Ã (7. Transmission over dispersive channels Figure 7. Time and frequency plots of raised cosine and square root raised cosine pulses for three values of the rolloff factor ².19.84) . ²/ D 2 B³ > cos @ 2 > > > > > > > :0 < jxj Ä 1C² 2 then H. We deﬁne 8 >1 > > > > > > > < 0 Ä jxj Ä 0 jxj 1 ² ²1 2 C A 1 2 jxj > ² 1 2 1C² 2 (7.x.83) ² rcos.
87) We note that. f / in (7.0/ 1 4 ²Á 4 (7.1 C ²/=T .3. the bandwidth of the baseband equivalent system is equal to .² 1=T df D T (7. from (7.84).85) It is easily proven that.0/ D 1 C1 (7.86) .88) .² df D 1 T rcos h.0/h. Baseband equivalent model of a QAM system 561 with inverse Fourier transform Â Â Ã Ä t t ³ sinc ² C h.89) Ã f . that is Ã Â Z C1 f (7.1 ²/ T T 4 T 4 Ä Â Ã½ Â Ã t 1 1 t C ² cos ³ sinc ² T 4 T 4 ½ Ä ½ Ä t t t C 4² cos ³.90) and the pulse energy is given by Z Eh D C1 1 T 2 rcos f .t/ D . from (7. the area of H.² H.84) is equal to one. Consequently.7. Later we will also refer to square root raised cosine pulses.91) .² df D 1 T rcos 1=T Â Â Ã s Â ² 1 4 ³ Ã (7.0/ D 1=T 1 and the energy is Â Ã Z C1 f 2 2 Eh D .1 C ²/ sin ³ .1 C ²/=.² df D T 1 T rcos 1=T 1 ²Á D H.1 ²/ T T T " D Â Ã2 # t t ³ 1 4² T T In this case Z h.1 ²/ sinc . f / D T rcos 1=T and inverse Fourier transform ½ Ä Â Ã½ Â Ã Ä t 1 t 1 t C ² cos ³ C sinc ² C h. with frequency response given by s Ã Â f (7.t/ D sinc T 4 T Â Ã Â Ã t t D sinc cos ³² T T 1 1 2 Ã Â t C sinc ² T 1 2 Ã½ 1 Ã Â t 2 2² T (7.83).2T /. for a QAM system the required bandwidth is .
called excess bandwidth parameter or rolloff factor.t0 1 q R . a graphic method to represent the effect of the choice of t0 for a given pulse q R .562 Chapter 7. Eye diagram From (7.t0 2T / C Ð Ð Ð (7. In the absence of ISI. 1. 1. therefore they offer a greater .t0 / is real: for a QAM system. We now illustrate.t0 / where i0 . for i 6D 0. f / in (7. i 6D0 ai q R .89) and (7. the pattern of y0 as a function of t0 is shown in Figure 7. is in the range between 0 and 1. i0 . We note that ² determines how fast the pulse decays in time.t0 /] and Im[y.67) the discretetime impulse response fh i g depends on the choice of the timing phase t0 (see Chapter 14) and on the pulse shape q R . We consider quaternary transmission with ak D Þn 2 A D f 3. with frequency response GC .94) and pulse q R as shown in Figure 7.21: it is seen that the possible values of y0 . must have a bandwidth equal to at least 1=. In relation to each possible value Þn of a0 . is given by y0 D y.1 From the Nyquist conditions we deduce that: 1.88). The parameter ². On the other hand. f / satisﬁes the Nyquist criterion and there is no noise. are further apart.88) is not the frequency response of a Nyquist pulse. given respectively by (7. 3g (7. Transmission over dispersive channels We note that H. Observation 7. at the decision point the sample y0 .t0 /] need to be represented. the channel.t0 / D C1 X i D 1. 2. are not sufﬁciently small with respect to h 0 . through an example. both Re[y.92) D a0 q R .t0 / D C1 X iD 1 ai q R . from (7.66) and (7.t0 / C i0 .t0 iT/ T / C a2 q R .t0 is the ISI. for various values of ² are shown in Figure 7. as a function of t0 . In the absence of noise.2T /.68) we observe that if the samples fh i g. otherwise intersymbol interference cannot be avoided. a data sequence can be transmitted with modulation rate 1=T without errors if H.19b. Plots of h.t/ and H. the ISI may result a dominant disturbance with respect to noise and impair the performance of the system.93) D ÐÐÐ Ca C T / C a1 q R . for Þn 2 A.t0 / D 0 and y0 D a0 q R .t0 iT/ (7.20. We consider a PAM transmission system where y.t0 /. f /.
.5 R 0 −0. 1. Desired component Þn qR .t0 / as a function of t0 . Þn 2 f 3.5 1 q (t) 0.3.5 −T 0 T t 2T 3T 4T Figure 7.7. Baseband equivalent model of a QAM system 563 1. Pulse shape for the computation of the eye diagram. 1.20. 3 αn=3 2 1 αn=1 αnqR(t0) 0 −1 α =−1 n −2 αn=−3 −3 −T 0 T 2T 3T 4T t0 Figure 7. 3g.21.
the values of y0 may be very close to each other.98) and iabs .t0 / (7. we show in Figure 7. For the considered pulse.t0 / D Þmax we have that imax . and this value is added to the desired sample a0 q R . In the general case. a0 DÞn fak g. a2 .t0 /. in general.t0 / (7.100) (7. it may result in i0 .t0 . the choice t0 D 1:5T guarantees the largest margin against noise.i.95) (7. the M 1 “pupils” of the eye diagram have a shape as illustrated in Figure 7.t0 /.t0 / D iabs . Transmission over dispersive channels margin against noise in relation to the peak of q R . . Þn / ½ iabs . a0 DÞn max min i0 .t0 i T /j (7. Þn / Þn q R . In this example. Þn / (7. : : : . For example a raised cosine pulse with ² D 1 offers greater immunity against errors in the choice of t0 as compared to the case ² D 0:125. that is both Þn and Þn belong to A.t0 / 6D 0.23 the eye diagram obtained with a raised cosine pulse q R .97) If the symbols fak g are statistically independent with balanced values.t0 .t0 / imin . it is easy to show that imax . We observe that as a result of the presence of ISI. i 6D0 jq R .t0 / i0 .t0 .t0 / around the desired sample Þn q R . the eye diagram is given in Figure 7.t0 . The price we pay is a larger bandwidth of the transmission channel.t0 / and imin . for a given t0 and for a given message : : : .22. a1 . for two values of the rolloff factor. The height a is an indicator of the noise immunity of the system. and therefore reduce considerably the margin against noise. a 1 . where there exists correlation between the symbols of the sequence fak g. symbols. Þn / D fak g. Consequently the eye may be wider as compared to the case of i.564 Chapter 7.24.t0 / is determined by the values imax .99) We note that both functions do not depend on a0 D Þn .t0 .t0 . In fact.t0 / D iabs . We also note that. which in this example occurs at instant t0 D 1:5T . For quaternary transmission.96) The eye diagram is characterized by the 2M proﬁles ( imax . The width b indicates the immunity with respect to deviations from the optimum timing phase. Þn / Ä iabs .d. however.t0 / C Þn 2 A imin . In general. The range of variations of y0 . Þn / D imin . deﬁning Þmax D max Þn n (7. the timing phase that offers the largest margin against noise is not necessarily found in relation to the peak of q R . where two parameters are identiﬁed: the height a and the width b.101) C1 X i D 1.
we select t1 so that the center of the eye falls in the center of the interval [t1 . Moreover. [t1 C 2T. Then the contours of the obtained eye diagram correspond to the different proﬁles (7. we must omit plotting the values of y. if the pulse q R .t/.4 Performance analysis Symbol error probability in the absence of ISI If the Nyquist conditions (7. t1 C T /.k ak 2 A (7. [t1 C T.79) are veriﬁed.t/ D s R . We now illustrate an alternative method to obtain the eye diagram.αn) 2 αnqR(t0)+imin(t0. To plot the eye diagram we need in principle to generate all the Mary symbol sequences of length Nh : in this manner we will reproduce the values of y.7. are mapped on the same interval. A long random sequence of symbols fak g is transmitted over the channel.i. Typically.d. has a duration equal to th . for example. 7. : : : ]. t1 C 2T /. If the contours of the eye do not appear.αn) 1 y0(t0) 0 −1 −2 −3 −T 0 T t0 2T 3T 4T Figure 7. Eye diagram for quaternary transmission and pulse qR of Figure 7.t/ for the ﬁrst and last Nh 1 modulation intervals.t/ in correspondence of the various proﬁles.t/ relative to the various intervals [t1 . on [t1 .t/ is at most equal to Nh .3. t1 C T /. it means that for all values of t0 the worst case ISI is larger than the desired component and the eye is shut. and deﬁne Nh D dth =T e.22.97). for transmission with i. Baseband equivalent model of a QAM system 565 3 αnqR(t0)+imax(t0.20. from (7.65) the samples of the received signal at the decision point are given by yk D ak C w R. t1 C T /. t 2 <. We note that. as they would be affected by the transient behavior of the system.3.102) . at every instant t 2 < the number of symbols fak g that contribute to y. symbols. t1 C 3T /. and the portions of the curve y.
5 (a) 5 4 3 2 1 0 −1 −2 −3 −4 −5 −0. .4 0. Eye diagram for quaternary transmission and raised cosine pulse qR with rolloff factor: (a) ² D 0:125 and (b) ² D 1.566 Chapter 7.2 0.2 −0.5 (b) Figure 7.23.24.4 0.1 0 t/T 0.3 0. Height a and width b of the ‘‘pupil’’ of an eye diagram.1 0.5 −0.1 0.5 −0.3 0.3 −0.4 −0.3 −0. a b t 0 Figure 7. Transmission over dispersive channels 5 4 3 2 1 0 −1 −2 −3 −4 −5 −0.2 0.2 −0.1 0 t/T 0.4 −0.
k ]] ! r D [r1 .I . ¦ I2 . hence from (7.105) If w R. then.57). yk. Baseband equivalent model of a QAM system 567 For a memoryless decision rule on yk . ¦ I2 is given by (7.106) D 2¦ I and using (6.k is needed and not its PSD.103) (7. for the purpose of computing Pe .k D yk D [yk. so that × 1.106) it follows that Â Pe ' 4 1 D 2 N0 E g Rc (7.k ]. r2 ]T ! s D [s1 . ak. With reference to Table 7. We also note that (7. However.k D [Re[w R.k g are statistically independent.I . and the values assumed by ak are equally likely. we have MPAM Â Pe D 2 1 1 M Ã Q. Im[w R. we consider dm D 2h 0 D 2. the above equation could lead to choosing a ﬁlter g Rc with very low energy. i.107) and (7. is obtained when the ratio is maximum.78). MQAM / (7. w2 ]T (7. only the variance of the noise w R.3. and still considering the ML detection criterion described in Section 6. and on the variance per dimension of the noise w R. regarding yk as an isolated sample.109) Apparently. the detection criterion leads to choosing the value of Þn 2 A that is closest to yk .Q ] w R. given the observation yk .9 Moreover.6. 9 We observe that this memoryless decision criterion is optimum only if the noise samples fw R. in this speciﬁc case between adjacent values Þn 2 A.79) for the absence of ISI is satisﬁed. The general case of computation of Pe in the presence of ISI and nonGaussian noise is given in Appendix 7. here g Rc is not arbitrary.7. we have the following correspondences: r R. .Q ] ak D [ak. (7. that is the minimum value of Pe . p / (7.43) holds.1. We will see in Chapter 8 a criterion to design the ﬁlter g Rc . Now. the error probability depends on the distance dm between adjacent signals.122) and (6.108) p M We note that. s2 ]T ! w D [w1 .107) Ã 1 p Q. Matched ﬁlter receiver Assuming absence of ISI.k .B.196). for a channel with white noise.1 and to Figure 7.108) imply that the best performance.e.k has a circularly symmetric Gaussian probability density function. as is the case if equation (7. deﬁning Ã Â dm 2 (7. but it must be chosen such that the condition (7. Hence.106) coincides with (6.104) (7.
.110) yields E g Rc D 1=E qC . Therefore (7.110) might imply.111) Substitution of (7.114). We stress the point that. from the condition h0 D F we obtain K D 1 E qC (7.114) where Ma =2 is the statistical power per dimension of the symbol sequence. with reference to the scheme of Figure 7. In Appendix 7. Transmission over dispersive channels Assuming that the pulse qC that determines the signal at the channel output is given.56) it is possible to determine the relation between the signaltonoise ratios at the decision point and 0 at the receiver input: for a QAM system it turns out MF D 0 1 2 Ma (7. The equation (7. f /Q C .E we describe a Monte Carlo method for simulations of a discretetime QAM system. In this case.2).12 and for white noise wC .114) is often used as an upper bound of the system performance. the carrier is omitted by using passband ﬁlters and exploiting the periodicity of the PSD of the sequence fak g. The scheme of a QAM system is repeated for convenience in Figure 7. it is not possible by varying the ﬁlter g Rc to obtain a higher at the decision point than (7. we have Ł G Rc .112) 1 [G Rc . In particular.10 on page 73) is provided by the receive ﬁlter g Rc matched to qC : hence the name matched ﬁlter (MF). f / D K Q C . f / e j2³ f t0 (7.25. In CAP systems. Using (7. and consequently a better Pe .4 Carrierless AM/PM (CAP) modulation The carrierless AM/PM (CAP) modulation is a passband modulation technique that is closely related to QAM (see Section 7. 7. f /] jtDt0 D K rqC .112) in (7. We further observe that the matched ﬁlter receiver corresponds to the optimum receiver developed in Chapter 6.110) where K is a constant.0/ D 1 (7. The only difference is that in that case the energy of the matched ﬁlter is equal to one. we note that we have ignored the possible presence of ISI at the decision point that the choice of (7. the solution (see Section 1.113) The matched ﬁlter receiver is of interest also for another reason. However. for a certain modulation system with a given pulse qC and a given 0.568 Chapter 7.109) assumes the form D MF D 2E qC N0 (7.
25.116) (7.Q are related by the Hilbert transform (1.25 is modiﬁed into the scheme of Figure 7.118) Applying Theorem 1.t/ sin. pb/ . pb/ .t/ sin. Consequently the two pulses (7.t/ D g Rc .26.29). Carrierless AM/PM (CAP) modulation 569 Figure 7. QAM implementation using passband ﬁlters. pb/ (7. if f 0 is larger than the bandwidth of h T x .t/ cos. Using passband ﬁlters.115) (7.t kT / e j2³ f 0 t kD 1 1 X kD 1 . where for simplicity we set '0 D 0.1 we observe that.2³ f 0 t/ g Rc.t/ D h T x . pb/ ak.7.119) kT / .I and g Rc. pb/ hypothesis always veriﬁed in practice.Q h T x. the .2³ f 0 t/ and the impulse responses of the receive ﬁlters are given by g Rc. QAM implementation using baseband ﬁlters.Q .I .t/ D h T x . pb/ .2³ f 0 t/ . pb/ .116) are orthogonal. pb/ ak.I h T x.t Q . in a QAM system the transmitted signal can be expressed as " # 1 X s Q AM .117) (7.I and h T x.t/ D g Rc .t/ cos.115) and (7. where the impulse responses of the transmit ﬁlters are given by h T x.2³ f 0 t/ h T x.Q .I . the same relation exists between the pulses g Rc.4. then the pulses h T x. Figure 7.t/ D Re ak h T x .26. the QAM scheme of Figure 7.163). pb/ .I . pb/ (7. From (7. Note that this property holds through the transmission channel.Q .Q .t Q D kT / .
Q . CAP modulation is obtained by leaving out the additional phase.t/e j2³ f 0 t D h T x. implemented by two realvalued ﬁlters with impulse responses given by (7.120) the transmitted signal is then given by " sC A P . the input to the modulation ﬁlters at instant k is given by ak D ak e j2³ f 0 kT . . pb/ kT / (7.t/ D h T x .26. On the other hand. an acquisition mechanism of the carrier must be introduced and typically QAM is adopted. In general. pb/ D .I h T x. If we deﬁne .t/ D Re 1 X kD 1 1 X kD 1 # Q ak h T x . as shown in Figure 7. ﬁltered by the transmission channel. In the case where f 0 is not much larger than 1=.2T /. if the transfer function of the transmission medium is unknown a priori.I D Re[ak e j2³ f 0 kT ] and ak.t/ and h T x.27). where ak.570 Chapter 7.118) (see Figure 7. for transmission channels that introduce frequency offset. the receive ﬁlters are adaptive (see Chapter 8). We note that CAP modulation is equivalent to QAM. CAP modulation may be preferred to QAM because it does not need carrier recovery.27. pb/ . pb/ kT / Because the pulses h T x. as usually occurs in data transmission systems over metallic cables.121) kT / ak.I . with the difference that in a QAM Q system the input symbols fak g are substituted by the rotated symbols fak g. then there is no difference between CAP and QAM. pb/ ak.Q h T x. pb/ Q h T x .Q .Q D Im[ak e j2³ f 0 kT ]. Therefore in the scheme of Q Q Figure 7.t/.Q .27. are still related by the Hilbert transform.t .I . the receiver uses a passband matched ﬁlter of the phasesplitter type. Transmission over dispersive channels Figure 7. Modulator and demodulator for a CAP system.I .t/ (7.2T /.117) and (7. If f 0 is an integer multiple of 1=T . QAM is usually selected if f 0 × 1=.t/ C j h T x. Q which is equal to the symbol ak with an additional deterministic phase that must be removed at the receiver.t .t .
On the other hand.i.29.122) Correspondingly we give in Figure 7.21 on page 457). the second compares the performance of an analog transmission system with that of a digital system for the transmission of analog signals represented by the PCM method.7.124) assuming errors are i. therefore the various distributions are obtainable starting from: O Pbit D P[b` 6D b` ] (7.123) Pbit /b ' (7. 1 bl 0 P bit 1. Binary channel associated with digital transmission.i. Regenerative PCM repeaters 571 7.28 (see also Figure 6.28..1 1 bPbit and Pe.5 for QAM..5 Regenerative PCM repeaters This section is divided into two parts: the ﬁrst considers a PCM encoded signal (see Chapter 5) and evaluates the effects of digital channel errors on the reproduced analog signal.1 Pbit /b (7. the transformation that O maps input bits fb` g in output bits fb` g is called binary channel and is represented in Figure 7.5. if Pbit − 1 it follows that .c ' bPbit Figure 7.d.29 the statistical model associated with a memoryless binary symmetric channel. .c D 1 .P bit P bit 1.P bit 1 ^ bl 0 Figure 7. 7.5. Memoryless binary symmetric channel.1 for PAM and to Figure 7. In the following it is useful to evaluate the error probability of words c composed of b bits: Pe. The simplest model considers errors to be i.1 PCM signals over a binary channel With reference to Figure 7. A binary channel is typically characterized by the bit rate Rb D 1=Tb (bit/s) and by the bit error probability.d.
572 Chapter 7.i C 1 /1 2 (7.k/ 2 f0. to simplify the analysis. which in turn is transmitted over a binary channel. : : : . with a quantization stepsize 1 given by (5. : : : .30 gives the composite scheme where an input analog signal is converted by an ADC into a binary sequence fb` g.k/2 j j jD0 > : Q i D −sat C . Q The operations that transform s. Figure 7.kTc / D Q i ! c. 1. with components c j .126) In (7. b 1. −sat ].125) where.t/ into fb` g are summarized as 1) sampling. Transmission over dispersive channels Linear PCM coding of waveforms As seen in Chapter 5.k/ D [cb 1 . Figure 7. Composite transmission scheme of an analog signal via a binary channel. c0 .k/ is the word of b bits transmitted over the binary channel.t/ is then reconstructed from the received bits by a DAC.k/] (7. We assume that each word is composed of b bits: hence there are L D 2b possible words corresponding to L quantizer levels. The inverse bit mapper of the ADC performs the following function: sq . PCM is essentially performed by an analogtodigital converter. The quantizer is assumed uniform in the range [ −sat . we assume the rule 8 b 1 > < i D X c . 1g.30.k/.125) c. . 2) quantization.25). and 3) coding. j D 0. The signal s . which represents the information contained in the instantaneous samples of an analog signal by words of b bits.
sq .k/ 6D c j .134) . : : : .kTc /] D Pe. c0 .125): c.kTc / Q (7. the error probability Q is given by: P[c j . the bit mapper of the Q Q DAC performs the inverse operation of (7.kTc / C eq .129) Given the onetoone map between words and quantizer levels.c ' bPbit s (7. Q If we denote with c. Hence. from (5.kTc / Q Therefore the overall relation is sq .131) where the two error terms are assumed uncorrelated.130) Overall system performance In Figure 7.41) it follows Meq D −2 12 D sat 12 3 Ð 22b (7. the presence of the quantization noise in the ADC.kTc / C eCh .5.k/.kTc / C eCh .k/2 j Q jD0 > : Q ıQ D −sat C .kTc / D s.k/] D Pbit Q (7. assuming the quantization noise uniform. the errors on the detection of the binary sequence at the output of the binary channel.133) (7.127) as illustrated in Figure 7. 2.kTc / and the binary channel reconstructs sq with a certain error eCh . if we express the generic detected bit at the output of the binary channel as c j .kTc / D sq .124) it follows that P[Qq . Regenerative PCM repeaters 573 We assume the binary channel symmetric and memoryless. 1g. The quantizer introduces an error eq such that sq .132) (7.30 the reconstructed analog signal s is different from the transmitted signal s Q for two reasons: 1.kTc / C eq .29.7.k/ 2 f0.kTc / D s.kTc / D Q ıQ Q Q where 8 b 1 X > <ıD Q c j .Q C 1 /1 ı 2 (7. using (7. as they are to be ascribed to different phenomena.k/ D [cb 1 .k/] the received word.128) (7.kTc / 6D sq .k/ ! sq . In particular.
141) 22 j D 12 Pbit (7.t/ s.k/ then (7. with probabilities given by 8 > P[" j D 1] D 1 Pbit > 2 < > > : Then. from (7.136) " j . from (7. Transmission over dispersive channels The computation of MeCh is somewhat more difﬁcult. First.137) 2 E[eCh .k/] D hence from (7. 0.574 Chapter 7. we get E[" j .143) .kTc /j2 ] s Meq Ms C MeCh (7.133) the output signaltonoise ratio is given by 3PCM D D D E[s 2 .k/] D Pbit Q For a memoryless binary channel E[" j1 . 1g.k/ D 0] j D P[c j .kTc / D 1 jD0 Let the error on the jth transmitted bit be Q " j .kTc /] D 12 Pbit b 1 X jD0 P[" j D 1] D 1 2 Pbit (7. Consequently.138).t/j2 ] s E[s 2 . observing (7.k/] D 1P[" j .135) becomes eCh .k//2 j Q (7.132) we have b 1 X .137) We note that " j .k/2 j (7.kTc / s.k/ (7.k/ 2 f 1. recalling footnote 3 on page 338.kTc / D 1 b 1 X jD0 c j .k/] D 0 and E[" 2 .139) (7.142) We note that.k/" j2 . the statistical power of the output signal of an interpolator ﬁlter in a DAC is equal to the statistical power of the input samples.135) eCh .k/ D c j .140) ( E[" 21 ] D Pbit j 0 for j1 D j2 for j1 6D j2 22b 1 3 (7.k/ 6D c j .k/ 6D 0] C 0P[" j .t/] E[jQ .c j .kTc /] E[jQq .138) P[" j D 0] D 1 Pbit (7.k/ c j .126) and (7.
5. and for a signaltoquantization noise ratio 3q D Ms =. equation (7. −sat . For example for Pbit D 10 4 . we get 3PCM D 3q 1 C 4Pbit . Regenerative PCM repeaters 575 55 50 b=8 45 40 b=6 35 (dB) 30 PCM Λ 25 b=4 20 15 b=2 10 5 0 −8 10 10 −7 10 −6 10 −5 10 P bit −4 10 −3 10 −2 10 −1 10 0 Figure 7. that is the output error is mainly due to the quantization error.144) We note that usually Pbit is such that Pbit 22b − 1: thus it results 3PCM ' 3q . Signaltonoise ratio of a PCM system as a function of Pbit . the above observations remain valid.144) is represented in Figure 7. −sat ] whereby 3q D 22b . To cover long distances it is therefore necessary to place repeaters along the transmission line to restore the signal.134) and (7.4 Ð 22b / the output signal is corrupted mainly by the quantization noise.31 for various values of b. however.7. In particular.142) and (7. .33)).31.144).22b 1/ (7.5.2 Regenerative repeaters The signal sent over a transmission line is attenuated and corrupted by noise. for a signal s 2 U. 7. going from b D 6 to b D 8 bits per sample yields an increment of 3PCM of only 2 dB.12 =12/ (see (5. Using (7.142).4 Ð 22b / the output is affected mainly by errors introduced by the binary channel. whereas for Pbit > 1=. For Pbit < 1=. We observe that in the general case of nonuniform quantization there are no simple expressions similar to (7.
t/. the only disturbance in s .92): 3D PsCh kT0 F A B (7. For a source at noise temperature T0 . Q . expressed as 3a D is given by 3a D 3 Ps D kT0 FB N (7.t/.148) it is assumed that (4. if F A is the noise ﬁgure of a single ampliﬁer.2 on page 271 with ž s. Hence.t/] E[jQ .t/. desired signal at the output of transmission section i. ž sCh . effective noise at the input of repeater i. then PsCh D 1 Ps ac (7. the noise builds up repeater after repeater and the overall signaltonoise ratio worsens as the number of repeaters increases. ž r.2.t/ C w. The cascade of ampliﬁers along a transmission line.t/j2 ] s (7.83) holds.t/ s. overall signal at the ampliﬁer input of repeater i. in a system with analog repeaters.77)). however.147) Obviously in the derivation of (7. deteriorates the signaltonoise ratio.t/. Moreover. the signaltonoise ratio at the ampliﬁer output of a single section is given by (4. if ac is the attenuation of the generic section i.t/ D sCh . it must be remembered that in practical systems. Transmission over dispersive channels Analog transmission The only solution possible in an analog transmission system is to place analog repeaters consisting of ampliﬁers with suitable ﬁlters to restore the level of the signal and eliminate the noise outside the passband of the desired signal. transmitted signal with bandwidth B and available power Ps .146) Analogously for N analog repeater sections. as the overall noise ﬁgure is equal to F D N Fsr (see (4. ž w.t/. possible distortion experienced by the desired signal through the various transmission channels and ampliﬁers also accumulates.148) E[s 2 .576 Chapter 7. contributing to an increase of the disturbance in s .t/ is due to additive noise introduced by the various Q devices. We consider the simpliﬁed scheme of Example 4. with available power PsCh . the overall signaltonoise ratio.t/. signal at the output of a system with N repeaters. ž s . Q We note that. as a statistical power ratio is equated with an effective power ratio.145) In this example both the transmission channel and the various ampliﬁers do not introduce distortion.
152) Analog repeaters: Pbit.N D 2. 10 We note that a more accurate study shows that the errors have a Bernoulli distribution [4]. then from (6.5.151) kT0 F A Bmin It is interesting to compare the bit error probability at the output of N repeaters in the two cases. regeneration allows a signiﬁcant saving in the power of the transmitted signal.125) we get ! r 3 2. the digital message fb` g is ﬁrst reconstructed. for a given overall Pbit .1 Pbit / N ' N Pbit (7.148). however. and ignoring the probability that a bit undergoes more errors along the various repeaters.150) M log2 M M2 1 where from (6.152) we used (7. it is necessary to specify the type of modulator. Ã Âr 1/ 3 0 (7. the bit error probability at the output of N regenerative repeaters is equal to10 Pbit. . Let us consider an MPAM system. we can resort to the regeneration of the signal.149) assuming Pbit − 1. Note. Basic scheme of digital regeneration. 11 To simplify the notation.t/. and then retransmitted by a modulator. Figure 7.1 on page 457) with error probability Pbit . To obtain an expression of Pbit . we have indicated with the same symbol s Ch the desired signal at the ampliﬁer input for both analog transmission and digital transmission.7. Even if a regenerative repeater is much more complex than an analog repeater. With reference to the scheme of O Figure 7.32.32. Regenerative PCM repeaters 577 Digital transmission In a digital transmission system.153) M2 1 2 0D Note that in (7. that in the ﬁrst case sCh depends linearly on s.M M Q M log2 M2 1 N Ã Âr 2. as an alternative to the simple ampliﬁcation of the received signal r.N ' 1 .M 1/ Q Pbit D 0 (7. Modeling each regenerative repeater by a memoryless binary symmetric channel (see Deﬁnition 6. and errors of the different repeaters statistically independent.t/. whereas in the second it represents the modulated signal that does not depend linearly on s.108)11 PsCh (7.N D M log M (7.M 1/ N Q 3 0 Regenerative repeaters: Pbit. given the signal r.
from (7. for example. To simplify the notation. and/or a modulator with higher spectral efﬁciency.157) 3PCM D 2b 2 > > > Âq Ã N regenerative repeaters > > > 3 > 1 C 4. that is assuming a uniform signal.155) in (7. valid for a uniform quantizer with 3q D 22b .t/ with the digital transmission. Using (7. by resorting to multilevel transmission. see (5.154) Consequently.22b : 22b 1/N Q Âq 3a N b Ã N analog repeaters (7. the modulation interval T is equal to log2 M=Rb . and the minimum bandwidth of the transmission channel is equal to Bmin D b 1 D B 2T log2 M (7.t/ and modulation of the message. For PCM coding of s.22b 1/N Q : b Or else. we get 8 22b > > > Ã Âq N analog repeaters > > > 3 > 1 C 4. for an MPAM modulator.44).152) and (7.22b 1/Q < b > > > > > > > 1 C 4. if M is small.22b 1/Q < bN (7. using (7.158) N regenerative repeaters 3PCM D . Obviously.148).151).144).146) we have 0D log2 M 3 b (7. Bmin may result very close to B or even smaller. we get 8 22b > > > Âq Ã > > > 3a > 1 C 4.156) The comparison between the two systems is based on the overall signaltonoise ratio for the same transmitted power and transmission channel characteristics. Substituting the value of 0 given by (7.578 Chapter 7.153). and recalling (7.t/ the bit rate of the message is given by Rb D b 2B (7. using a more efﬁcient digital representation of waveforms.155) We note that the digital transmission of an analog signal may require a considerable expansion of the required bandwidth. for example by CELP.156) for M D 2 in (7. initially we will consider a 2PAM as modulator. which includes PCM coding of s. Transmission over dispersive channels Comparison between analog and digital transmission We now compare the analog transmission of a signal s.
Consequently. The parameter b denotes the number of bits for linear PCM representation. . depending on the applications. Using regenerative repeaters. then Pbit is very small. Regenerative PCM repeaters 579 45 b=7 40 b=6 35 b=5 30 (dB) 25 b=4 Λ PCM 20 b=3 15 b=2 10 5 0 0 5 10 15 20 Λ a 25 (dB) 30 35 40 45 Figure 7. We note that 3PCM is typically higher than 3a . in practice it is interesting to determine the minimum value of 3 (or 0) so that 3PCM and 3a reach a certain value. the quantization error becomes predominant at the receiver.7. We show also a comparison for the same required bandwidth.156). for example N D 20 in Figure 7. for the same Pbit the modulator requires an increment of about 6.35. therefore. However.b 1/. which implies a modulator with M D 2b levels.b 1/ 10 log10 . the PCM system is penalized by the increment of the bandwidth of the transmission channel.33.b 1/ dB in terms of 0. While the previous graphs relate 3PCM directly to 3a . plotted in Figure 7. the increment in terms of 3 is equal to 6. from (7. with respect to 2PAM.35 these relations by varying the number N of repeaters and using a PCM encoder with b D 7. Therefore also for the same bandwidth. The curve of 3PCM as a function of 3 for 128PAM. digital transmission is more efﬁcient than analog transmission if the number of repeaters is large. the plot of 3PCM as a function of 3a is given in Figure 7. as long as a sufﬁciently large number of bits and 3a larger than 17 dB are considered. In this case. of the order of 20–40 dB. We illustrate in Figure 7. In the case of analog repeaters. assuming an adequate number of bits for PCM coding is used. 3PCM is always much higher than 3a . is shifted to the right by about 28 dB with respect to 2PAM.5.34.33. say. 3PCM as a function of 3a for analog repeaters and 2PAM. We note the threshold effect of Pbit as a function of 0 in a digital transmission system: if the ratio 0 is higher than a certain threshold.
35. as a function of 3 (signaltonoise ratio of each repeater section). 3a for analog transmission obtained by varying the number N of analog repeaters. 3PCM as a function of 3a for 2PAM transmission and N D 20 regenerative repeaters. The parameter b is the number of bits for linear PCM representation. Λ a (dB) ΛPCM(N=1000) 25 Λa(N=10) Λa(N=100) Λ (N=1000) a 20 15 10 5 0 10 15 20 25 30 35 40 Λ (dB) 45 50 55 60 65 Figure 7. The dashed line represents 3PCM for 128PAM and b D 7. obtained by varying the number N of regenerative repeaters.34. Transmission over dispersive channels 45 b=7 40 b=6 35 b=5 30 Λ PCM (dB) 25 b=4 20 b=3 15 b=2 10 5 0 0 5 10 15 20 Λ a (dB) 25 30 35 40 45 Figure 7. and 3PCM for digital transmission with 2PAM and b D 7.580 Chapter 7. 45 Λ PCM (N=10) 40 Λ 35 PCM (N=100) 30 Λ PCM . .
for analog transmission and digital transmission with three different modulators. Roden.. Papoulis. G. NJ: PrenticeHall. 1997. [3] M. The number of bits for PCM coding is b D 7. 1996. 1994. Couch. Digital and analog communication systems. Salehi. NJ: PrenticeHall. 3rd ed. NJ: PrenticeHall. S. for three different modulators. Bibliography 581 Figure 7. 3PCM D 3a D 36 dB (7. W. Upper Saddle River. Analog and digital communication systems. New York: McGrawHill.36 the minimum value of 3 as a function of the number of regenerative repeaters. [4] A. .7. 1991. Englewood Cliffs.36. Minimum value of 3 as a function of the number N of regenerative repeaters required to guarantee an overall signaltonoise ratio of 36 dB. Communication system engineering. for a given objective. Finally. Proakis and M. Probability.159) we illustrate in Figure 7. Upper Saddle River. Bibliography [1] L. random variables and stochastic processes. [2] J.
IEEE Trans. Kabal and P. 563–568. G. 3rd ed. Pasupathy. Gitlin. L. IEEE Trans. Messerschmitt. Data communication principles. 1965. New York: Plenum Press. on Information Theory. . IEEE Trans. Birkoff and S. New York: Macmillan Publishing Company. 1994. A. Lee. vol. July 1974. Duttweiler. 20. pp.. Boston. 1992. Sept. July 1968. 490–497. and K. [7] D. Hayes. Mazo. Weinstein. 1992.582 Chapter 7. “Intersymbol interference error bounds with application to ideal bandlimited signaling”. 1999. S. New York: Plenum Press. 2nd ed. vol. Saltzberg. Digital communication. G. Biglieri. Messerschmitt and E. Benedetto and E. vol.. MacLane. New York: Kluwer Academic Publishers. [11] R. “An upper bound on the error probability in decisionfeedback equalization”. [9] D. C. [8] G. [6] P. on Information Theory. E. 23. J. [10] B. Shanmugan. 921–934. MA: Kluwer Academic Publishers. Balaban. and D. [12] M. R. Simulation of communication systems. pp. Transmission over dispersive channels [5] S. J. and S. 1975. “Partialresponse signaling”. Jeruchim. Principles of digital transmission with wireless applications. A survey of modern algebra. 9. pp. on Communications. P.
38. 3. 1. NRZ level (NRZL) or. . 3. NRZ mark (NRZM): “1” is represented by a level transition.A. and match it to the characteristics of the channel (see (7. several representations of binary symbols are listed. especially in case the information message contains long sequences of ones or zeros. b` 2 f0. the binary sequence fb` g. Line codes for PAM systems 583 Appendix 7. NRZ space (NRZS): “1” is represented by no level transition.7. to facilitate synchronization at the receiver. “0” by a level transition. The sequence fak g is produced by a line encoder. every other case is represented by the zero level. this task may be performed also by the transmit ﬁlter.37.1 Line codes With reference to Figure 7. Figure 7. 2. The channel input is a PAM signal s. for transmission over AWGN channels in the absence of ISI.37. to shape the spectrum of the transmitted signals. in particular [1. obtained by modulating a rectangular pulse h T x . “0” by no level transition.A. is represented by a level transition.A Line codes for PAM systems The functions of line codes are: 1.t/. “10” or “01”. 1g. 7. to improve system performance in terms of Pe . or be the output of a channel encoder. Dicode NRZ: A change of polarity in the sequence fb` g. simply. This appendix is divided in two parts: in the ﬁrst. 2. Four formats are illustrated in Figure 7. 5]. partial response systems are introduced. PAM transmitter with line encoder. NRZ: “1” and “0” are represented by two different levels.17)). in the second. Nonreturntozero (NRZ) format The main feature of the NRZ family is that NRZ signals are antipodal signals: therefore NRZ line codes are characterized by the lowest error probability. could be directly generated by a source. For indepth study and analysis of spectral properties of line codes we refer to the bibliography. 4.
This property is usually not desirable.584 Chapter 7. 2. for example. RZ line codes are illustrated in Figure 7. Long sequences of ones or zeros in the sequence fb` g . Returntozero (RZ) format 1. Biphase level (B. as. we observe that the signal does not have zero mean.L) or Manchester NRZ: “1” is represented by a transition from high level to low level. is represented by a level transition. Biphase (Bφ) format 1. “0” by a zero pulse. Bipolar RZ or alternate mark inversion (AMI): Bits equal to “1” are represented by rectangular pulses having duration equal to half a bit interval.39. Polar RZ: “1” and “0” are represented by opposite pulses with duration equal to half a bit interval. “0” by a transition from low level to high level. sequentially alternating in sign. using a pulse having duration equal to half a bit interval. for transmission over coaxial cables.38. 3. 4. bits equal to “0” by the zero level. NRZ line codes. Dicode RZ: A change of polarity in the sequence fb` g. every other case is represented by the zero level. Unipolar RZ: “1” is represented by a pulse having duration equal to half a bit interval. “10” or “01”. Transmission over dispersive channels NRZ−L 2 NRZ−M 2 1 1 0 1 1 0 0 0 1 1 0 1 1 1 0 1 1 0 0 0 1 1 0 1 0 0 −1 −1 −2 0 2 4 6 t/T 8 10 −2 0 2 4 6 t/T 8 10 NRZ−S 2 Dicode NRZ 2 1 1 0 1 1 0 0 0 1 1 0 1 1 1 0 1 1 0 0 0 1 1 0 1 0 0 −1 −1 −2 0 2 4 6 t/T 8 10 −2 0 2 4 6 t/T 8 10 Figure 7.
Biphase line codes are illustrated in Figure 7. that this line code leads to a doubling of the transmission bandwidth. 2. a transition occurs at the end of the bit interval. Each block of K bits is then mapped into a block of N symbols belonging to an alphabet of cardinality M.A. This code shapes the spectrum similar to the Manchester code. Line codes for PAM systems 585 Unipolar RZ 2 1. Block line codes The input sequence fb` g is divided into blocks of K bits.5 0 0 1 1 0 0 0 1 1 0 1 1 1 0 1 1 0 0 0 1 1 0 1 0 −1 −0. Biphase space (B.S): A transition occurs at the beginning of every bit interval. “1” is represented by a second transition within the bit interval.40.M) or Manchester 1: A transition occurs at the beginning of every bit interval. “0” is represented by a constant level.40.39. 3. It is easy to see. “0” is represented by a constant level. if “0” is followed by another “0”. Biphase mark (B. “1” is represented by a constant level. however. “0” is represented by a second transition within the bit interval. RZ line codes. with . but requires a lower bandwidth.7. do not create synchronization problems.5 −1 0 2 4 6 t/T 8 10 −2 0 2 4 6 t/T 8 10 Bipolar RZ 2 Dicode RZ 2 1 1 0 1 1 0 0 0 1 1 0 1 1 1 0 1 1 0 0 0 1 1 0 1 0 0 −1 −1 −2 0 2 4 6 t/T 8 10 −2 0 2 4 6 t/T 8 10 Figure 7. The delay modulation line code is illustrated in Figure 7.5 Polar RZ 2 1 1 0. Delay modulation or Miller code “1” is represented by a transition at midpoint of the bit interval.
161) At the decoder the bits of the information sequence may be recovered by O O bk D a k C bk O Note that ak 2 f 1.40.586 Chapter 7. 1g (7. 0.163) From (7. Transmission over dispersive channels Biphase−L 2 Biphase−M 2 1 1 0 1 1 0 0 0 1 1 0 1 1 1 0 1 1 0 0 0 1 1 0 1 0 0 −1 −1 −2 0 2 4 6 t/T 8 10 −2 0 2 4 6 t/T 8 10 Biphase−S 2 Delay Modulation 2 1 1 0 1 1 0 0 0 1 1 0 1 1 1 0 1 1 0 0 0 1 1 0 1 0 0 −1 −1 −2 0 2 4 6 t/T 8 10 −2 0 2 4 6 t/T 8 10 Figure 7.160) The KBNT codes are an example of block line codes where the output symbol alphabet is ternary f 1. that is a k D bk bk 1 with bk 2 f0. f / D Pb . 1g. 0. f / 4 sin2 .161). f / j1 e j2³ f T 2 j D Pb . Alternate mark inversion (AMI) We consider a differential binary encoder.³ f T / . the constraint 2K Ä M N (7. in particular ( ak D 1 (7. 1g.and delay modulation line codes. the relation between the PSDs of the sequences fak g and fbk g is given by Pa . B.162) š1 0 if bk 6D bk if bk D bk 1 1 (7.
161) we have ma D 0. We observe that the AMI line code is a particular case of the partial response system named dicode [6]. 0. In any case.t/ is an additive white Gaussian noise. O Moreover. observing O (7.165) if bk D 1 if bk D 0 (7. given that an error occurs in fak g. Also in this case ma D 0.41 for different values of p.1 2 p/ sin2 ³ f T Ð The plot of Pa e j2³ f T is shown in Figure 7.166) In other words. . 7. 1g. This problem can be solved by precoding: from the O sequence of bits fbk g we ﬁrst generate the sequence of bits fck g.7. Consequently. it results in ( š1 ak D 0 ck 1 1 (7. and p D P[bk D 1].M 1/. : : : . Next. because a detector at the receiver must now decide among three levels.A. If the power of the transmitted signals is constrained. f / exhibits zeros at frequencies that are integer multiples of 1=T . with ck 2 f0.169) and w. from (7. and a bit bk D 1 is mapped alternately in ak D C1 or ak D 1.166) decoding may be performed simply by taking the magnitude of the detected symbol: O O bk D jak j (7. 1g. Hence. a k D ck with ak 2 f 1.168) p2 C . that is for ak 2 f 1. generates a sequence of bits fbk g that are in O error until another error occurs in fak g. . . independently of the distribution of fbk g. for MQAM systems the results can be extended to the signals on the I and Q branches. we have Á sin2 . the biggest problem is the error propagation at the decoder.167) It is easy to prove that for a message fbk g with statistically independent symbols.164) (7. we recall in Figure 7. a disadvantage of the encoding method (7.M 3/. long sequences of information bits fbk g that are all equal to 1 or 0 generate sequences of symbols fak g that are all equal: this is not desirable for synchronization. Line codes for PAM systems 587 Therefore Pa . 12 In the present analysis only MPAM systems are considered.1 p/ (7.1.M 3/. .161) is a reduced noise immunity with respect to antipodal transmission. in particular at f D 0. by c k D bk ý c k where ý denotes the modulo 2 sum.2 Partial response systems From Section 7. Moreover. 1g.162). where the symbols fak g belong to the following alphabet12 of cardinality M: ak 2 A D f .A. from (7.M 1/g (7.42 the block diagram of a baseband transmission system. which. a bit bk D 0 is mapped into the symbol ak D 0.³ f T / Pa e j2³ f T D 2 p. Note that the PSD presents a zero at f D 0.
ej2³ fT / of an AMI encoded message.D/ D N 1 X i D0 li D i (7.k D w R . Block diagram of a baseband transmission system.43a. Power spectral density Pa .t0 C i T /g. and w R. We assume that fh i g is equal to zero for i < 0 and i ½ N . obtained by ﬁltering w.588 Chapter 7. Transmission over dispersive channels Figure 7. . is added to the desired signal.t/.43b. and D is the unit delay operator. as illustrated in Figure 7.t/ D h T x Ł g Rc .41. ak T h Tx s(t) g sCh (t) r (t) Ch w(t) Ch g rR (t) yk t 0 +kT ^ ak Rc Figure 7.42. We assume that the transmission channel is ideal: the overall system can then be represented as an interpolator ﬁlter having impulse response q.171) where the coefﬁcients fli g are equal to the samples fh i g.t0 C kT /.t/ by the receive ﬁlter. where fh i D q.170) A noise signal w R . The partial response (PR) polynomial of the system is deﬁned as l.t/ (7. Sampling the received signal at instants t0 CkT yields the sequence fyk g. The discretetime equivalent of the system is shown in Figure 7.
that forces the system to have an overall discretetime impulse response equal to fh i g. and g is an analog ﬁlter satisfying the Nyquist criterion for the absence of ISI.44. allows simpliﬁcation of the study of the properties of the ﬁlter h.44 suggests two possible ways to implement the system of Figure 7. from (7.173) Note that the overall scheme of Figure 7. C1 X mD 1 G f mÁ DT T (7.e j2³ f T /.44 is equivalent to that of Figure 7. observing (7. Line codes for PAM systems 589 Figure 7. Equivalent schemes to the system of Figure 7.42 is decomposed into two parts: ž a ﬁlter with frequency response l.43. on one hand. periodic of period 1=T . In other words.43a with q.D/ in Figure 7.174) Also. ak T l(D) ak (t) g rR (t) yk t 0 +kT ^ ak w R (t) Figure 7.172). The scheme of Figure 7. PR version of the system of Figure 7.D/.t iT/ (7. to design an efﬁcient receiver.42.A. ž an analog ﬁlter g that does not modify the overall ﬁlter h.t/ D N 1 X i D0 li g.44 are given by . A PR system is illustrated in Figure 7. and. on the other.44.44.D/ and limits the system bandwidth.172) The symbols at the output of the ﬁlter l.42: . As it will be clear from the analysis.t/ ak D N 1 X i D0 li ak i (7.42. the decomposition of Figure 7.D/ is deﬁned in (7. the equivalent discretetime model is obtained for h i D li . where l.174) the system of Figure 7.171).7.
f / (7.177) into (7. On the other hand.t/j asymptotically decays as 1=jtjnC1 . then the transmit ﬁlter h . With the aim of maximizing the transmission bit rate. f / G Rc .D/ has a zero of multiplicity greater than one in D D 1. f / Tx Rc (7.176) g is a Nyquist ﬁlter.D/ is implemented as a component of the transmitter by a digital ﬁlter. The choice of the PR polynomial Several considerations lead to the selection of the polynomial l.P R/ .2T / can be widened.t/ D li sinc T i D0 (7.179) b) Spectral zeros at f D 1=.1 C D/n as a factor. f / around f D 1=. then jq.e. therefore the transmit ﬁlter h T x and the receive ﬁlter g Rc must satisfy the relation HT x . Transmission over dispersive channels ak T l(D) ak(t) h Tx (PR) s(t) g Ch g (PR) Rc rR (t) yk t 0 +kT ^ ak w(t) Figure 7.t/ D sinc 2T : T 0 elsewhere l.t/.D/ has . if l. f / G .172) yields the following conditions on the ﬁlter g: 8 Â Ã 1 < F 1 t T jfj Ä G. It is easily proven that in a minimum bandwidth system.172) that in both relations (7. From the theory of signals. observing (7. f / D Q. f / D l.178) ! g. The continuity of Q.P R/ . Digital: the ﬁlter l. from (7.D/. f / D (7. it is known that if Q.n 1/th derivative of Q.175) 2. Note from (7.P R/ and receive ﬁlter g . the .n 1/ derivatives are continuous and the nth derivative is discontinuous.175) it must be 1 (7.P R/ must satisfy the relation Tx Rc H.174) the ﬁlter q assumes the expression Ã Â N 1 X t iT q.e j2³ f T / G. then the transition band of G. i. Implementation of a PR system using a digital ﬁlter.177) 2T Substitution of (7. f / is continuous if and only if l. f / D G. f / D 0 jfj> Correspondingly. f / and its ﬁrst . f / and of its derivatives helps to reduce the portion of energy contained in the tails of q.590 Chapter 7. a) System bandwidth.176) The implementation of a PR system using a digital ﬁlter is shown in Figure 7.2T /.175) and (7.e j2³ f T / G.45. . many PR systems are designed for minimum bandwidth. Analog: the system is implemented in analog form. thus simplifying the design of the analog ﬁlters. 1.45.
for the implementation of SSB modulators (see Example 1. Example 7.N 1/=2.182) is given by (7. or for transmission over channels with frequency responses that exhibit a spectral null at the frequency f D 0.2T / and has the following expression l. f / and q. if l.e j2³ f T / D 2j e j³ f T sin. as well as the correQ sponding expressions of Q.e j2³ f T / at f D 0.D/ have an alphabet A. detection of the sequence fak g by a threshold detector will cause a loss in system performance. Note that a zero of l.t/ holds n l .D/ has been selected. e.1 (Dicode ﬁlter) The dicode ﬁlter introduces a zero at frequency f D 0 and has the following expression l.2 (Duobinary ﬁlter) The duobinary ﬁlter introduces a zero at frequency f D 1=. e) Some examples of minimum bandwidth systems. the symbols at the output of the ﬁlter l.t/ .t/ of the output alphabet A. If we indicate with n l the number of coefﬁcients of l.t/ Ä M n l (7. A transmitted signal with attenuated spectral components at low frequencies is desirable in many cases.D/ different from zero. Q In the next three examples.M 1/ C 1.t/ . f / In Table 7.t/ D q t Q 2 (7. observing (7.4 on page 58).D/ contains more than one factor .183) The frequency response.N 1/T Q. obtained by setting D D e l. d) Number of output levels. f / and q.D/ D 1 D j2³ f T .M 1/ C 1 Ä M . From (7. then n l increases and. Line codes for PAM systems 591 c) Spectral zeros at f D 0. then the following inequality for M . As the coefﬁcients fli g are generally symmetric or antisymmetric around i D .D/ that are often found in practical applications of PR systems are considered.2 the more common polynomials l.A. if the coefﬁcients fli g are all equal.A. We note that. it is possible to evaluate the expression of Q.180) In particular. (7.t/ signal is constrained.7. f / D e j³ f .184) . In the case of minimum bandwidth systems. polynomials l.D/ in D D 1 corresponds to a zero of l. then M . also the number of output levels increases.t/ of cardinality M .N 1/T q.g.A.D/ D 1 C D (7. it is convenient to consider the timeshifted pulse Ã Â .180).t/.173). If the power of the transmitted .t/ once the polynomial l.³ f T / Example 7.D/ are described.7.t/ D n l .1 š D/.181) Q Q. and the cardinality M .
³ f T / sin.³t=T / ³ 4t 2 T 2 2T 2 sin.³t=T / 2 ³t t T2 Â Ã 2 2T 3t 2T sin. Example 7.179) we have q.e j2³ f T D/ .4t 2 T 2 / 8T 3 sin.2³ f T / (7.³ f T / D3 j 4T cos.e Observing (7.4³ f T / C j 3T sin.185) Â Ã Â Ã t T t C sinc T T (7. Transmission over dispersive channels Table 7.2³ f T / 4T cos2 . f / for j f j Ä 1=.D/ D .2³ f T / 4T sin2 .³ f T / sin.188) .2³ f T / 4T sin.³t=T / ³t t 2 4T 2 Â Ã 3t T T2 sin.186) The plot of the impulse response of a duobinary ﬁlter is shown in Figure 7.2³ f T / T C T cos.³t=T / ³ T 2 4t 2 8T t cos.4t 2 3T 2 / ³ .4t 2 T 2/ M .4t 2 9T 2 /.³ f T / j 2T sin.3 (Modiﬁed duobinary ﬁlter) The modiﬁed duobinary ﬁlter combines the characteristics of duobinary and dicode ﬁlters.4t 2 9T 2 /.t/ 1C D 1 D D2 2T cos.2 Properties of several minimum bandwidth systems.A. in line with what was stated at point b) regarding the aymptotical decay of the pulse of a PR system with a zero in D D 1.1 The frequency response becomes l.³t=T / ³ t2 T 2 2T 3 sin. and has the following expression l.2³ f T / C j 3T sin.t/ D sinc j2³ f T / D 2e j³ f T cos. l.2³ f T / T C T cos.³t=T /.t/ Q 4T 2 cos.187) /D1 e D 2j e j2³ f T (7.³t=T / 2 2 ³t t 4T 4M 3 1 2D 2 C D 4 D2 D4 4M 3 2C D D2 4M 3 2 4M 3 The frequency response is given by l. We notice that the tails of the two sinc functions cancel each other.2T / q.³ f T / (7.³t=T / 64T 3 t ³ .4³ f T / 2M 2M 1 1 1 2M 1 1 C 2D C D 2 D2 4M 3 1C D 4M 3 1 D D2 C D3 16T 2 cos.³ f T / j 2T sin.D/ Q Q.³t=T / ³t T 2 t 2 cos.1 C D/ D 1 j4³ f T D2 sin.46 with a continuous line.592 Chapter 7.
Line codes for PAM systems 593 1. the spectrum of the transmitted signal is given by (see (7.47 the PSD of a minimum bandwidth PR system is compared with that of a PAM system. The spectrum of the sequence of symbols fak g is assumed white.190) Tx þ T For a minimum bandwidth system.46 with a dashed line. so that the spectrum is obtained as the . With reference to the PR system of Figure 7.189) The plot of the impulse response of a modiﬁed duobinary ﬁlter is shown in Figure 7. Plot of q.45. f / given by (7. For the PR system. f / < jfj Ä a 2T Ps .5 1 0.t/ D sinc Â Ã t T Â sinc t 2T T Ã (7. Using (7. f / D (7.17)) þ þ2 þ1 þ þ l. a modiﬁed duobinary ﬁlter is considered.P R/ . (7. f /þ Pa .7.5 q(t) 0 −0.46.178).A.191) 1 > :0 jfj > 2T In Figure 7.t/ for duobinary ( ) and modiﬁed duobinary (.190) simpliﬁes into Tx 8 1 > jl.) ﬁlters.5 −1 −3 −2 −1 0 1 t/T 2 3 4 5 Figure 7. f / D þ (7.P R/ . with H. f / Ps . f) Transmitted signal spectrum.e j2³ f T / H.179) it results in q.e j2³ f T /j2 P .
product of the functions jl. .2³ f T /j2 and jH.47. and the spectrum is plotted with a dashed line.D/ is used. as it is deliberately introduced.k D ak D l0 ak C N 1 X i D1 li ak i (7. 1. whereas the summation represents the ISI term that is often designated as “controlled ISI”. The study of the other solutions should be postponed until the equalization methods of Chapter 8 are examined. For the PAM system. At the equalizer output. e j2³ f T /j2 D j2 sin.P R/ . A zeroforcing linear equalizer (LEZF) having D transform equal to 1=l. .k .192) The term l0 ak is the desired part of the signal s R.t/ s R. f /j2 D T 2 rect. PSD of a modiﬁed duobinary PR system and of a PAM system. Transmission over dispersive channels Figure 7. f T /.43b. the signal s R.k g. Tx plotted with continuous lines. LEZF. the transmit ﬁlter h T x is a square root raised cosine with rolloff factor ² D 0:5.t/ The receiver detects the symbols fak g using the sequence of samples fyk D ak C w R.D/ in the following form . Symbol detection and error probability We consider the discretetime equivalent scheme of Figure 7. 13 We discuss four possible solutions. at instant k the symbol ak plus a noise term is 13 For a ﬁrst reading it is suggested that only solution 3 is considered.k can be expressed as a function of symbols fak g and coefﬁcients fli g of the ﬁlter l.594 Chapter 7.
as shown in Figure 7.48d.48b. O as illustrated in Figure 7.k C (7. 2.192) in (7. Viterbi algorithm.A. Threshold detector with M . It yields the best performance. O N 1 X i D1 1 yk D ak C Q l0 ! li ek i w R. A second solution resorts to a decisionfeedback equalizer (DFE). DFE. however.t/ levels. We observe that at the decision point the signal yk has the expression N 1 yk D Q l0 If we indicate with ek D ak we obtain .t/ the M . 4.D/=l0 . then substituting (7.193) ak a detection error. We note.48a. shown in Figure 7.48. exploits .t/ ary nature of the symbols ak . An Mlevel threshold detector is also employed by the DFE. the detected symbols fak g are obtained by an Mlevel threshold detector.194) shows that a wrong decision negatively inﬂuence successive decisions: this phenomenon is known as error propagation. . but there is no noise ampliﬁcation as the ISI is removed by the feedback ﬁlter. and makes use of a threshold detector with . shown in Figure 7. 3. This solution.k N 1 X i D1 ! li ak O i (7. This structure does not lead to noise ampliﬁcation M as solution 1. This solution. that the ampliﬁcation of noise by the ﬁlter 1=l. Line codes for PAM systems 595 Figure 7.7.t/ ak C w R. because the noise is eliminated by the threshold detector.D/ is inﬁnite at frequencies f such that l. obtained.t/ levels followed by a LEZF. however.193).194) The equation (7. having D transform equal to 1 l. there is still the problem of error propagation. corresponds to maximumlikelihood sequence detection (MLSD) of fak g.e j2³ f T / D 0.48c. Four possible solutions to the detection problem in the presence of controlled ISI.
596
Chapter 7. Transmission over dispersive channels
Solution 2 using the DFE is often adopted in practice: in fact it avoids noise ampliﬁcation and is simpler to implement than the Viterbi algorithm. However, the problem of error propagation remains. In this case, using (7.194) the error probability can be written as þ # Ã "þ Â N 1 þ þ X 1 þ þ P þw R;k C Pe D 1 li ek i þ > l0 (7.195) þ þ M i D1 A lower bound Pe;L can be computed for Pe by assuming the error propagation is absent, or setting fek g D 0, 8k, in (7.195). If we denote by ¦w R the standard deviation of the noise w R;k , we obtain Ã Ã Â Â l0 1 (7.196) Q Pe;L D 2 1 M ¦w R Assuming w R;k white noise, an upper bound Pe;U is given in [7] in terms of Pe;L : Pe;U D .M=.M M N 1 Pe;L 1// Pe;L .M N
1
1/ C 1
(7.197)
From (7.197) we observe that the effect of the error propagation is that of increasing the error probability by a factor M N 1 with respect to Pe;L . A solution to the problem of error propagation is represented by precoding, which will be investigated in depth in Chapter 13.
Precoding
We make use here of the following two simpliﬁcations: 1. the coefﬁcients fli g are integer numbers; 2. the symbols fak g belong to the alphabet A D f0; 1; : : : ; M because arithmetic modulo M is employed. We deﬁne the sequence of precoded symbols fak g as: N ! N 1 X . p/ . p/ li ak i mod M N a k l0 D a k N
i D1 . p/
1g; this choice is made
(7.198)
We note that (7.198) has only one solution if and only if l0 and M are relatively prime [8]. In case l0 D Ð Ð Ð D l j 1 D 0 mod M, and l j and M are relatively prime, (7.198) becomes ! N 1 X . p/ . p/ ak j l j D ak N li ak i mod M N (7.199)
i D jC1
For example, if l.D/ D 2C D Therefore (7.199) is used.
D 2 and M D 2, (7.198) is not applicable as l0 mod M D 0.
7.A. Line codes for PAM systems
597
Applying the PR ﬁlter to fak g we obtain the sequence N ak D
.t/ N 1 X i D0
. p/
li ak N
. p/ i
(7.200)
From the comparison between (7.198) and (7.200), or in general (7.199), we have the fundamental relation
.t/ ak mod M D ak
(7.201)
.t/ Equation (7.201) shows that, as in the absence of noise we have yk D ak , the symbol ak can be detected by considering the received signal yk modulo M; this operation is O memoryless, therefore the detection of ak is independent of the previous detections fak i g, O i D 1; : : : ; N 1. Therefore the problem of error propagation is solved. Moreover, the desired signal is not affected by ISI. If the instantaneous transformation
ak
. p/
. p/
D 2ak N
. p/
.M
1/
(7.202)
is applied to the symbols fak g, then we obtain a sequence of symbols that belong to the N . p/ in (7.169). The sequence fa . p/ g is then input to the ﬁlter l.D/. Precoding alphabet A k consists of the operation (7.198) followed by the transformation (7.202). However, we note that (7.201) is no longer valid. From (7.202), (7.200), and (7.198), we obtain the new decoding operation, given by ! .t/ ak ak D C K mod M (7.203) 2 where K D .M 1/
N 1 X i D0
li
(7.204)
A PR system with precoding is illustrated in Figure 7.49. The receiver is constituted by a threshold detector with M .t/ levels that provides the symbols fak g, followed by a block O .t/ that realizes (7.203) and yields the detected data fak g. O
Error probability with precoding
To evaluate the error probability of a system with precoding, the statistics of the symbols .t/ fak g must be known; it is easy to prove that if the symbols fak g are i.i.d., the symbols .t/ fak g are also i.i.d.
ak a (p) precoder
k
a (t) l(D)
k
yk
^ (t) ak
decoder
^ ak
Figure 7.49. PR system with precoding.
598
Chapter 7. Transmission over dispersive channels
If we assume that the cardinality of the set A.t/ is maximum, i.e. M .t/ D M n l , then the .t/ output levels are equally spaced and the symbols ak result equally likely with probability
.t/ P[ak D Þ] D
1 M nl
Þ 2 A.t/
(7.205)
.t/ In general, however, the symbols fak g are not equiprobable, because several output levels are redundant, as can be deduced from the following example.
Example 7.A.4 (Dicode ﬁlter) We assume M D 2, therefore ak D f0; 1g; the precoding law (7.198) is simply an exclusive or and ak N
. p/ . p/
D ak ý ak N
. p/ 1
(7.206)
The symbols fak g are obtained from (7.202), ak they are antipodal as ak are given by
. p/ . p/
D 2ak N
. p/
1
(7.207)
D f 1; C1g. Finally, the symbols at the output of the ﬁlter l.D/
.t/ ak D ak . p/ . p/ 1 . p/ . p/ 1
ak
D 2 ak N
ak N
Á
(7.208)
.t/ The values of ak 1 , ak , ak and ak are given in Table 7.3. We observe that both output N N levels š2 correspond to the symbol ak D 1 and therefore are redundant; the three levels are not equally likely. The symbol probabilities are given by .t/ P[ak D š2] D .t/ P[ak 1 4 1 2
. p/
. p/
(7.209)
D 0] D
Figure 7.50a shows the precoder that realizes equations (7.206) and (7.207). The decoder, O realized as a map that associates the symbol ak D 1 to š2, and the symbol ak D 0 to 0, is O illustrated in Figure 7.50b.
Table 7.3 Precoding for the dicode ﬁlter.
ak N
. p/ 1
ak 0 1 0 1
ak N
. p/
.t/ ak
0 0 1 1
0 1 1 0
0 C2 0 2
7.A. Line codes for PAM systems
599
0 1
D
(a) precoder
^ (t) a
k
0 2
0 1
^ ak
(b) decoder
Figure 7.50. Precoder and decoder for a dicode ﬁlter l.D/ with M D 2.
Alternative interpretation of PR systems
Up to now we have considered a general transmission system, and looked for an efﬁcient design method. We now assume that the system is given, i.e. that the transmit ﬁlter as well as the receive ﬁlter are assigned. The scheme of Figure 7.44 can be regarded as a tool for the optimization of a given system where l.D/ includes the characteristics of the .t/ transmit and receive ﬁlters: as a result, the symbols fak g no longer are the transmitted symbols, but are to be interpreted as the symbols that are ideally received. In the light of these considerations, the assumption of an ideal channel can also be removed. In this case the ﬁlter l.D/ will also include the ISI introduced by the channel. We observe that the precoding/decoding technique is an alternative equalization method to the DFE that presents the advantage of eliminating error propagation, which can considerably deteriorate system performance. Q In the following two examples [9], additive white Gaussian noise w R;k D wk is assumed, and various systems are studied for the same signaltonoise ratio at the receiver. Example 7.A.5 (Ideal channel g) a) Antipodal signals. We transmit a sequence of symbols from a binary alphabet, ak 2 f 1; 1g. The received signal is Q yk D ak C w A;k
2 where the variance of the noise is given by ¦w A D ¦ I2 . Q At the receiver, using a threshold detector with threshold set to zero, we obtain
Â Pbit D Q
1 ¦I
Ã (7.211)
← ←
ak
ak
(p)
(p) 1 a k +1
← ←
(7.210)
600
Chapter 7. Transmission over dispersive channels
.t/ b) Duobinary signal with precoding. The transmitted signal is now given by ak D ak C . p/ . p/ ak 1 2 f 2; 0; 2g, where ak 2 f 1; 1g is given by (7.202) and (7.198). The received signal is given by
. p/
yk D ak C w B;k Q
2 where the variance of the noise is ¦w B D 2¦ I2 , as ¦ 2.t/ D 2. Q ak
.t/
(7.212)
At the receiver, using a threshold detector with thresholds set at š1, we have the following conditional error probabilities: Ã Â 1 .t/ P[E j ak D 0] D 2Q ¦w B Q Ã Â 1 .t/ .t/ P[E j ak D 2] D P[E j ak D 2] D Q ¦w B Q Consequently, at the detector output we have Pbit D P[ak 6D ak ] O
.t/ .t/ D P[E j ak D 0] 1 C P[E j ak D š2] 1 2 2 Ã Â 1 D 2Q p 2 ¦I
We observe a worsening of about 3 dB in terms of the signaltonoise ratio with respect to case a). c) Duobinary signal. given by
.t/ The transmitted signal is ak D ak C ak 1.
The received signal is (7.213)
yk D ak C ak
1
C wC;k Q
2 where ¦wC D 2¦ I2 . We consider using a receiver that applies MLSD to recover the data; Q from Example 8.12.1 on page 687 it results in p ! Â Ã 8 1 (7.214) Pbit D K Q DKQ 2¦wC ¦I Q
where K is a constant. We note that the PR system employing MLSD at the receiver achieves a performance similar to that of a system transmitting antipodal signals, as MLSD exploits the correlation .t/ between symbols of the sequence fak g. Example 7.A.6 (Equivalent channel g of the type 1 C D) In this example it is the channel itself that forms a duobinary signal.
7.A. Line codes for PAM systems
601
d) Antipodal signals. Transmitting ak 2 f 1; 1g, the received signal is given by yk D ak C ak
1
C w D;k Q
(7.215)
2 where ¦w D D 2¦ I2 . Q An attempt at preequalizing the signal at the transmitter by inserting a ﬁlter l.D/ D .t/ 1=.1 C D/ D 1 D C D 2 C Ð Ð Ð would yield symbols ak with unlimited amplitude; therefore such a conﬁguration cannot be used. Equalization at the receiver using the scheme of Figure 7.48a would require a ﬁlter of the type 1=.1 C D/, which would lead to unlimited noise enhancement. Therefore we resort to the scheme of Figure 7.48c, where the threshold detector has thresholds set at š1. To avoid error propagation, we precode the message and transmit the . p/ sequence fak g instead of fak g. At the receiver we have
yk D ak
. p/
C ak
. p/ 1
C w D;k Q
(7.216)
We are therefore in the same conditions as in case b), and Ã Â 1 Pbit D 2Q p 2 ¦I
(7.217)
e) MLSD receiver. To detect the sequence of information bits from the received signal (7.215), MLSD can be adopted. Pbit is in this case given by (7.214).
602
Chapter 7. Transmission over dispersive channels
Appendix 7.B
Computation of Pe for some cases of interest
7.B.1
Pe in the absence of ISI
yk D h 0 ak C w R;k ak 2 A
In the absence of ISI, the signal at the decision point is the type (7.102) (7.218)
where w R;k is the sample of an additive noise signal. Assuming fw R;k g stationary with probability density function pw .¾ /, from (7.218) for ak D Þn 2 A we have p yk jak .² j Þn / D pw .² Therefore the MAP criterion (6.26) becomes ² 2 Rm a k D Þm O if Þm D arg max pn pw .²
Þn
h 0 Þn /
(7.219)
h 0 Þn /
(7.220)
We consider now the application of the MAP criterion to an MPAM system, where Þn D 2n 1 M n D 1; : : : ; M (7.221)
The decision regions fRn g, n D 1; : : : ; M, are formed by intervals, or, in general, by the union of intervals, whose boundary points are called decision thresholds f−i g, i D 1; : : : ; M 1. Example 7.B.1 (Determination of the optimum decision threholds) We consider a 4PAM system with the following symbol probabilities: ¦ ² 3 3 1 1 ; ; ; f p1 ; p2 ; p3 ; p4 g D 20 20 2 5 The noise is assumed to have an exponential probability density function pw .¾ / D þ e 2
j¾ jþ
(7.222)
(7.223)
2 where þ is a constant; the variance of the noise is given by ¦w D 2=þ 2 . The curves
pn pw .²
h 0 Þn /
n D 1; : : : ; 4
(7.224)
are illustrated in Figure 7.51. We note that, for the choice in (7.222) of the symbols probabilities, the decision thresholds, also shown in Figure 7.51, are obtained from the intersections between curves in (7.224) relative to two adjacent symbols; therefore they are given by the solutions of the M 1 equations pi pw .−i h 0 Þi / D pi C1 pw .−i h 0 Þi C1 / i D 1; : : : ; M 1 (7.225)
7.B. Computation of Pe for some cases of interest
603
pnpw(ρh oαn), n=1,2,3,4
τ
1
τ
2
τ
ρ
3
Figure 7.51. Optimum thresholds for a 4PAM system with nonequally likely symbols.
We point out that, if the probability that the symbol ` is sent is very small, p` − 1, the measure of the corresponding decision interval could be equal to zero, and consequently this symbol would never be detected. In this case the decision thresholds will be fewer than M 1. Example 7.B.2 (Computation of Pe for a 4PAM system) We indicate with Fw .x/ the probability distribution of w R;k : Z x pw .¾ / d¾ Fw .x/ D
1
(7.226)
For a MPAM system with thresholds −1 ; −2 , and −3 , the probability of correct decision is given by (6.18): 4 X Z pn pw .² h 0 Þn / d² P[C] D
nD1 Rn
Z
−1 1
D p1 Z C p3
pw .² pw .²
h 0 Þ1 / d² C p2
Z
−2 −1
pw .²
C1
h 0 Þ2 / d² (7.227) h 0 Þ4 / d² h 0 Þ2 /] Fw .−3 h 0 Þ4 /]
−3 −2
h 0 Þ3 / d² C p4
Z
−3
pw .²
D p1 [Fw .−1 C p3 [Fw .−3
h 0 Þ1 /] C p2 [Fw .−2 h 0 Þ3 / Fw .−2
h 0 Þ2 /
Fw .−1
h 0 Þ3 /] C p4 [1
604
Chapter 7. Transmission over dispersive channels
We note that, if Fw is a continuous function, optimum thresholds can be obtained by equating to zero the derivative of the expression in (7.227) with respect to −1 ; −2 , and −3 . In the case of equally likely symbols and equidistant thresholds, i.e. −i D h 0 .2i equation (7.227) yields P[C] D 1 Â 2 1 1 M Ã Fw . h 0 / (7.229) M/ i D 1; : : : ; M 1 (7.228)
We note that (7.229) is in agreement with (6.122) obtained for Gaussian noise.
7.B.2
Pe in the presence of ISI
We consider MPAM transmission in the presence of ISI. We assume the symbols in (7.221) are equally likely and the decision thresholds are of the type given by (7.228). With reference to (7.65), the received signal at the decision point assumes the following expression: yk D h 0 ak C ik C w R;k where ik represents the ISI and is given by X ik D h i ak
i 6D0
(7.230)
i
(7.231)
and w R;k is Gaussian noise with statistical power ¦ 2 and statistically independent of the i.i.d. symbols of the message fak g. We examine various methods to compute the symbol error probability in the presence of ISI.
Exhaustive method
We refer to the case of 4PAM transmission with Ni D 2 interferers due to one nonzero precursor and one nonzero postcursor. Therefore we have ik D ak
1h1
C akC1 h
1
(7.232)
We deﬁne the vector of symbols that contribute to ISI as a0 D [ak k Then ik can be written as a function of a 0 as k ik D i.a0 / k (7.234)
1 ; akC1 ]
(7.233)
Therefore, ik is a random variable that assumes values in an alphabet with cardinality L D M Ni D 16.
7.B. Computation of Pe for some cases of interest
605
Starting from (7.230) the error probability can be computed by conditioning with respect to the values assumed by a0 D [Þ .1/ ; Þ .2/ ] D α 2 A2 . For equally likely symbols and k thresholds given by (7.228) we have Ã Ã Â Â h 0 i.a0 / 1 X k P[a0 D α] Pe D 2 1 Q k M ¦ α2A2 (7.235) Ã Ã Â Â 1 1 X h 0 i.α/ D2 1 Q M L ¦ 2
α2A
This method gives the exact value of the error probability in the presence of interferers, but requires the computation of L terms. This method can be costly, especially if the number of interferers is large: it is therefore convenient to consider approximations of the error probability obtained by simpler computational methods.
Gaussian approximation
If interferers have a similar amplitude and their number is large, we can use the central limit theorem and approximate ik as a Gaussian random variable. As the process w R;k is Gaussian, the process z k D ik C w R;k is also Gaussian with variance
2 ¦z2 D ¦i C ¦ 2 2 where ¦i is given by (7.72). Then
(7.236)
(7.237) Â Ã (7.238)
Â Pe D 2 1
1 M
Ã Q
h0 ¦z
It is seen that this method, although very convenient, is rather pessimistic, especially for large values of 0. As a matter of fact, we observe that the amplitude of ik is limited by the value X imax D .M 1/ jh i j (7.239)
i 6D0
whereas the Gaussian approximation implies that the values of ik are unlimited.
Worstcase bound
This method substitutes ik with the constant imax deﬁned in (7.239). In this case Pe is equal to Ã Â Ã Â h 0 imax 1 Q (7.240) Pe D 2 1 M ¦ This bound is typically too pessimistic, however, it yields a good approximation if ik is mainly due to one dominant interferer.
606
Chapter 7. Transmission over dispersive channels
Saltzberg bound
With reference to (7.230), deﬁning z k as the total disturbance given by (7.236), in general we have Ã Â 1 P[z k > h 0 ] Pe D 2 1 (7.241) M Let Þmax D maxfÞn g D M
n
1
(7.242)
in the speciﬁc case, and I be any subset of the integers Z 0 , excluding zero, such that X
i 2I
jh i j <
h0 Þmax
(7.243)
Moreover, let I C be the complementary set of I with respect to Z 0 . Saltzberg applied a Chernoff bound to the probability P[z k > h 0 ] [10], obtaining 0 B B B P[z k > h 0 ] < exp B B B @ !2 1 h 0 Þmax jh i j C C C i 2I 0 1C C C X 2 2 2A A 2 @¦ C ¦a jh i j X
i 2I C
(7.244)
The bound is particularly simple in the case of binary signaling, where fak g 2 f 1; 1g, 0 B B B Pe < exp B B B @ !2 1 C h0 jh i j C C i 2I 0 1C C C X 2A A @¦ 2 C 2 jh i j X
i 2I C
(7.245)
P where I is such that i 2I jh i j < h 0 . In this case it is rather simple to choose the set I so that the limit is tighter. We begin with I D Z 0 . Then we remove from I one by one the indices i that correspond to the larger values of jh i j; we stop when the exponent of (7.245) has reached the minimum. Considering the limit of the function Q given by (6.364), we observe that for I D Z 0 and I C D ;, the bound in (7.244) practically coincides with the worstcase limit in (7.240). Taking instead I D ; and I C D Z 0 we obtain again the limit given by the Gaussian approximation for z k that yields (7.238). For the mathematical details we refer to [10]; for a comparison between the Saltzberg bound and other bounds we refer to [5, 11].
7.B. Computation of Pe for some cases of interest
607
GQR method
The GQR method is based on a technique for the approximate computation of integrals called Gauss quadrature rule (GQR). It offers a good compromise between computational complexity and approximation accuracy. If we assume a very large number of interferers, to the limit inﬁnite, ik can be modelled as a continuous random variable. Then Pe assumes the expression Ã Z C1 Â Ã Ã Â Â 1 h0 ¾ 1 pik .¾ / d¾ D 2 1 I (7.246) Q Pe D 2 1 M ¦ M 1 By the GQR method we obtain an approximation of the integral, given by I D
Nw X jD1
Â wj Q
h0 ¦
¾j
Ã (7.247)
In this expression the parameters f¾ j g and fw j g are called, respectively, abscissae and weights of the quadrature rule, and are obtained by a numerical algorithm based on the ﬁrst 2Nw moments of ik . The quality of the approximation depends on the choice of Nw [5].
608
Chapter 7. Transmission over dispersive channels
Appendix 7.C
General scheme
Coherent PAMDSB transmission
For transmission over a passband channel, a PAM signal must be suitably shifted in frequency by a sinusoidal carrier at frequency f 0 . This task is achieved by DSB modulation (see Example 1.6.3 on page 41) of the signal s.t/ at the output of the baseband PAM modulator ﬁlter. In the case of a coherent receiver, the passband scheme is given in Figure 7.52. For the baseband equivalent model, we refer to Figure 7.53a. Now we consider the study of the PAMDSB transmission system in the uniﬁed framework of Figure 7.12. Assuming the receive ﬁlter g Rc realvalued, we apply the operator Re [ ] to the channel ﬁlter impulse response and to the noise signal, and we split the factor 1=2 evenly among the channel ﬁlter and the receive ﬁlter responses; setting 1 g Rc .t/ D g Rc .t/ p , we thus obtain the simpliﬁed scheme of Figure 7.53b, where the noise 2 signal contains only the inphase component w0I .t/ with PSD Pw0I . f / D and " gC .t/ D Re or, in the frequency domain, GC . f / D e
j .'1 '0 / G . f Ch
N0 (V2 /Hz) 2 #
(7.248)
e
j .'1 '0 / g .bb/ .t/ Ch
p 2 2
(7.249)
C f 0 /1. f C f 0 / C e j .'1 p 4 2
'0 / G Ł . Ch
f C f 0 /1.
f C f0 / (7.250)
For a noncoherent receiver we refer to the scheme developed in Example 6.11.6 on page 516.
Figure 7.52. PAMDSB passband transmission system.
7.C. Coherent PAMDSB transmission
609
Figure 7.53. PAMDSB system.
Transmit signal PSD
Considering the PSD of the message sequence, the average PSD of the modulated signal s.t/ is given by (7.28), 1 N Ps . f / D [Pa . f 4T 2 f 0 / jHT x . f f 0 /j2 C Pa . f C f 0 / jHT x . f f 0 /j2 ] (7.251)
Consequently the transmitted signal bandwidth is equal to twice the bandwidth of h T x . The minimum bandwidth is given by Bmin D 1 T (7.252)
Recalling the deﬁnition (6.103), the spectral efﬁciency of the transmission system is given by ¹ D log2 M (bit/s/Hz) which is halved with respect to MPAM (see Table 6.9). (7.253)
Signaltonoise ratio
We assume the function e
j .'1 '0 / g .bb/ .t/ Ch
p 2 2
(7.254)
is realvalued; then from Figure 7.53a, using (1.295), we have the following relation: E[jsC .t/j2 ] D
.bb/ E[jsCh .t/j2 ] D E[jsCh .t/j2 ] 2
(7.255)
610
Chapter 7. Transmission over dispersive channels
Setting qC .t/ D h T x Ł gC .t/ from (6.105) and (7.252) we have 0D where, for an MPAM system (6.110), Ma D M2 1 3 (7.258) Ma E qC N0 (7.257) (7.256)
In the absence of ISI, for deﬁned in (7.106), (7.107) still holds; moreover, using (7.257), for a matched ﬁlter receiver, (7.113) yields
MF
D
E qC 20 D N0 =2 Ma r Q ! 1
(7.259)
Then the error probability is given by Â Pe D 2 1 1 M Ã 60 M2 (7.260)
We observe that the performance of an MPAMDSB system and that of an MPAM system are the same, in terms of Pe as a function of the received power. However, because of DSB modulation, the required bandwidth is doubled with respect to both baseband PAM transmission and PAMSSB modulation.14 This explains the limited usage of PAMDSB for digital transmission.
14 The PAMSSB scheme presents in practice considerable difﬁculties because the ﬁlter for modulation is non
ideal: in fact, this causes distortion of the signal s.t/ at low frequencies that may be compensated for only by resorting to line coding (see Appendix 7.A).
7.D. Implementation of a QAM transmitter
611
Appendix 7.D
Implementation of a QAM transmitter
Three structures, which differ by the position of the digitaltoanalog converter, may be considered for the implementation of a QAM transmitter. In Figure 7.54 the modulator employs for both inphase and quadrature signals a DAC after the interpolator ﬁlter h T x , followed by an analog mixer that shifts the signal to passband. This scheme works if the sampling frequency 1=Tc is much greater than twice the bandwidth B of h T x . For applications where the symbol rate is very high, the DAC is placed right after the bit mapper and the various ﬁlters are analog (see Chapter 19). In the implementation illustrated in Figure 7.55, the DAC is placed instead at an intermediate stage with respect to the case of Figure 7.54. Samples are premodulated by a digital mixer to an intermediate frequency f 1 , interpolated by the DAC and subsequently remodulated by a second analog mixer that shifts the signal to the desired band. The intermediate frequency f 1 must be greater than the bandwidth B and smaller than 1=.2Tc / B, thus avoiding overlap among spectral components. We observe that this scheme requires only one DAC, but the sampling frequency must be at least double as compared to the previous scheme.
Figure 7.54. QAM with analog mixer.
Figure 7.55. QAM with digital and analog mixers.
612
Chapter 7. Transmission over dispersive channels
Figure 7.56. Polyphase implementation of the ﬁlter hTx for Tc D T=8.
For the ﬁrst implementation, as the system is typically oversampled with a sampling interval Tc D T =4 or Tc D T =8, the frequency response of the DAC, G I . f /, may be considered as a constant in the passband of both the inphase and quadrature signals. For the second implementation, unless f 1 − 1=Tc , the distortion introduced by the DAC should be considered and equalized by one of these methods (see page 338): ž including the compensation for G I . f / in the frequency response of the ﬁlter h T x , ž inserting a digital ﬁlter before the DAC, ž inserting an analog ﬁlter after the DAC. We recall that an efﬁcient implementation of interpolator ﬁlters h T x is obtained by the polyphase representation, as shown in Figure 7.56 for Tc D T =8, where Â Ã T h .`/ .m/ D h T x mT C ` ` D 0; 1; : : : ; 7 m D 1; : : : ; C1 (7.261) 8 To implement the scheme of Figure 7.56, once the impulse response is known, it may be convenient to precompute the possible values of the ﬁlter output and store them in a table or RAM. The symbols fak;I g are then used as pointers for the table itself. The same approach may be followed to generate the values of the signals cos.2³ f 1 nTc / and sin.2³ f 1 nTc / in Figure 7.55, using an additional table and the index n as a cyclic pointer.
7.E. Simulation of a QAM system
613
Appendix 7.E
Simulation of a QAM system
In Figure 7.12 we consider the baseband equivalent scheme of a QAM system. The aim is to simulate the various transformations in the discretetime domain and to estimate the bit error probability. This simulation method, also called Monte Carlo, is simple and general because it does not require any special assumption on the processes involved; however, it is intensive from the computational point of view. For alternative methods, for example semianalytical, to estimate the error probability, we refer to speciﬁc texts on the subject [12]. We describe the various transformations in the overall discretetime system depicted in Figure 7.57, where the only difference with respect to the scheme of Figure 7.12 is that the
(a) Transmitter and channel block diagram.
(b) Receiver block diagram.
Figure 7.57. Baseband equivalent model of a QAM system with discretetime ﬁlters and sampling period TQ D T=Q0 . At the receiver, in addition to the general scheme, a multirate structure to obtain samples of the received signal at the timing phase t0 is also shown.
614
Chapter 7. Transmission over dispersive channels
ﬁlters are discretetime with quantum TQ D T =Q 0 , Which is chosen to accurately represent the various signals. Binary sequence fb` g. The sequence fb` g is generated as a random sequence or as a PN sequence (see Appendix 3.A), and has length K . Bit mapper. The bit mapper maps patterns of information bits to symbols; the symbol constellation depends on the modulator (see Figure 7.6 for two constellations). Interpolator ﬁlter h T x from period T to TQ . The interpolator ﬁlter is efﬁciently implemented by using the polyphase representation (see Appendix 1.A). For a bandlimited pulse of the raised cosine or square root raised cosine type, the maximum value of TQ , submultiple of T , is T =2. In any case, the implementation of ﬁlters, for example, the ﬁlter representing the channel, and nonlinear transformations, for example, the transformation due to a power ampliﬁer operating near saturation (not considered in Figure 7.57), typically require a larger bandwidth, leading, for example, to the choice TQ D T =4 or T =8. In the following examples we choose TQ D T =4. For the design of h T x the window method can be used (Nh odd): ÄÂ Ã ½ Nh 1 TQ w Nh .q/ q D 0; 1; : : : ; Nh 1 (7.262) h T x .q TQ / D h i d q 2 where typically w Nh is the discretetime rectangular window or the Hamming window, and h i d is the ideal impulse response. Frequency responses of h T x are illustrated in Figure 7.58 for h i d square root raised cosine pulse with rolloff factor ² D 0:3, and w Nh rectangular window of length Nh , for various values of Nh (TQ D T =4). The corresponding impulse responses are shown in Figure 7.59. Transmission channel. For a radio channel the discretetime model of Figure 4.35 can be used, where in the case of channel affected by fading, the coefﬁcients of the FIR ﬁlter that model the channel impulse response are random variables with a given power delay proﬁle. For a transmission line the discretetime model of (4.150) can be adopted. We assume the statistical power of the signal at output of the transmission channel is given by MsCh D MsC . Additive white Gaussian noise. Let w I .q TQ / and w Q .q TQ / be two Gaussian statistically N N independent r.v.s, each with zero mean and variance 1=2, generated according to (1.655). To generate the complexvalued noise signal fwC .q TQ /g with spectrum N0 , it is sufﬁcient to use the relation wC .q TQ / D ¦wC [w I .q TQ / C j w Q .q TQ /] N N (7.263)
7.E. Simulation of a QAM system
615
0
N = 17 h N = 25 h N = 33
h
−10
−20
 HT (f)  (dB)
−30
x
−40
−50
−60
0
0.2
0.4
0.6
0.8
1 fT
1.2
1.4
1.6
1.8
2
Figure 7.58. Magnitude of the transmit ﬁlter frequency response, for a windowed square root raised cosine pulse with rolloff factor ² D 0:3, for three values of Nh (TQ D T=4).
0.3
h (q T )
Nh=17
Q Tx
0.2
0.1
0
−0.1
−5
0
5
q / TQ
10
15
20
0.3
h (q T )
Nh=25
Q Tx
0.2
0.1
0
−0.1
0
5
10
q / TQ
15
20
25
0.3
h (q T )
Nh=33
Q Tx
0.2
0.1
0
−0.1
0
5
10
15 q / TQ
20
25
30
Figure 7.59. Transmit ﬁlter impulse response, fhTx .qTQ /g, q D 0; : : : ; Nh 1 , for a windowed square root raised cosine pulse with rolloff factor ² D 0:3, for three values of Nh (TQ D T=4).
616
Chapter 7. Transmission over dispersive channels
where
2 ¦wC D N0
1 TQ
(7.264)
Usually the signaltonoise ratio 0 given by (6.105) is given. For a QAM system, from (7.51) and (7.55) we have MsC MsC D 2 (7.265) 0D N0 .1=T / ¦wC .TQ =T / The standard deviation of the noise to be inserted in (7.263) is given by r MsC Q 0 ¦wC D (7.266) 0 We note that ¦wC is a function of MsC , of the oversampling ratio Q 0 D T =TQ , and of the given ratio 0. In place of 0, the ratio E b =N0 D 0= log2 M may be assigned. Receive ﬁlter. As will be discussed in Chapter 8, there are several possible solutions for the receive ﬁlter. The most common choice is a matched ﬁlter g M , matched to h T x , of the square root raised cosine type. Alternatively, the receive ﬁlter may be a simple antialiasing FIR ﬁlter g A A , with passband at least equal to that of the desired signal. The ﬁlter attenuation in the stopband must be such that the statistical power of the noise evaluated in the passband is larger by a factor of 5–10 with respect to the power of the noise evaluated in the stopband, so that we can ignore the contribution of the noise in the stopband at the output of the ﬁlter g A A . If we adopt as bandwidth of g A A the Nyquist frequency 1=.2T /, the stopband of an ideal ﬁlter with unit gain goes from 1=.2T / to 1=.2TQ /: therefore the ripple Žs in the stopband must satisfy the constraint 1 N0 2T Â 1 Žs N0 2T Q from which we get the condition Žs < 10 Q0
1
1 2T
Ã > 10
(7.267)
1
(7.268)
Usually the presence of other interfering signals forces the selection of a value of Žs that is smaller than that obtained in (7.268). Interpolator ﬁlter. The interpolator ﬁlter is used to increase the sampling rate from 1=TQ 0 to 1=TQ : this is useful when TQ is insufﬁcient to obtain the accuracy needed to represent the timing phase t0 . This ﬁlter can be part of g M or g A A . From Appendix 1.A, the efﬁcient 0 0 implementation of fg M . pTQ /g is obtained by the polyphase representation with TQ =TQ branches. To improve the accuracy of the desired timing phase, further interpolation, for example, linear, may be employed.
7.E. Simulation of a QAM system
617
Timing phase. Assuming a training sequence is available, for example, of the PN type fa0 D p.0/; a1 D p.1/; : : : ; a L 1 D p.L 1/g, a simple method to determine t0 is to choose the timing phase in relation to of the peak of the overall impulse response. Let 0 fx. pTQ /g be the signal before downsampling. If we evaluate
0 m opt D arg max jrxa .mTQ /j m
þ þ L 1 þ1 X þ þ þ 0 D arg max þ x.`T C mTQ / pŁ .`/þ m þL þ `D0 then
0 t0 D m opt TQ 0 m min TQ 0 m max TQ
(7.269)
0 0 0 m min TQ < mTQ < m max TQ
(7.270)
In (7.269) and are estimates of minimum and maximum system delay, 0 respectively. Moreover, we note that the accuracy of t0 is equal to TQ and that the amplitude 0 /=r .0/. of the desired signal is h 0 D rxa .m opt TQ a Downsampler. The sampling period after downsampling is usually T or Tc D T =2, with timing phase t0 . The interpolator ﬁlter and the downsampler can be jointly implemented, 0 according to the scheme of Figure 1.81. For example, for TQ D T =4, TQ D T =8, and Tc D 0 T =2 the polyphase representation of the interpolator ﬁlter with output fx. pTQ /g requires two branches. Also the polyphase representation of the interpolatordecimator requires two branches. Equalizer. After downsampling, the signal is usually input to an equalizer (LE, FSE or DFE, see Chapter 8). The output signal of the equalizer has always sampling period equal to T . As observed several times, to decimate simply means to evaluate the output at the desired instants. Data detection. The simplest method resorts to a threshold detector, with thresholds determined by the constellation and the amplitude of the pulse at the decision point. Viterbi algorithm. An alternative to the threshold detector, which operates on a symbol by symbol basis, is represented by maximum likelihood sequence detection by the Viterbi algorithm (see Chapter 8). Inverse bit mapper. The inverse bit mapper performs the inverse function of the bit mapper. It translates the detected symbols into bits that represent the recovered information bits. Simulations are typically used to estimate the bit error probability of the system, for a certain set of values of 0. We recall that caution must be taken at the beginning and at the N end of a simulation to consider transients of the system. Let K be the number of recovered bits. The estimate of the bit error probability Pbit is given by number of bits received with errors O Pbit D N number of received bits, K (7.271)
618
Chapter 7. Transmission over dispersive channels
N O It is known that as K ! 1, the estimate Pbit has a Gaussian probability distribution N N with mean Pbit and variance Pbit .1 Pbit /= K . From this we can deduce, by varying K , O the conﬁdence interval [P ; PC ] within which the estimate Pbit approximates Pbit with an assigned probabilty, that is O P[P Ä Pbit Ä PC ] D Pconf (7.272)
N For example, we ﬁnd that with Pbit D 10 ` and K D 10`C1 , we have a conﬁdence interval O of about a factor 2 with a probability of 95%, that is P[1=2Pbit Ä Pbit Ä 2Pbit ] ' 0:95. This is in good agreement with the experimental rule of selecting N K D 3 Ð 10`C1 (7.273)
For a channel affected by fading, the average Pbit is not very signiﬁcant: in this case it is meaningful to compute the distribution of Pbit for various channel realizations. In pratice N we assume the transmission of a sequence of N p packets, each one with K p information bits N to be recovered: typically K p D 1000–10000 bits and N p D 100–1000 packets. Moreover, the channel realization changes at every packet. For a given average signaltonoise ratio O N 0 (see (6.347)), the probability Pbit .n p /, n p D 1; : : : ; N p is computed for each packet. As O a performance measure we use the percentage of packets with Pbit .n p / < Pbit , also called bit error probability cumulative distribution function (cdf), where Pbit assumes values in a certain set. This performance measure is more signiﬁcant than the average Pbit evaluated for a very N long, continuous transmission of N p K p information bits. In fact the average Pbit does not show that, in the presence of fading, the system may occasionally have a very large Pbit , and consequently an outage.
Chapter 8
Channel equalization and symbol detection
With reference to PAM and QAM systems, in this chapter we will discuss several methods to compensate for linear distortion introduced by the transmission channel. Next, as an alternative to a memoryless threshold detector, we will analyze detection methods that operate on sequences of samples. Recalling the analysis of Section 7.3, we ﬁrst review three techniques relying on the zeroforcing ﬁlter, linear equalizer, and DFE, respectively, that attempt to reduce the ISI in addition to maximizing the ratio deﬁned in (7.106).
8.1
Zeroforcing equalizer (LEZF)
From the relation (7.66), assuming that HTx . f / and GC . f / are known, and H. f / is given, for example, by (7.84), then the equation H. f / D Q R . f /e j2³ f t0 D HTx . f /GC . f /GRc . f /e j2³ f t0 can be solved with respect to the receive ﬁlter, yielding Ã Â f ;² T rcos 1=T GRc . f / D e HTx . f /GC . f / (8.1)
j2³ f t0
(8.2)
From (8.2), the magnitude and phase responses of GRc can be obtained. In practice, however, although the condition (8.2) leads to the suppression of the ISI, hence the ﬁlter gRc is called linear equalizer zeroforcing (LEZF), it may also lead to the enhancement of the noise power at the decision point, as expressed by (7.75). In fact, if the frequency response GC . f / exhibits strong attenuation at certain frequencies in the range [ .1 C ²/=.2T /; .1 C ²/=.2T /], then GRc . f / presents peaks that determine a 2 large value of ¦w R . In any event, the choice (8.2) guarantees the absence of ISI at the decision point, and from (7.109) we get
LE ZF
D
2 N0 E gRc
(8.3)
620
Chapter 8. Channel equalization and symbol detection
Obviously, based on the considerations of Section 7.3, it is
LE ZF
Ä
MF
(8.4)
where MF is deﬁned in (7.113). In the particular case of an ideal channel, that is GCh . f / D G0 in the passband of the system, and assuming h Tx is given by s Ã Â f (8.5) ;² HTx . f / D T rcos 1=T then from (7.42) (8.6) p where from (7.38), k1 D G0 for a PAM system, whereas k1 D .G0 = 2/e j .'1 '0 / for a QAM system. Moreover, from (8.2), neglecting a constant delay, i.e. for t0 D 0, it results that s Ã Â 1 f (8.7) ;² GRc . f / D rcos k1 1=T In other words gRc .t/ is matched to qC .t/ D k1 h Tx .t/, and
LE ZF
QC . f / D HTx . f / GC . f / D k1 HTx . f /
D
MF
(8.8)
Methods for the design of a LEZF ﬁlter with a ﬁnite number of coefﬁcients are given in Section 8.7.
8.2
Linear equalizer (LE)
We introduce an optimization criterion for GRc that takes into account the ISI as well as the statistical power of the noise at the decision point.
8.2.1
Optimum receiver in the presence of noise and ISI
With reference to the scheme of Figure 7.12 for a QAM system, the criterion of choosing the receive ﬁlter such that the signal yk is as close as possible to ak in the meansquare sense is widely used.1 Let h Tx and gC be known. Deﬁning the error ek D a k yk yk j2 ] (8.9)
the receive ﬁlter gRc is chosen such that the meansquare error J D E[jek j2 ] D E[jak is minimized.
1
(8.10)
It would be desirable to ﬁnd the ﬁlter such that P[ak 6D ak ] is minimum. This problem, however, is usually O very difﬁcult to solve. Therefore we resort to the criterion of minimizing E[jyk ak j2 ] instead.
8.2. Linear equalizer (LE)
621
The following assumptions are made: 1. the sequence fak g is wide sense stationary (WSS) with spectral density Pa . f /; 2. the noise wC is complexvalued and WSS. In particular we assume it is white with spectral density PwC . f / D N0 ; 3. the sequence fak g and the noise wC are statistically independent. The minimization of J in this situation differs from the classical problem of the optimum Wiener ﬁlter because h Tx and gC are continuoustime pulses. By resorting to the calculus of variations (see Appendix 8.A), we obtain the general solution GRc . f / D
Ł QC . f / e N0 j2³ f t0
Pa . f / 1 T C Pa . f / T
C1 X `D 1
þ Â 1 þ þQC f N0 þ
Ãþ ` þ2 þ T þ
(8.11)
where QC . f / D HTx . f /GC . f /. Considerations on the joint optimization of the transmit and receive ﬁlters are discussed in Appendix 8.A. 2 If the symbols are statistically independent and have zero mean, then Pa . f / D T ¦a , and (8.11) becomes:
Ł GRc . f / D QC . f /e j2³ f t0 2 ¦a þ Â C1 1 X þ 2 þQC f N0 C ¦a T `D 1 þ
Ãþ ` þ2 þ T þ
(8.12)
The expression of the cost function J in correspondence of the optimum ﬁlter (8.12) is given in (8.40). From the decomposition (7.62) of GRc . f /, in (8.12) we have the following correspondences:
Ł G M . f / D QC . f /e j2³ f t0
(8.13)
and C.e j2³ f T / D
2 ¦a þ Â C1 1 X þ 2 þQC f N0 C ¦a T `D 1 þ
Ãþ ` þ2 þ T þ
(8.14)
The optimum receiver thus assumes the structure of Figure 8.1. We note that g M is the matched ﬁlter to the impulse response of the QAM system at the receiver input.2 The ﬁlter c is called linear equalizer (LE). It attempts to ﬁnd the optimum tradeoff between removing the ISI and enhancing the noise at the decision point.
2
As derived later in the text (see Observation 8.13 on page 681) the output signal of the matched ﬁlter, sampled at the modulation rate 1=T , forms a “sufﬁcient statistic” if all the channel parameters are known.
that is if jQC . q is the overall impulse response of the system at the sampler input: q. x k . O Ł However. With reference to the scheme of Figure 8. Channel equalization and symbol detection Figure 8. is very interesting from a theoretical point of view. D.t t0 //. In the absence of noise. ž ﬁlter output signal. 1. 2. In the absence of ISI at the output of g M . expresses in number of symbol intervals the delay introduced by the equalizer.t/ D rqC .e.1 and for any type of ﬁlter g M . it is possible to determine the coefﬁcients of the FIR equalizer ﬁlter c using the Wiener formulation.t/ ' 0. not necessarily matched. We analyze two particular cases of the solution (8. i.16) .622 Chapter 8.t t0 / (8. We notice the presence of the parameter D that denotes the lag of the desired signal: this parameter.1.15) is the linear equalizer zeroforcing.e j2³ f T / is constant and the equalizer can be removed. with the following deﬁnitions: ž ﬁlter input signal. ek D dk yk .15) Note that the system is perfectly equalized. for which g M . i.t/ D qC Ł g M . Alternative derivation of the IIR equalizer Starting from the receiver of Figure 8. which must be suitably estimated. f /j2 is a Nyquist pulse. there is no ISI. the particular case of a matched ﬁlter.t/ D h Tx Ł gC Ł g M .t/ D qC . wC .e j2³ f T / D 1 Â C1 þ X þ 1 þQC f T `D 1 þ Ãþ ` þ2 þ T þ (8. In this case the ﬁlter (8. as it completely eliminates the ISI. Optimum receiver structure for a channel with additive white noise. it may be IIR. and C. . We assume that the ﬁlter c may have an inﬁnite number of coefﬁcients. yk . The overall delay from the emission of ak to the generation of the detected symbol ak is equal to t0 C DT seconds. ž desired output signal.e.12). dk D ak ž estimation error.2a. then C.
f /j2 (8.t/ D qC .t0 tqC .0.− / D N0 rqC . f / D jQC .2a.t/ is given by PqC .t0 / is taken in relation to the maximum value of jq. the desired sample q.− / D rwC Ł rqC .2. we have w R .20) . In any case from (8. Hence.t/ D [qC .t0 t/ has support .− / (8. Linear equalizer (LE) 623 Figure 8.t 0 / Ł qC .t/ with autocorrelation function given by rw R . t0 /.17) The Fourier transform of rqC . to obtain a causal ﬁlter g M the minimum value of t0 is tqC .16). then g M . In Figure 8.t/j.18) Ł We note that if qC has a ﬁnite support . where rqC is the autocorrelation of the deterministic pulse qC .t/ (8. assuming wC is white noise. tqC /.19) (8. given by Ł rqC .2. as rqC is a correlation function.t/ D wC Ł g M . t 0 /]. Linear equalizer as a Wiener ﬁlter.8.
0/ D E qC The sequence fh n g has ztransform given by 8.z/ Px . from (8. f /j2 In Figure 8.nT /] D N0 8.z/ (8.25) (8. f / D N0 jQC .22) The discretetime equivalent model is illustrated in Figure 8.29) The Wiener solution that gives the optimum coefﬁcients is given in the ztransform domain by (2.z/ (8.nT / In particular.z/ D Z[r dx . using the properties of Table 1.90).z/8 Ł Ł (8.24) (8. the ztransform of the autocorrelation of the noise samples wk D w R .e j2³ f T / D F[h n ] D T `D 1 þ T þ (8.n/] (8.z/ D Z[h n ] D PqC .n/] D Z[rw R .26) 8.50): Copt .27) Moreover. the Fourier transform of (8.z/ D 8 Ł Ł z On the other hand.23) is given by Â Ãþ C1 þ ` þ2 1 X þ þ þQC f 8.23) Ł which. the correlation sequence of fh n g has ztransform equal to Â Ã 1 Z[rh .3.t0 C nT / D rqC .20).z/ D Z[c n ] D where Pdx .nT / D rqC . f / D N0 PqC .n/] and Px . it results in h 0 D rqC .21) ai h k i C wk Q (8.2b.t0 C Q kT / is given by: Z[rw .z/ Q (8. by the Hermitian symmetry of an autocorrelation sequence. sampling at instants tk D t0 C kT yields the sampled QAM signal xk D C1 X iD 1 (8.31) Pdx .2a. nT /.28) z Also. rqC . satisﬁes the relation: Â Ã 1 (8.m/] D 8.30) . Channel equalization and symbol detection Then the spectrum of w R is given by: Pw R . The discretetime overall impulse response is given by h n D q. from (1.z/ D Z[r x .624 Chapter 8.
the computation of the autocorrelation of the process fx k g yields (see also Table 1. Q Then the crosscorrelation between fdk g and fx k g is given by Ł rdx .36) C N0 8.28).z/ (8.26).z/8 Ł Ã (8.35) Thus.2. Linear equalizer (LE) 625 We assume the following assumptions hold: 1.n/ Q (8.n/ D ¦a h Ł D n (8. using (8. fak g and fwk g are statistically independent and hence uncorrelated.8. from (8.n/ C rw .z/ D 1 2 Ł C N0 8. Ã 1 z D zŁ Â Ã Copt .z/ D 2 ¦a z D 2 N0 C ¦a 8.32) 2. (8. from assumption 1. 2 rdx . f / D T ¦a (8.6): Ł 2 rx .34) Under the same assumptions 1 and 2. 2 ra . with symbols that are statistically independent and with zero mean.z/ ¦a 8.38) . The sequence fak g is WSS.n/ D ¦a Žn 2 and Pa . Finally.z/8 zŁ 2 ¦a 8Ł Â (8.n/ D E[x k x k n ] D ¦a rh .37) is simpliﬁed as Copt .n/ D E[dk x k n ] " D E ak C1 X iD 1 D C1 X iD 1 !Ł # ai h k n i C wk Q n (8.z/ D ¦ a 8Ł Â 1 zŁ Ã z Â 1 zŁ D 2 Px .z/ D ¦ a 8.37) Taking into account the property (8.z/ Therefore.33) D hŁ k Ł n i E[ak D ai ] using assumption 2. we obtain 2 Pdx .30).
i/ dx (8.36) in (8.z//. Hence.40) may be computed by evaluating 2 2 the coefﬁcient of the term z 0 of the function ¦a N0 =.n is the impulse response of the optimum ﬁlter (8.N0 C ¦a 8.en Ł copt.h n Ł copt. or by using the partial fraction expansion method (see (1.39) yields Jmin D 2 ¦d Z T Z 1 2T 1 2T 1 2T 1 2T Ł Pdx .z/ is a rational function of z. the integral (8. (8.40) 2 D ¦a T N0 df 2 N0 C ¦a 8.14).39) Ł Pdx . f / C opt .53): 2 Jmin D ¦d N 1 X i D0 copt.43) We assume that in (8. Channel equalization and symbol detection It can be observed that. which accounts for a possible delay introduced by the equalizer. as i D .23) and copt.n /k w (8. At the decision point we have yk D D ak D C C1 X iD 1 i 6D D i ak D i C . and Jmin D 2 ¦a N0 2 N0 C ¦a E qC (8.42) where fh n g is given by (8.38). substitution of the relations (8. we determine the minimum value of the cost function.e j2³ f T / d f 2 D ¦d Z 1 2T 1 2T Finally.44) LE ¦I .626 Chapter 8.e j2³ f T / If 8. We note that in the absence of ISI.e j2³ f T /Copt . (7.z/ D h 0 D E qC .41) Signaltonoise ratio γ We deﬁne the overall impulse response at the equalizer output. which can be obtained by series expansion of the integrand.43) the total disturbance given by ISI plus noise is modeled as Gaussian noise with variance 2¦ I2 .e j2³ f T / d f (8. for z D e j2³ f T . In relation to the optimum ﬁlter C opt . sampled with a sampling rate equal to the modulation rate 1=T .z/. apart from the term z D . at the output of the MF we get 8.38) corresponds to (8. We recall the general expression for the Wiener ﬁlter (2. for a minimum distance among symbols of the constellation equal to 2.n /i (8.i rŁ .131)).106) yields Ã2 Â D D (8.
Two alternative approaches are usually considered. the ﬁlter g M may be designed according to the average characteristics of the channel.43) coincides with ek . The receiver is represented in Figure 8.45) where Jmin is given by (8.40).4. in real time. LE with a ﬁnite number of coefﬁcients 627 In case the approximation D ' 1 holds. . it is necessary to design a receiver that tries to identify the channel characteristics and at the same time to equalize it through suitable adaptive algorithms.3. otherwise.3 LE with a ﬁnite number of coefﬁcients In practice.3. Second solution.  Figure 8. the total disturbance in (8. if it is possible to rely on some a priori knowledge of the channel. Therefore the equalization task is left to the ﬁlter c. 8.44) becomes LE ' 2 Jmin (8. because the noise wC Figure 8. with F0 ½ 2.8.4.3. In particular if the desired signal sC has a bandwidth B and x is sampled with period Tc D T =F0 . The matched ﬁlter g M is designed assuming an ideal channel. Moreover. hence 2¦ I2 ' Jmin . to equalize the channel by adapting its coefﬁcients to the channel variations. where F0 is the oversampling index. The classical block diagram of an adaptive receiver is shown in Figure 8. The ﬁlter c is then an adaptive transversal ﬁlter that attempts. then the passband of gAA should extend at least up to frequency B. if the channel is either unknown a priori or it is time variant. First solution. The antialiasing ﬁlter gAA is designed according to speciﬁcations imposed by the sampling theorem. and (8. Receiver implementation by an analog matched ﬁlter followed by a sampler and a discretetime linear equalizer. Receiver implementation by discretetime ﬁlters.
and the output signal yk is deﬁned over a discretetime domain with period T . to equalize the channel.5. which employs the Wiener formulation and requires the computation of the matrix R and the vector p. Adaptive LE We analyze now the solution illustrated in Figure 8. g A A should also attenuate the noise components outside the passband of the desired signal sC .5. The description of the direct method is postponed to Section 8. 3. hence the cutoff frequency of gAA is between B and F0 =. two strategies may be used to determine an equalizer ﬁlter c with N coefﬁcients: 1. 2. to ﬁlter the residual noise outside the passband of the desired signal sC .t/jtDt0 CnT (8. Thus the discretetime ﬁlter c needs to accomplish the following tasks: 1.2 on page 641).t0 C nT / D h Tx Ł gC Ł g M . Discretetime equivalent scheme associated with the implementation of Figure 8.4 is implemented as a decimator ﬁlter (see Appendix 1. where fh n g is the discretetime impulse response of the overall system. to act as a matched ﬁlter.2T /. to simplify the implementation of the ﬁlter gAA . the adaptive method. which we will describe next (see Chapter 3). the direct method. it is convenient to consider a wide transition band. Channel equalization and symbol detection is considered as a wideband signal.3: the discretetime equivalent scheme is illustrated in Figure 8. given by h n D q.A). . 2. Note that the ﬁlter c of Figure 8.46) Figure 8.628 Chapter 8. In turn. In practice.5 (see Observation 8.t0 C nTc / is deﬁned over a discretetime domain with period Tc D T =F0 . where the input signal xn D x.3.
Deﬁne the performance measure of the system. the automatic identiﬁcation of the channel characteristics takes place.48) 2.t/. L TS C D 1 (8. is transmitted. 1. c N 1. k D 0. The MSE criterion is typically adopted: J . LE with a ﬁnite number of coefﬁcients 629 and wk D w R . The design strategy consists of the following steps.k ] (8.52) 3.50) b) coefﬁcient vector T ck D [c0. x k N C1 ] (8.3 on page 641).53) Evaluation of the error in training mode is possible if a sufﬁciently long sequence of L TS symbols known at the receiver. for an FIR ﬁlter c with N coefﬁcients using the LMS algorithm (see Section 3.t/ D wC Ł g M . : : : .A). fak g.51) c) adaptation gain 0<¼< 2 N rx .47) with w R . Select the law of coefﬁcient update. .8.0/ (8. : : : . x k 1 .k . 1. As the spectrum of the training sequence must be wide.49) (8. allowing the computation of the optimum coefﬁcients of the equalizer ﬁlter c.2) we have ckC1 D ck C ¼ek xŁ k where a) input vector T xk D [x k . L TS 1. a) Training mode ek D a k D yk k D D. The duration of the transmission of TS is equal to L TS T . : : : . called training sequence (TS).3. and consequently channel equalization.1. For example. c1.k/ D E[jek j2 ] (8.t0 C kT / Q (8. To evaluate the signal error ek to be used in the adaptive algorithm we distinguish two modes. During this time interval. typically a PN sequence is used (see Appendix 3. We note that even the direct method requires a training sequence to determine the vector p and the matrix R (see Observation 8. : : : .k .
6. Therefore ak ' ak . The implementation of the above equations is illustrated in Figure 8.t/jtDt0 CnTc Q (8. 8.6.55) .54) Once the transmission of the TS is completed.t0 C nTc / D wC Ł gAA .t/ D h Tx Ł gC Ł gAA . In (8. Channel equalization and symbol detection Figure 8.57) (8.53) we then substitute the known transmitted symbol with the detected symbol to obtain (8.t/ The noise is given by wn D w R . we assume that the equalizer has arrived at convergence. b) Decision directed mode O ek D a k D yk k ½ L TS C D (8. Linear adaptive equalizer implemented as a transversal ﬁlter with N coefﬁcients.4 Fractionally spaced equalizer (FSE) We consider the receiver structure with oversampling illustrated in Figure 8.7. and the transmission of information symbols may O start. shown in Figure 8. has impulse response given by h i D q.4.54).t0 C i Tc / where q.630 Chapter 8. The discretetime overall system.56) (8.
62) In Section 7. If F0 is an integer.8. Let h.61) 0 As mentioned earlier. we recall the Nyquist problem.t/: s R .t nT / t 2< (8. Before analyzing this system.4.t/ D C1 X nD 1 an h. in a practical implementation of the ﬁlter the sequence fyn g is not explicitly generated.58) ci xn i (8. Let us consider a QAM system with pulse h. c y’n Tc F0 yk T ^ akD T For the analysis. t 2 <. only the sequence fyk g is produced (see Appendix 1. n integer. the discretetime pulse satisﬁes the Nyquist criterion if h.3 we considered continuoustime Nyquist pulses h. However. introducing the downsampler helps to illustrate the advantages of operating with an oversampling index F0 > 1.7. we have 0 yk D yk F0 (8.t/ be deﬁned now on a discretetime domain fnTc g. Fractionally spaced linear equalizer (FSE). and the Nyquist conditions . for all integers ` 6D 0. Fractionally spaced equalizer (FSE) 631 {hi =q(t0 +iTc )} ak T h xn Tc ~ wn Figure 8.0/ 6D 0.t/. the ﬁlter c is decomposed into a discretetime ﬁlter with sampling period Tc that is cascaded with a downsampler.A). where Tc D T =F0 . is given by i D h Ł ci (8.`F0 Tc / D 0.60) 0 If we denote by fyn g the sequence of samples at the ﬁlter output. and h. In the particular case F0 D 1 we have Tc D T . the nth sample of the sequence is given by xn D The output of the ﬁlter c is given by 0 yn D N 1 X i D0 C1 X kD 1 hn k F0 ak C wn Q (8. deﬁned on the discretetime domain with sampling period Tc . The input signal to the ﬁlter c is the sequence fxn g with sampling period Tc D T =F0 .59) We note that the overall impulse response at the ﬁlter output. and by fyk g the downsampled sequence.
2T / we have Q Â 1 2T Ã D Ae j' and Q Â 1 2T Ã D Ae j' (8.8. using (8.e j2³ f Tc /H .nT / D 0.60). Recalling the input–output downsampler relations in the frequency domain. a pulse of the type shown in Figure 8.632 Chapter 8.63) The task of the equalizer is to yield a pulse f i g that approximates a Nyquist pulse. 1 2T 0 1 2T 1 T f h(t) H(e j2π fTc ) 0 T 2T t=nT (b) F0 D 1. In fact. Channel equalization and symbol detection h(t) H(e j2π fTc ) 0 T 2T t=nTc (a) F0 D 2.64) .e j2³ f Tc / D C. sampling the equalizer input signal with period equal to T .55).e.8. in the frequency domain a discretetime Nyquist pulse is equal to a constant.e j2³ f Tc / Â C1 X j2³ f Tc 1 D C.56). the QAM pulse deﬁned on the discretetime domain with period Tc is given by (8.e j2³ f Tc / assumes very small values at frequencies near f D 1=. We note that.t/ is deﬁned in (8.7. let us assume q.55). it is easy to deduce the behavior of a discretetime Nyquist pulse in the frequency domain: two examples are given in Figure 8. for n 6D 0. Using the 1 polar notation for Q. i. because of an incorrect choice of the timing phase t0 . where q. i.e / Q f T `D 1 F0 ` T Ã e Â Ã F j2³ f ` T0 t0 (8. and h. the pulse f i g at the equalizer output before the downsampler has the following Fourier transform: 9. With reference to the scheme of Figure 8. Discretetime Nyquist pulses and relative Fourier transforms.e.t/ is real with a bandwidth smaller than 1=T . it may happen that H . for F0 D 1.2T /.0/ 6D 0. for F0 D 2 and F0 D 1. 1 2T 0 1 2T 1 T f Figure 8. From (8. We see that choosing F0 D 1. impose that h.8.
For this choice. from (8. in a practical implementation the equalizer output is not generated at every sampling instant multiple of T =2.8.68) The adaptive FSE may incur a difﬁculty in the presence of noise with variance that is small with respect to the level of the desired signal: in this case some eigenvalues of the autocorrelation matrix of xŁ may assume a value that is almost zero and consequently 2k . this problem is avoided because aliasing between replicas of Q. f / is Hermitian. the FSE receiver presents two advantages over Tspaced equalizers: 1. if c has an input signal sampled with sampling period T =2 it also acts as an interpolator ﬁlter.5 (see Observation 8. It is an optimum structure according to the MSE criterion. It is less sensible to the choice of t0 . Fractionally spaced equalizer (FSE) 633 as Q. Therefore the choice of t0 may be less accurate. the input samples of the ﬁlter c have sampling period T =2.65) 2i C 1 t0 D ³ T 2 i integer (8. we consider now the adaptive method as depicted in Figure 8.e j2³ 2T T / D 0 1 1 T Ã e 1 j2³ 2T Á 1 T t0 (8.4. In fact.66) (8.2T /.7 on page 644).67) In this situation the equalizer will enhance the noise around f D 1=.9. The LMS adaptation equation is given by: ckC1 D ck C ¼ek xŁ 2k (8. the correlation method (7. In conclusion. whose output can be used to determine the optimum timing phase. but only at alternate sampling instants.7. In fact.63). and the output samples have sampling period T . Note that coefﬁcient update takes place every T seconds. With respect to the basic scheme of Figure 8. f / does not occur. as we will see in Chapter 14. Therefore. Â Ã Â þ t0 1 1 þ e j2³ 2T C Q H .e j2³ f T /þ DQ 1 f D 2T 2T 2T Ã Â t0 D 2A cos ' C ³ T If t0 is such that 'C³ then H . If F0 ½ 2 is chosen. The choice of the oversampling index F0 D 2 is very common. Adaptive FSE The direct method to compute the coefﬁcients of a FSE is described in Section 8. or converge with difﬁculty in the adaptive case.269) with 0 accuracy TQ D T =2 is usually sufﬁcient to determine the timing phase. 2. in the sense that it carries out the task of both matched ﬁlter (better rejection of the noise) and equalizer (reduction of ISI).
1.1 with 0 < Þ − rx .9.k ACC ek µ + yk a kD Figure 8.634 Chapter 8. As a result.k ACC * c 2.69) Þck / ¼Þ/ck C ¼ek xŁ 2k (8. 2. the coefﬁcients of the ﬁlter c may vary in time and also assume very large values. the problem of ﬁnding the optimum coefﬁcients become illconditioned. To mitigate this problem.ek xŁ 2k D . Adaptive FSE (F0 D 2).71) . we consider two methods that slightly modify the cost function. The leaky LMS algorithm. with numerous solutions that present the same minimum value of the cost function.70) # jci j (8.k ACC * c N1. In both cases we attempt to impose a constraint on the amplitude that the coefﬁcients may assume. in the limit case of absence of noise. Let " J1 D J C Þ E then ckC1 D ck C ¼.0/. Let " J2 D J C Þ E N 1 X i D0 N 1 X i D0 # jci j 2 (8. This effect can be illustrated also in the frequency domain: outside the passband of the input signal the ﬁlter c may assume arbitrary values.k ACC * c 1. Channel equalization and symbol detection x2k T/2 x 2k1 T/2 x 2k2 T/2 x 2kN+1 * c 0. recalling the leaky LMS algorithm (see page 187).
Q We assume.74) becomes: x k D h 0 ak C .7).75) In addition to the actual symbol ak that we desire to detect from the observation of x k . then the DFE cancels the ISI due to postcursors. (8. For example.5 Decision feedback equalizer (DFE) We consider the sampled signal at the output of the analog receive ﬁlter (see Figure 8. : : : . the desired signal is given by: sk D s R . : : : . n D 1. M2 . and those with negative time indices precursors. N2 .75) two terms are identiﬁed in parentheses: one that depends only on past symbols ak 1 .h N1 akCN1 C ÐÐÐ C h 1 akC1 / C . : : : . given by (8. We note O that this is done without changing the noise wk that is present in x k . : : : . for n D N2 C 1. ak N2 . : : : . for i D 1. N1 C1. akCN1 .74) where wk is the noise. and ak i D ak i .ek xŁ 2k D ck Þ sgn ck / (8. Explicitly writing terms that include precursors and postcursors. M2 .10. If the past symbols and the impulse response fh n g were perfectly known.11. we obtain a scheme to cancel in part ISI.5. in (8.k D b1 ak 1 C Ð Ð Ð C b M2 a k O M2 (8. bn D h n . where. we refer to [1]. : : : . as illustrated in Figure 8.47). : : : .76) If M2 ½ N2 . and output given by O xFB. we could use an ISI cancellation scheme limited only to postcursors. in general. including convergence properties. N2 .8. and another that depends only on future symbols. akC1 . bn D 0.73) where the sampled pulse fh n g is deﬁned in (8.t0 C kT / D C1 X iD 1 ai h k i (8. 8. that fh n g has ﬁnite duration and support [ N1 . In the presence of noise we have x k D s k C wk Q (8. in the scheme of Figure 8.72) ¼Þ sgn ck C ¼ek xŁ 2k For an analysis of the performance of the FSE. Substituting the past symbols with O their detected versions fak 1 . Simulations results also demonstrate better performance of the FSE with respect to an equalizer that operates at the symbol rate [2]. as O illustrated in Figure 8.h 1 ak 1 C Ð Ð Ð C h N 2 ak Q N 2 / C wk (8. the feedback ﬁlter has impulse response fbn g.5. N2 ]. : : : . Q .5 or Figure 8. Decision feedback equalizer (DFE) 635 then ckC1 D ck C ¼. N2 1. ak N2 g. The samples with positive time indices are called postcursors.46). for n D 1.
Discretetime pulses in a DFE.77) .k D ci x k i (8. where two ﬁlters and the detection delay are outlined: 1. The general structure of a DFE is shown in Figure 8. (b) After the FF filter. Figure 8.12. M1 1 X i D0 z k D xFF. with M1 coefﬁcients. Channel equalization and symbol detection (a) Before the FF filter.10. Feedforward (FF) ﬁlter c.636 Chapter 8.
so the FF ﬁlter can effectively equalize. with M2 coefﬁcients. We note that the FF ﬁlter may be implemented as a FSE.k C xFB. whereas the feedback ﬁlter operates with sampling period equal to T . Figure 8.8. with respect to the desired sample D . Now (see Figure 8.4. however.k D M2 X i D1 bi ak i D (8.78) Moreover.k (8. 1. General structure of the DFE. yk D xFF. M1 T =F0 (timespan of the FF ﬁlter) at least equal to .10) ideally the task of the feedforward ﬁlter is to obtain an overall impulse response f n D h Ł cn g with very small precursors and a transfer function Z[ n ] that is minimum phase (see Example 1.5. xFB. .3). In this manner almost all the ISI is cancelled by the FB ﬁlter.12. Simpliﬁed scheme of a DFE. The following guidelines. Decision feedback equalizer (DFE) 637 Figure 8. The choice of the various parameters depends on fh n g.79) We recall that for a LE the goal is to obtain a pulse f n g free of ISI.11. Feedback (FB) ﬁlter b. are usually observed. where only the feedback ﬁlter is included.N1 C N2 C 1/T =F0 (timespan of h). 2.
In practice. for which we have N1 C N2 × M1 . For very dispersive channels.M1 1/T =F0 . ak D 1 .13. to reduce the constraints on the coefﬁcients of the FF ﬁlter the value of D is lowered and the system is iteratively designed.80) (8. 3. D .81) (8. : : : .82) Figure 8. to simplify the FB ﬁlter. equal to the detection delay. x k M1 C1 . M2 T (timespan of the FB ﬁlter) equal to or less than .638 Chapter 8.M1 1/T =F0 (timespan of the FF ﬁlter minus one). The detection delays discussed above are referred to a pulse fh n g “centered” in the origin. that does not introduce any delay. : : : . : : : .N 1/=2. which determines the number of postcursors. b M2 ] T M1 1 X i D0 M2 X jD1 ci x k i C b j ak D j (8. DT is approximately equal to N 2 1 F0 . T For a LE. Implementation of a DFE. b2 . ak D M2 ] T 1 . instead. DT is equal to or smaller than .M1 1/ =F0 . it results M2 T × M10T . Adaptive DFE We consider the scheme implemented in Figure 8. b1 . The choice of DT. : : : . c M1 and the input vector ξ k D [x k . If the precursors are not negligible. F 4. where the output signal is given by yk D Deﬁning the coefﬁcient vector ζ D [c0 . . is obtained by initially choosing a large delay. x k O O O 1 .13. M2 depends also on the delay D. c1 . Channel equalization and symbol detection 2. the criterion is that the center of gravity of the coefﬁcients of the ﬁlter c is approximately equal to . ak D 2 .
we recall the following results: 1.73). ak O The LMS adaptation is given by ζ kC1 D ζ k C ¼ek ξ Ł k (8.88) where rh .n/ D ¦a h Ł n (8. with the usual assumptions of symbols that are i. autocorrelation of x k 2 rx .n/ Q (8.n/ C rw .nT / Q (8.n/ D N0 rg M .n/ D Deﬁning p N2 X jD N1 h j hŁ j n rw . crosscorrelation between ak and x k 2 rax .d. during the transmission of a training sequence. For a generic sequence fh i g in (8.n/ D ¦a rh .83) yk D (8.89) D h Ł cp D M1 1 X `D0 c` h p ` (8. Decision feedback equalizer (DFE) 639 we express (8.80) becomes yk D N2 CM1 1 X pD N1 p ak p C M1 1 X i D0 ci wk Q i C M2 X jD1 b j ak D j (8.d.84) D ak D.8.90) equation (8. for the MSE criterion with J D E[jak D yk j2 ] (8. We recall that.91) .86) the Wiener ﬁlter theory may be applied to determine the optimum coefﬁcients of the DFE ﬁlter in the case ak D ak .87) 2.n/ Q are known.85) Design of a DFE with a ﬁnite number of coefﬁcients If the channel impulse response fh i g and the autocorrelation function of the noise rw .5. and O statistically independent of the noise.80) in vector form: yk D ζ T ξ k The error is given by O ek D a k D (8.
1.96) and. using (8. : : : . 1.98) 2 D ¦a 1 M1 1 X `D0 .92). : : : .` h i CD ` i D 1.93) To obtain the Wiener–Hopf solution. we get 2 Jmin D ¦a M1 1 X `D0 copt. : : : . p q/ M2 X jD1 h jCD Ł q h jCD p p.95) h j hŁ j .` [p]Ł ` ! copt.640 Chapter 8.80) yields yk D M1 1 X i D0 i CD i D 1. M1 ÄÂ [R] p. : : : . the optimum feedback ﬁlter coefﬁcients are given by bi D M1 1 X `D0 copt. from (8. p Q q/ 1 p ak D j2 (8.` h D ` (8. q D 0.92) in (8. Channel equalization and symbol detection Observing (8.91). M2 (8. the optimum choice of the feedback ﬁlter coefﬁcients is given by bi D Substitution of (8.94) 1 p D 0. M1 Therefore the optimum feedforward ﬁlter coefﬁcients are given by copt D R 1 p (8.q D E xk q M2 X j1 D1 Ã h j1 CD q ak D j1 Â xk 2 D ¦a N2 X jD N1 p M2 X j2 D1 ÃŁ ½ h j2 CD ! C rw . the following correlations are needed: " [p] p D E ak D xk p M2 X jD1 !# Ł h jCD p ak D j 2 D ¦a h Ł D p (8.97) Moreover. 2.94).92) ci xk i M2 X jD1 ! h jCD i ak j D (8. M2 (8.
2. Decision feedback equalizer (DFE) 641 Observation 8. e.9 on page 119) of the impulse response.1 In the particular case in which all the postcursors are cancelled by the feedback ﬁlter. for example. Efﬁcient methods to determine the inverse of the matrix are described in Section 2.3.117).16)).B.2 The equations to determine copt for a LE are identical to (8.n/ is easily determined. if the statistical power of wk is known.3 For white noise wC . however. Observation 8. while the expression of the elements of the matrix R in (8. The overall discretetime system impulse response obtained by sampling the output signal of the antialiasing ﬁlter gAA (see Figure 8.98). Observation 8. the component with largest energy. the autocorrelation of wk is proportional to the autocorrelation of the Q receive ﬁlter impulse response: consequently. with M2 D 0. in any case it is (semi)deﬁnite positive. which is determined by methods described in Chapter 14.4 For a LE.100) Observation 8. We then implement the MF g M by choosing.5. that is for M2 C D D N 2 C M1 (8. the Q autocorrelation rw .q D ¦a D X jD N1 1 (8. We recall that a ﬁne estimate of t0.A. a larger sampling period of the signal at the MF input is considered. the matrix R is Hermitian and Toeplitz. In particular. the vector p in (8.5 The deﬁnition of fh n g depends on the value of t0 . thus realizing the MF criterion (see also (8. This is equivalent to selecting among the four possible components with sampling period T =2 of the sampled output signal of the ﬁlter gAA the component with largest statistical power. p Q q/ (8. while for a DFE it is only Hermitian. The sampling period.g. .M F is needed if the sampling period of the signal at the MF output is equal to T .95) is simpliﬁed as 2 [R] p.94)–(8.M F at the MF output.8.94) is not modiﬁed. by estimation. T =2.99) hj Ł qh j p C rw . Finally. Observation 8. for example T =8.4) is assumed to be known.95) is modiﬁed by the terms including the detected symbols. To reduce implementation complexity. A particularly useful method to determine the impulse response fh n g in wireless systems (see Chapter 18) resorts to a short training sequence to achieve fast synchronization. This method is similar to the timing estimator (14. the coefﬁcients of the channel impulse Q Q response fh n g and the statistical power of wk used in R and p can be determined by the methods given in Appendix 3. among the four polyphase components (see Section 1. is in principle determined by the accuracy with which we desire to estimate the timing phase t0.
If fxn g is the input signal of the FF ﬁlter. It is usually chosen either as the time at which the ﬁrst useful sample of the overall impulse response occurs. or the time at which the peak of the impulse response occurs.A A for the signal at the input of g M is determined during the estimation of the channel impulse response. thus exploiting the knowledge of the training sequence. and an FB ﬁlter with M2 coefﬁcients. where B is the backward operator deﬁned on page 27. Also the FB ﬁlter will be anticausal. In the particular case fh n g is a BŁ correlation sequence.A). it is convenient to process the observed signal fx k g starting from the end of the block. Otherwise the function of the MF may be performed by a discretetime ﬁlter placed in front of the ﬁlter c. 1. if f n g is ideally minimum phase and causal with respect to the BŁ timing phase. Note that. let’s say from k D K 1 to 0. : : : .A A C t M F . 1. the extension to an FSE with an oversampling factor F0 > 2 is straightforward. besides reducing the complexity of c (see task 2 on page 628).12. where the FF ﬁlter c is now a fractionally spaced ﬁlter. the symbols are assumed i. facilitates the optimum choice of t0 (see Chapter 14). for k D 0. In fact. The overall receiver structure is shown in Figure 8.102) . i. shifted by a number of modulation intervals corresponding to a given number of precursors. Now if f n D h Ł cn g and fbn g are the optimum impulse responses if the signal is processed in the forward mode.642 Chapter 8. We consider the scheme of Figure 8. We recall that the MF.269). and statistically independent of the noise signal. Channel equalization and symbol detection The timing phase t0. are the optimum impulse responses in the backward mode for k D K 1.101) xn D kD 1 The signal fyk g at the DFE output is given by yk D M1 1 X i D0 ci x2k i C M2 X jD1 b j ak D j (8.6 In systems where the training sequence is placed at the end of a block of data (see the GSM frame in Appendix 17.7. : : : .M F D t0. if t M F denotes the duration of g M . it is easy to BŁ BŁ verify that f n g and fbn g. The criterion (7. then f n g can be obtained using as FF ﬁlter the ﬁlter having impulse BŁ response fcn g Design of a fractionally spaced DFE (FSDFE) We brieﬂy describe the equations to determine the coefﬁcients of a DFE comprising a fractionally spaced FF ﬁlter with M1 coefﬁcients and sampling period of the input signal equal to T =2.e. then the timing phase at the output of g M is given by t0. now f n g is maximum phase and anticausal with respect to the new instant of optimum sampling. K 1. 0.14. Observation 8. apart from a constant delay. according to which t0 is chosen in correspondence of the correlation peak. where the ﬁlter gAA may be more sophisticated than a simple antialiasing ﬁlter and partly perform the function of the MF. as illustrated in Figure 8. is a particular case of this procedure. we have C1 X h n 2k ak C wn Q (8.d.i. As usual.
106) h 2n q q hŁ 2n p C rw . : : : .m/ D N0 rgAA m Q 2 (8.108) K Ł 2 x2k i ] D ¦a h Ł 2K Ł x2k 2 p ] D ¦a C1 X nD 1 i (8.57).i CD/ D M1 1 X `D0 c` h 2. M1 h 2n q h Ł p 2n M2 X jD1 1 ! C rw . Â Ã T rw .104) With this choice (8. Let p D h Ł cp D M1 1 X `D0 c` h p ` (8.110) [R] p. : : : . p Q (8.10 on page 72) E[ak E[x 2k where. q D 0. jCD/ q h Ł jCD/ p 2. 1. jCD/ (8.8.105) Using the following relations (see also Example 1. Decision feedback equalizer (DFE) 643  Figure 8.105) are given by 2 [p] p D ¦a h Ł 2D 2 ¦a p p D 0. p Q q/ (8.14. : : : .103) then the optimum choice of the coefﬁcients fbi g is given by bi D 2. 1. 1 p. jCD/ i ak .102) becomes yk D M1 1 X i D0 ci x2k i M2 X jD1 ! h 2.109) q/ (8.q D C1 X nD 1 h 2.107) the components of the vector p and the matrix R of the Wiener problem associated with (8.9. M1 . FSDFE structure. from (8.i CD/ ` i D 1.5. M2 (8.
109)–(8. Channel equalization and symbol detection The feedforward ﬁlter is obtained by solving the system of equations R copt D p (8.111) with M2 D 0. Observation 8.B. For an FSE.269). Observation 8. it is possible to achieve the minimum value of Jmin . because it may be illconditioned. Salz derived the expression of Jmin for this case. this method avoids the need for the estimate of the overall channel impulse response.3). with appropriate changes. Note that the matrix R is Hermitian but in general it is no longer Toeplitz.` h 2D ` (8. it requires a greater computational complexity with respect to the method described in this section.104).1–8. for example by the correlation method (7.8 Two matrix formulations of the direct method to determine the coefﬁcients of a DFE and an FSDFE are given in Appendix 8.111). a solution consists in adding a positive constant to the elements on the diagonal of R.27). so that the performance of the optimum solution does not change signiﬁcantly.111) and the feedback ﬁlter is determined from (8. or FSLE. obviously the value of this constant must be rather small.7 Observations similar to the observations 8. so that R becomes invertible. a formulation uses the correlation of the equalizer input signal fxn g. In particular. . and the crosscorrelation of the two signals: using suitable estimates of the various correlations (see the correlation method and the covariance method considered in Section 2.` [p]Ł ` ! copt. The minimum value of the cost function is given by 2 Jmin D ¦a M1 1 X `D0 copt. Signaltonoise ratio γ Using FF and FB ﬁlters with an inﬁnite number of coefﬁcients. the correlation of the sequence fak g.3 hold for a FSDFE. the equations to determine copt are given by (8. Similarly to the procedure outlined on page 187. In this case the timing phase t0 after the ﬁlter gAA can be determined with accuracy T =2.e j2³ f T / 2T where 8 is deﬁned in (8. given by [3] 1 0 Z 1 2T N0 2 Jmin D ¦a exp @T dfA (8.113) ln 2 1 N0 C ¦a 8.112) 2 D ¦a 1 M1 1 X `D0 A problem encountered with this method is the inversion of the matrix R in (8.644 Chapter 8. however.
and the FB ﬁlter at the transmitter as a precoder. we can compare the performance of a linear equalizer given by (8. as in 2 Figure 1. An analogous relation holds for an FSDFE.15c. the performance is a function of the channel and of the choice of M1 . Remarks 1.e j2³ f T / 6D constant and the absence of detection errors in the DFE. R f . simulations indicate that for typical channels and symbol error probability smaller than 5 Ð 10 2 . for inﬁniteorder ﬁlters the value Jmin of a DFE is always smaller than Jmin of a LE.8.44).112).5. However. such that Q Q .a/ da (8.40) with that of a DFE given by (8. For channels with impulse response fh i g. If FF and FB ﬁlters with a ﬁnite number of coefﬁcients are employed. as in such cases the linear equalizer tends to enhance the noise. The DFE is deﬁnitely superior to the linear equalizer for channels that exhibit large variations of the attenuation in the passband. 4.98). assuming 8.113): the result is that. 2.113). instead of the DFE structure it is better to implement the linear FF equalizer at the receiver. such that detection errors may spread catastrophically. for a given ﬁnite number of coefﬁcients M1 C M2 . and observe the performance of adaptive LE and DFE by resorting to a speciﬁc channel realization.6.115) where Jmin is given by (8. with Jmin given by (8. or nonminimum phase. 3. using the precoding method discussed in Appendix 7. in analogy with (8.A and Chapter 13. error propagation is not catastrophic. 8. is either minimum phase. M2 ! 1) performance than the linear equalizer. h min . In the absence of errors of the data detector. Error propagation leads to an increase of the error probability.6 Convergence behavior of adaptive equalizers We consider the digital transmission model of Figure 8. as in Figure 1. The additive channel noise wk is AWGN with statistical power ¦w . fh n g.114) to (8. because they produce incorrect cancellations.a/ da Z Ä e e f . the DFE has better asymptotic (M1 . In particular. h nom .15a. Convergence behavior of adaptive equalizers 645 Applying the Jensen’s inequality. also for a DFE we have 2 DFE Jmin (8. However. we analyze two cases in which the discretetime overall impulse response of the system. Detection errors tend to spread.
d show curves of meansquare error convergence for standard LMS and RLS algorithms (see Section 3. Channel equalization and symbol detection 2 2 the signaltonoise ratio 0 D ¦a Ð rh . d and 8. we consider a LE with N D 15 coefﬁcients. respectively. b for the two channels. the best results are obtained for a delay D D 0 in the case of h min and D D 8 in the case of h nom .2) for minimum and nonminimum phase channels.15a. and a LE employing the LMS with ¼ D 0:062 or the RLS.n 0.1. Q The impulse response fcopt.5 1  c 0. In terms of meansquare error at convergence. The Q sequence of symbols ak 2 f 1.0/=¦w at the equalizer input is equal to 20 dB.646 Chapter 8. (a) 1. Adaptive LE With reference to the scheme of Figure 8. Jmin represents the minimum value of J achieved with optimum coefﬁcients computed by the direct method.5 0 0 5 10 0 5 10 15 n 0 (c) n LMS J(k) (dB) −5 −10 −15 −20 0 J min 10 20 30 40 50 60 (d) −15 J(k) (dB) −20 −25 −30 −35 0 Jmin RLS 10 20 30 40 50 60 k Figure 8. and the delay D is the sum of the delays introduced by fh n g and fcn g.16a.5 0 ψn 15 opt.16c.15c. Figures 8. estimated over 500 realizations. whereas the LMS algorithm still presents a considerable offset from the optimum conditions. b and 8. System impulse responses and curves of meansquare error convergence. in the plots.15. 1g is a PN training sequence of length L D 63.n g are depicted in Figures 8.n g of the optimum LE and the overall system impulse response f n D h Ł copt.6. even though a large adaptation gain ¼ is chosen. for a channel with minimum phase impulse response. We note that Jmin is 4 dB higher 2 than the value given by ¦w . we observe that the overall impulse response h nom is not centered at the origin.5 1 (b) 1. The curves of convergence of J . . because of the noise and residual ISI at the decision point.k/ indicate that the RLS algorithm succeeds in achieving convergence by the end of the training sequence.
estimated over 500 realizations. System impulse responses and curves of meansquare error convergence.5 0 0 0.5 0 ψn 10 opt.5 5 10 15 0 0 5 10 15 n 0 (c) n J(k) (dB) LMS −5 −10 −15 −20 0 Jmin 10 20 30 40 50 60 (d) −15 Jmin RLS J(k) (dB) −20 −25 −30 −35 0 10 20 30 40 50 60 k Figure 8.n  ψn 1 1 0.5 0 5 0 n 0 (c) 0 5 10 n LMS J(k) (dB) −5 −10 −15 −20 0 Jmin 10 20 30 40 50 60 (d) −15 J(k) (dB) Jmin −20 −25 −30 −35 0 RLS 10 20 30 40 50 60 k Figure 8.8. (a) 1 1 (b)  c 0. for a channel with nonminimum phase impulse response.5  copt.n 0. and a DFE employing the LMS with ¼ D 0:063 or the RLS.5 (b) 2 1. for a channel with minimum phase impulse response. Convergence behavior of adaptive equalizers 647 (a) 2 1. . and a LE employing the LMS with ¼ D 0:343 or the RLS.17. System impulse responses and curves of meansquare error convergence.6.16. estimated over 500 realizations.
M2 D 5. the signal at the output of a LE with N coefﬁcients (see (8. Channel equalization and symbol detection (a) 2 (b) 2 1.17c. Also in this case the chosen value of D gives the best results in terms of the value of J at convergence. Adaptive DFE We consider now the performance of a DFE as illustrated in Figure 8.17a. and a DFE employing the LMS with ¼ D 0:143 or the RLS. estimated over 500 realizations.n g of the optimum FF ﬁlter and the overall system impulse response f n D h Ł copt. respectively.n g are depicted in Figures 8.5 1 ψn 1 0. and D D 9. for both h min and h nom .13. d and 8.116) . The impulse response fcopt. b and 8.18c.18. System impulse responses and curves of meansquare error convergence. b. 8. for the two channels.5  copt.91)) is given by N 1 X i D0 CN N2X 1 pD N1 yk D ci x k i D p ak p (8.5 5 10 0 n 0 (c) 0 5 10 n LMS J(k) (dB) −5 −10 −15 −20 0 Jmin 10 20 30 (d) 40 50 60 −15 Jmin J(k) (dB) −20 −25 −30 −35 0 RLS 10 20 30 40 50 60 k Figure 8.7 LEZF with a ﬁnite number of coefﬁcients Ignoring the noise. for a channel with nonminimum phase impulse response.5 0 0 0.n  1. d show curves of meansquare error convergence for standard LMS and RLS algorithms. with parameters M1 D 10.648 Chapter 8.18a. Figures 8. for minimum and nonminimum phase channels.
8. we assume t0 D 0 and D D 0.8 DFE: alternative conﬁgurations We determine the expressions of the FF and FB ﬁlters of a DFE in the case of IIR ﬁlter structure.15. DFEZF We consider a receiver with a matched ﬁlter g M (see Figure 8. will be presented in Section 8. With reference to Figure 8.8.90). 0. An adaptive ZF equalization method is discussed in Appendix 8. where.1) followed by the DFE illustrated in Figure 8. 8. alternatively. and the result windowed in the time domain so that the ﬁlter coefﬁcients are given by the N consecutive coefﬁcients that maximize the energy of the ﬁlter impulse response.117) N1 . it can be inverted. is known. that can be solved by the method of the pseudoinverse (see (2. : : : . then the matrix of the system (8. An alternative robust method. If the overall impulse response fh n g. N2 . An approximate solution is obtained by forcing the condition (8.C.z/ to remove the ISI: therefore the LEZF output is given by x E. n D N1 . the matched ﬁlter output x k is input to a linear equalizer zero forcing (LEZF) with transfer function 1=8.25). The ztransform of the QAM system impulse response is given by 8.119) . if the determinant is different from zero.117) centered around D.19.z/. : : : .N 1/ (8. DFE: alternative conﬁgurations 649 where.185)).108)).117) is square and. as deﬁned in (8.118) only for N values of p in (8. N 2 C N 1 For a LEZF it must be p D Žp D (8. to simplify the notation. otherwise the equalizer coefﬁcients may deviate considerably from the desired values. a method to determine the coefﬁcients of the LEZF consists in considering the system (8.k D ak C w E.118) where D is a suitable delay. from (8.117) with Nt D N1 C N2 C N equations and N unknowns.k (8. Note that all these methods require an accurate estimate of the overall impulse response. : : : . which does not require the knowledge of fh n g and can be extended to FSEZF systems.19. the solution can be found in the frequency domain by taking the Nt point DFT of the various signals (see (1. N 1 X `D0 p D c` h p ` D c0 h p C c1 h p pD 1 C Ð Ð Ð C cN 1 h p .
The ﬁnal result is F. using the property (8. is obtained by considering a minimumphase prediction error ﬁlter A.121). where fr x . associated with an anticausal sequence f f Łn g.z/ is the ztransform of a correlation sequence. associated with a causal sequence ffn g.k is given by Pw E .z/ D C1 X nD0 fn z n (8.z/.122) is a minimumphase function. On the other hand.23). Channel equalization and symbol detection Figure 8.19.26). The equation to determine the coefﬁcients of A.z/ of w E. it can be factorized as (see page 53) Â Ã 1 Ł (8. we obtain that the spectrum Pw E .121) 8.z/ is given by (2.z/ D F.n/g is now substituted by frqC .N z N (see 1.z/ 1 8. designed using the ACS frqC .z/ As 8.nT /g deﬁned by (8. DFE zeroforcing.85).z/8 Ł Â 1 zŁ Ã (8.z/ in (8.N page 147).z/ D N 0 8.650 Chapter 8. that is with poles and zeros inside the unit circle.z/ F zŁ where F. with a computational complexity that is proportional to the square of the number of ﬁlter coefﬁcients.29). F Ł .z/ D f 0 =A.1=z Ł / is a function with zeros and poles outside the unit circle.9 A useful method to determine the ﬁlter F. Observation 8. From (8.z/ D 1 C a 0 z 1 C Ð Ð Ð C a0N .nT /g. .120) D N0 1 8.
z/ D 1 9. DFE: alternative conﬁgurations 651 We choose as transfer function of the ﬁlter w in Figure 8. sampler. the noise is white with statistical power. Therefore the ﬁlter w is called whitening ﬁlter (WF).z/ (8.z/ 1. because it is anticausal.z/ D F.20. As yk is not affected by ISI.z/ D 1 W .z/ D 8. hence there are no precursors. The relation between ak and the desired signal in z k is instead governed by 9. If ak D ak then. for O B.8.125) DFE ZF Summarizing. The overall receiver structure is illustrated in Figure 8. that is the energy of the impulse response is mostly concentrated at the beginning of the pulse.z/. the relation between x k and z k is given by 1 1 F.N0 =f2 /.20. the ﬁlter composed of the cascade of LEZF and w is also a WF.z/ f0 1 Â f0 FŁ 1 zŁ Ã (8. .123) The ISI term in z k is determined by W .8.20 is nonrealizable. is called whitened matched ﬁlter (WMF).126) With this ﬁlter the noise in z k is white.z/=f 0 and B. for which we obtain D 2jf0 j2 N0 (8. the overall discretetime system is causal and minimum phase. the WF of Figure 8. Note that the impulse response at the WF output has no precursors. and whitening ﬁlter. DFEZF as whitened matched ﬁlter followed by a canceller of ISI. In principle. In other words. where the block including the matched ﬁlter.z/ 1 f0 (8.124) the FB ﬁlter removes the ISI present in z k and leaves the white noise unchanged.19 the function W . 0 In any case. this structure is called DFEZF. In practice we can implement it in two ways: Figure 8.z/ D F.
is given by rqC .1 C a 2 / . 0.1 a 2 / 1 C a 2 2a cos.e j2³ f T / D E qC and presents a minimum for f D 1=.1=z Ł /.1 a 2 / az 1 /.652 Chapter 8. We observe that the choice F.129) We note that the frequency response of (8.10 With reference to the scheme of Figure 8.128) qC .8.1 az/ az (8.130) aDe þT <1 (8. For a data detector based on x E. for k D K 1.nT /] D E qC D E qC . in (8. Example 8.z/ D f 0 =A. sampled at instant nT.1 a 2 / az 1 C .20.3).20 is illustrated by an example.131) . : : : .119).2T /.z/ is discussed in Observation 8.nT / D E qC a jnj Then 8.127) where rw E .0/ is determined as the coefﬁcient of z 0 in N0 =8.t/ D E qC 2þe þt 1.z/ D Z[r qC .z/. the ratio is LE ZF D 2 rw E .k given by (8. Let p (8.k .z/ is placed after the WF to produce the signal x E.t/ be the overall system impulse response at the MF input.129) is 8. using a LEZF instead of a DFE structure means that a ﬁlter with transfer function f0 =F. f0 Observation 8.1 (WF for a channel with exponential impulse response) A method to determine the WF in the scheme of Figure 8. 1. and processing the output samples in the forward mode for k D 0. This expression is alternative to (8.0/ (8. b) by processing the output samples of the IIR WF in the backward mode.1 . : : : . leads to an FIR WF with transfer function 12 AŁ .128) E qC is the energy of qC .z/. Channel equalization and symbol detection a) by introducing an appropriate delay in the impulse response of an FIR WF. where A. The autocorrelation of qC . starting from the end of the block of samples. . K 2.2³ f T / (8.9 on page 650.
1 a 2 / 1 q E qC .z/ D f0 D1 D C1 X nD1 an z 1 az n (8.z/ D Q CC .z/ inside the unit circle.8. can be implemented by a simple FIR with two coefﬁcients. it is easy to identify the poles and zeros of 8.E qC .z/Q CC Ł z where Q CC . The FB ﬁlter is a ﬁrstorder IIR ﬁlter with transfer function 1 1 F.132) n In particular.134) 1 z.z/ D p E qC .2 (WF for a tworay channel) In this case we directly specify the autocorrelation sequence at the matched ﬁlter output: Â Ã 1 Ł (8. DFE: alternative conﬁgurations 653 With reference to the factorization (8.136) 8. a C z D E qC .1 E qC .137) . apart from a delay of one sample .q0 C q1 z 1 / (8.1 a2/ a2/ 1 az a z n 1 C1 X nD0 1 (8.8.8.1 a 2 // and 1=.133) 1 zŁ / In this case the WF. hence F.1 a 2 //.1 a 2 / az/ (8.1 a2/ (8.1 E qC .121).E qC .135) 1 1 az 1 1 az 1 Example 8.D D 1/.20 is expressed as 1 Â f0 F Ł ÃD 1 . whose values are equal to a=. the coefﬁcient of z 0 is given by f0 D The WF of Figure 8.z/ D D q q E qC .
q0 C q1 e j2³ f T / (8. which.145) p E qC q 0 (8.q0 Žn C q1 Žn 1 / The frequency response is given by QCC .bz/ n (8. p F.137) we get 8.bz/ i D 1 X nD0 .nT /g and Q CC . Equation (8.z/ D E qC .2T /.139).146) .137) represents the discretetime model of a wireless system with a tworay channel.141) We note that if q0 D q1 .144) 1 Ł Ł /.136) and (8.z/ D E qC .q0 C q1 z/ (8.q0 C q1 z hence.z/ is minimum phase.z/ (8.143) 1 zŁ bz/ We note that the WF has a pole for z D b 1 .140) qCC .139). lies outside the unit circle.138) (8. 1 1 bz D 0 X iD 1 . in order to have a stable ﬁlter.142) 1 / D Q CC .1 bz/ with an anticausal sequence.139) In this way E qC is the energy of fqCC . recalling assumption (8.nT / D E qC . The impulse response is given by p (8. it is convenient to associate the ztransform 1=. From (8. the frequency response has a zero for f D 1=. recalling (8.1 (8. In this case. Channel equalization and symbol detection with q0 and q1 such that jq0 j2 C jq1 j2 D 1 jq0 j > jq1 j (8.q0 C q1 z and f0 D The WF is given by 1 Â f0 F Ł where Â bD q1 q0 ÃŁ ÃD 1 E qC jq0 j2 .654 Chapter 8. f / D p E qC .
In fact.8. DFE as ISI and noise predictor A variation of the scheme of Figure 8.z//.21.z/ the scheme of Figure 8.N 1/ Cz N ] Consequently the WF.bz/ n E qC jq0 j2 nD0 1 zŁ (8. DFE: alternative conﬁgurations 655 On the other hand.z/ C . apart from a delay D D N .z/ D f0 q1 z q0 1 (8. the FB ﬁlter input is colored noise.z/ D Z[a k ].21 consists in using as FF ﬁlter. we can approximate the series by considering only the ﬁrst .z/ D X E . with minimum variance. .1 D X E .z/W .19 is redrawn as in Figure 8. By removing the O correlated noise from x E. We refer to the scheme of Figure 8.21.147) D 1 z N [b N C Ð Ð Ð C bz E qC jq0 j2 . obtaining 1 Â f0 F Ł Ã' N X 1 .8. where the ﬁlter c is given Figure 8.k .z//A.1 W . where the FB ﬁlter acts as a noise predictor. plus the desired symbol ak .A. From the identity Y .148) DFEZF as a noise predictor Let A. can be implemented by an FIR ﬁlter with N C 1 coefﬁcients.149) W .z// (8.z/ C . a minimumMSE linear equalizer. we obtain yk that is composed of white noise. Predictive DFE: the FF ﬁlter is a linear equalizer zero forcing.22.z/ X E .N 1/ terms. for ak D ak . as jbj < 1. The FB ﬁlter in this case is a simple FIR ﬁlter with one coefﬁcient 1 1 F.
The spectrum of the noise in z k is given by Â Ã 1 Pnoise . the ISI in z k is given by: 8.z/¦ a D N0 2 .153) Therefore the spectrum of the disturbance vk in z k .26) and the fact that the symbols are uncorrelated with Pa .z/C.151) Hence.z/ D As t0 D 0.z/C.z/ 2 N0 C ¦a 8.z/ Â 1 zŁ Ã (8.N0 /2 2 . Predictive DFE with the FF ﬁlter as a minimumMSE linear equalizer.z/ D 2 N0 ¦a 2 N0 C ¦a 8.z/ (8.z/ 1D N0 2 N0 C ¦a 8. Channel equalization and symbol detection Figure 8. by (8. The ztransform of the overall impulse response at the FF ﬁlter output is given by: 8.z/ N0 2 N0 C ¦a 8.z/C.z/ D ¦ a .154) We note that the FF ﬁlter could be an FSE and the result (8.z/ D N 0 8.152) D 2 ¦a . the spectral density of the ISI has the following expression: PI S I .22. is given by Pv .z/ (8.z// 2 (8. composed of ISI and noise.656 Chapter 8. .z/ N0 2 N0 C ¦a 8Ł 2 ¦a 8.38).150) (8.N0 C ¦a 8.z/C Ł Ł z 4 8.154) would not change.z/ D Pa .N0 C ¦a 8.z// 2 2 using (8.
z/ D An alternative form is B. although suboptimum with respect to the DFE.z/ (8. needs to remove the predictable components of z k . it results that the prediction error signal yk is a white noise process with 2 statistical power equal to ¦ y .9 Benchmark performance for two equalizers We compare limits on the performance of the two equalizers.z/ (8. LEZF and DFEZF.159) . In conclusion.158) Â BŁ 1 zŁ Ã½ D [1 Ä B. An adaptive version of the basic scheme of Figure 8.154). the noise sequence fw E.526): Pv .z/] 1 with Pv . For O a predictor of inﬁnite length. with input ak z k D O ak z k D vk (assuming ak D ak ).z/A zŁ 2 ¦y (8.z/ given by (8. Performance comparison From (8.83).157) is the forward prediction error ﬁlter deﬁned in (2. the FB ﬁlter. c and b.z/ D C1 X nD0 0 an z n C1 X nD1 bn z n (8.z/ D 1 where A.22 suggests that the two ﬁlters. are separately adapted through the error signals fe F. in terms of the signaltonoise ratio at the decision point.z/ we use the spectral factorization in (1.8. Benchmark performance for two equalizers 657 To minimize the power of the disturbance in yk .120).156) a0 D 1 0 (8. we set B. 8.k g. . respectively. is used in conjunction with trelliscoded modulation (see Chapter 12) [4].155) A. This conﬁguration.z/ D 1 F.k g can be modeled as the output of a ﬁlter having transfer function C F .z/ D 2 ¦y Â Ã 1 Ł A.9.k g and fe B. To determine B.
0/ D N0 where.163) the comparison between (8.n g.z/ is causal.1 N0 1 C a 2 E qC 1 a 2 E qC 1 a 2 N0 1 C a 2 (8.1 C a 2 /. c F. LE ZF for the two simple systems introduced in Examples 8.131) assumes a minimum value close to zero. consequently.127).z/ is equal to rw E . from (8.n j2 ½ jc F.0 j2 D (8. Channel equalization and symbol detection and input given by white noise with PSD N0 .166) Using the expression of we get MF (7. LEZF Channel with exponential impulse response. also C F .167) Therefore the loss due to the ISI. 1 a2 1 C a2 LE ZF D MF (8.1/ D Using the inequality 1 X nD0 1 X nD0 jc F.164) Equalizer performance for two channel models We now analyze the value of and 8. is the ﬁlter impulse response.125) and (8.161) 1 1 D F.z/ D 1 X nD0 c F. n ½ 0. In this case the frequency response in (8. obtained for a MF receiver in the absence of ISI.1 a 2 /=. From (8.n j2 (8.160) where fc F.130) the coefﬁcient of z 0 in N0 =8.0 D C. can be very large if a is close to 1.658 Chapter 8. Then we can express the statistical power of w E.k as: rw E .8.113).127) yields LE ZF Ä DFE ZF (8.0/ D and.2. .165) D2 (8.159).z/ is causal: C F .8.n z n (8.1/ f0 1 jf0 j2 (8. Because F.162) jc F. given by the factor . from (8.
q0 C q1 z/ By a partial fraction expansion. the performance of LE and DFE are similar to the performance of LEZF and DFEZF.8.1 2 a2/ (8. Substitution of (8. we ﬁnd only the pole for z D circle.jq0 j2 which may be substantial if jq1 j ' jq0 j. respectively. From (8.133) in (8.168) q 1 =q0 lies inside the unit N0 N0 Â ÃD q1 E qC .142) we have 1 1 D Ł Ł 8. hence rw E .q0 C q1 z 1 /.1 C a 2 /.125). we have derived the conﬁguration of Figure 8.144) in (8.10 Optimum methods for data detection Adopting an MSE criterion at the decision point.10. Anyway.jq0 j2 jq1 j2 / Ł Ł E qC q 0 q 0 q 1 q0 (8.1 for an LE. that is for low noise levels.z/ E qC . DFEZF Channel with exponential impulse response.jq0 j jq1 j2 / (8. for both LE and DFE. we get DFE Z F D D 2 E q .0/ D (8. Optimum methods for data detection 659 Tworay channel. for the two systems of Examples 8. In this case the advantage with respect to LEZF is given by the factor jq0 j2 =. Tworay channel.1 and 8.12 for a DFE.125) yields DFE Z F D 2 E q jq0 j2 D N0 C 2 MF jq0 j (8.172) jq1 j2 /. In both cases. We recall that in case E qC × N0 .1 N0 C MF .171) a / We note that DFE Z F is better than LE ZF by the factor .8. .2 the values of in terms of Jmin are given in [4]. 8.170) Also in this case we ﬁnd that the LE is unable to equalize channels with a spectral zero. the decision on a transmitted symbol ak D is based only on yk through a memoryless threshold detector.169) Then LE ZF D 2 MF .8. and that of Figure 8. Substituting the expression of f0 given by (8.
v. modeled as a sequence of r. Sequence of received samples. : : : . Á 1 g are the precursors.g.174) where Á0 is the sample of the overall system impulse response. : : : .173) Án ak n (8.660 Chapter 8. Þ L 1 CK 1] T 1] T ai 2 A O (8. coherent demodulation of CPM signals (see Appendix 18. the decision criterion that minimizes the probability that an error occurs in the detection of a symbol of the sequence fak g requires in general that the entire sequence of received samples is considered for symbol detection.A).s from a ﬁnite alphabet a D [a L 1 . We assume the coefﬁcients fÁn g are known. : : : . which are typically negligible with respect to Á0 . Sequence of detected symbols.v. however. a L 1 CK 1] T ai 2 A (8.s from a ﬁnite alphabet O a D [a L 1 . a L 1 C1 .. Sequence of transmitted symbols. hence the samples fwk g are uncorrelated and therefore statistically independent.90).175) 2. or information message. Sequence of detected symbol values α D [Þ L 1 . and obviously detection of sequences transmitted over channels with ISI. with equal statistical power in 2 the two I and Q components given by ¦ I2 D ¦w =2. Recalling the expression of the pulse f p g given by (8. e. a L 1 CK O O O 3.v. decoding of convolutional codes (see Chapter 11). obtained in correspondence of the optimum timing phase. they are estimated by the methods discussed in Appendix 3. : : : .B. We assume a sampled signal having the following structure: z k D u k C wk where: ž u k is the desired signal that carries the information. 1. a L 1 C1 .176) Þi 2 A (8. modeled as a sequence of r. K 1 (8. possible applications span.177) 4. z 1 . uk D L2 X nD L 1 k D 0. in practice.178) . In this section. ž wk is a circularly symmetric white Gaussian noise. z K 1] (8. Channel equalization and symbol detection Actually. : : : . Þ L 1 C1 . fÁ L 1 . : : : . We introduce the following vectors with K components (K may be very large): 1. modeled as a sequence of complex r. a general derivation of optimum detection methods is considered. we have Án D nCD .s z D [z 0 .
for which the probability to observe z D ρ is maximum.185) An efﬁcient realization of the MAP criterion will be developed in Section 8.18).1.ρ j α/P[a D α] α (8.10.184) (8.179) Let M be the cardinality of the alphabet A.and a D α/ if α : max pzja . ² K 1] T ²i 2 C ρ 2 CK (8.10. into M K nonoverlapping regions Rα α 2 AK (8. we divide the vector space of the received samples.ρ j α/ D pz . or observed sequence ρ D [²0 .ρ/ the decision criterion becomes: O a D arg max P[a D α j z D ρ] α O . the following criteria may be adopted to minimize (8.183) P[a D α j z D ρ] P[a D α] (8. the MAP criterion coincides with the maximum likelihood sequence detection (MLSD) criterion O a D arg max pzja . C K . If all data sequences are equally likely.181) D X P[z 2 Rα j a D α]P[a D α] pzja .182) α2A K D X Z Rα α2A K As in (6.8. then the sequence α is detected: O if ρ 2 Rα H) a D α The probability of a correct decision is computed as follows: P[C] D P[O D a] a X D P[O D α j a D α]P[a D α] a α2A K (8. Maximum a posteriori probability (MAP) criterion ρ 2 Rα Using the identity pzja . . ²1 . the sequence α is chosen. : : : .ρ j α/dρ P[a D α] (8. Sequence of received sample values.182).ρ j α/ α (8.180) such that.186) In other words. By analogy with the analysis of Section 6.2. Optimum methods for data detection 661 5. if ρ belongs to Rα .
both a and u D [u 0 . given the sequence of observed samples. However. we are interested in detecting the symbols fak g and not the components fu k g. Then the MLSD criterion is formulated as O a D arg min α K 1 X kD0 j²k u k j2 (8.K 1/. as in the case of i.191) We note that (8. for each possible data sequence α of length K .i. : : : .173) we get:3 pzja .187) formally requires that the vector a is extended to include the symbols a L 2 .189) where nonessential constant terms have been neglected. or metric.30). Recalling that the noise samples are statistically independent.190) is a particular case of the minimum distance criterion (6. . 3 We note that (8.10.192) The detected sequence is the sequence that yields the smallest value of 0.d.174). u K 1 ]T are ﬁxed. with zero mean and variance ¦w . symbols there are M K possible sequences. of both members we get ln pzja . : : : .190) where u k .662 Chapter 8. is a function of the transmitted symbols expressed by the general relation u k D fQ. ak L2 / (8.ρ j α/ D K 1 Y kD0 1 e 2 ³ ¦w 1 u j2 2 j² ¦w k k (8.M K /. A direct computation method requires that. which is a monotonic increasing function. the corresponding K output samples. As the vector a of transmitted symbols is hypothesized to assume the value α. this method has a complexity O.v.1 Maximum likelihood sequence detection We now discuss a computationally efﬁcient method to ﬁnd the solution indicated by (8. : : : .²k j α/ (8. : : : . it follows pzk ja . elements of the vector u. from (8. should be determined. and it suggests a detecting the vector u that is closest to the observed vector ρ. Channel equalization and symbol detection 8. should be computed as 0.186). and the relative distance.akCL 1 .K 1/ D K 1 X kD0 j²k u k j2 (8. deﬁned by (8.187) pzja . ak . a L 1 1 .ρ j α/ D K 1 Y kD0 2 As wk is a complexvalued Gaussian r.188) Taking the logarithm.ρ j α/ / K 1 X kD0 j²k u k j2 (8.
u.u. as discussed in Appendix 8. we deﬁne 2 dmin D min d 2 .197) With the assumption of i.a/.d. it is convenient to describe fu k g as the output sequence of a ﬁnite state machine (FSM).akCL 1 . is utilized instead. : : : .194) Then the lower limit (6.α/.8.11 We denote by s 0 k ak L 2 .β/jj2 (8. Then 1 the vector that is obtained by removing from s k sk D . and Nmin is the number of vectors u. We denote by S the set of the states. the exhaustive method for the computation of the expressions in (8. sk 1/ as (8.191). Observation 8.D. that is u D u.i.10. the state is sk D .α/.sk . the number of states is equal to Ns D M L 1 CL 2 .akCL 1 . s 0 k 1/ 1 the oldest symbol.195) where M K is the number of vectors u.β// D jju.191) and (8.192) and (8.198) 1 From (8. With reference to (8. As the noise is white.196) and the output is given in general by (8.α/ u. that will be discussed in the next section.u.199) . we compute the distances d 2 .β (8.a/ whose distance from another vector is dmin . In this case the input is akCL 1 .196) we may deﬁne u k as a function of s k and s k u k D f . σ Ns g (8. that is the set of possible values of sk : sk 2 S D fσ 1 .a/. the Viterbi algorithm. u.86) can be used. In the signal space spanned by u. In practice.194) is not used. Optimum methods for data detection 663 Lower limit to error probability using the MLSD criterion We interpret the vector u as a function of the sequence a. ak L 2 C1 / (8. ak .193) for each possible pair of distinct α and β in A K . akCL 1 1 .β// α. The Viterbi algorithm (VA) The Viterbi algorithm efﬁciently implements the ML criterion. symbols. (8. and we get Ã Â Nmin dmin Pe ½ Q MK 2¦ I (8. σ 2 .191). : : : . : : : .
: : : . With each state σ j .205) are represented in Figure 8. akCL 1 / (8.sk D σ j /: L. the variable that determines a transition is akCL 1 . According to (8.198). : : : .akC1 .203) (8. σ 2 D . s1 .10. Example 8. deﬁned as: 0.sk D σ j / D s0 .664 Chapter 8. sk D σ j / D . 1/. at instant k we associate two quantities: 1. is called a trellis diagram. deﬁned as the sequence of symbols that ends in that state and determines 0. Channel equalization and symbol detection Deﬁning the metric 0k D the following recursive equation holds: 0k D 0k or.204) D σi ! sk D σ j (8.207) . the survivor sequence. ak / and the set of states contains Ns D 22 D 4 elements: S D fσ 1 D .sk D σ j / D .:::.199). ak 2 f 1. 1/g The possible transitions sk 1 (8. and overall impulse response characterized by L 1 D L 2 D 1.1. using (8. u k . In this case we have sk D . associated with the sequence of output samples u 0 .1 Let us consider a transmission system with symbols taken from a binary alphabet. and a branch indicates a possible transition between two states at consecutive instants. likewise there are M transitions that arrive to each state sk .s1 .1.201) C j²k f . sk 2 1 /j (8. Note that the metric is a function of the sequence of states s0 .s0 . 1. 1/. We note that in this case there are exactly M transitions that leave each state sk 1 .sk . The following example illustrates how to describe the transitions between states of the ﬁnite state machine. u 1 .23. : : : . Figure 8.sk Dσ j min 0k (8. the path metric.200) C j²k u k j2 (8. σ 3 D . : : : . j D 1. : : : . where a dot indicates a possible value of the state at a certain instant. 1.23. sk .206) 2.202) Thus we have interpreted fu k g as the output sequence of a ﬁnite state machine.192). 1/. 0k D 0k 1 1 k X i D0 j²i u i j2 (8. s1 . extended for all instants k. σ 4 D .a L 1 . and we have expressed recursively the metric (8. Ns . 1g. or cost function. that is M D 2.
s K 1 D σ jopt / associated with s K 1 D σ jopt having minimum cost. Note that the notion of survivor sequence can be equivalently applied to a sequence of symbols or to a sequence of states.8.sk 1 D σ iopt / C j²k f .sk 1 D σ i / C j²k f . with initial state s 1 .σ j .ak1 ) σ 1 =(1.211) 1 otherwise Analogously. σ j / (8.10.1) σ 3 =(1.s 1 D σ i / D (8.ak ) (1. σ i /j2 (8.sk D σ j / D . which may be known or arbitrary. the procedure is repeated until k D K 1.1) a k+N A 1 Figure 8.L. Optimum methods for data detection 665 s k1 =(a k .sk 1 D σ iopt /. it is convenient to assign to the states s 1 the following costs: ( 0 for σ i D σ i0 0. if the ﬁnal state s K is equal to σ f 0 .1) (1.210) Starting from k D 0. σ i /j2 is called branch metric.sk D σ j / D 0. which is determined as follows: σ iopt D arg The term j²k sk 1 Dσ i 2S !sk Dσ j min 0.1) σ 2 =(1. In fact. the optimum sequence of states coincides with the survivor sequence associated with the state s K 1 D σ j having minimum cost among those that admit a transition into s K D σ f 0 .1) (1. Therefore.σ j . Portion of the trellis diagram showing the possible transitions from state sk 1 to state sk . the same sequence includes sk 1 D σ iopt .209) and the survivor sequence is augmented as follows: L. These two quantities are determined recursively. it is easy to verify that if. we obtain: 0.1) σ 4=(1. If the state s 1 is known and equal to σ i0 . as a function of the symbol akCL1 2 A.1) 1 1 1 1 1 1 1 1 s k=(ak+1. The optimum sequence of states is given by the survivor sequence L. .208) f . σ iopt /j2 (8.1) (1. a survivor sequence of states includes sk D σ j then.σ j . at instant k.23. at instant k 1.
f . 1g. 1/ we have 0. 1.s 0. The development of the survivor sequences on the trellis diagram from k D 0 to k D K 1 D 3 is represented in Figure 8.24b illustrates how the survivor sequences of Figure 8. 1. σ i /j2 . Trellis diagram and determination of the survivor sequences.24a.2 Let us consider a system with the following characteristics: ak 2 f 1.σ j .24a are determined. 1/. L 2 D 2. we note that some paths are abruptly interrupted and not extended at the following instant: for example the path ending at state σ 3 at instant k D 1. Starting with s 1 D . The survivor paths associated with each state are represented in bold. Channel equalization and symbol detection Example 8. The branch metric.10. .s 0. associated with each transition is given in this example and is written j²k above each branch.s 0. L 1 D 0. K D 4.24. Figure 8.212) Figure 8. and s 1 D .s 1 1 1 1 D σ 1/ D 0 D σ 2/ D 1 D σ 3/ D 1 D σ 4/ D 1 (8.666 Chapter 8.
Optimum methods for data detection 667 We apply (8. and σ 4 .s1 D σ 3 / D minf1 C 1.s3 D σ 2 / D 3 (8.24b. i D 1. 1g D 1 (8. 0.217) The same procedure is repeated for k D 2. for k D 1. in sequence.:::. that typically is between 3 and 10 times the length of the channel impulse response: this means that at every instant k we decide on ak K d CL 1 . Ns .208) for k D 0. . the length of the survivor sequences is limited to a value K d .216) (8.s 1 D σ 2 / C 3g (8. the ﬁnal metrics and the survivor sequences are shown in the second diagram of Figure 8. called trellis depth or path memory depth. a value that is then removed from the diagram. The number of additions and comparisons is proportional to the number of transitions. 3.10. The minimum among the values assumed by 0. if the parameter K is large.sk / may become very large. then the latter value is usually normalized by subtracting the same amount from all the metrics.s 1 D σ 1 / C 1.214) (8. Obviously.8. Computational complexity of the VA Memory. : : : . the complexity is linear in K . there is no interest in determining the survivor sequence for states with metric equal to 1. Next.s3 D σ i / D 0. 1g D 2 0. 1/.s1 D σ 1 / D minf1 C 1. 1/ and the survivor sequences is Computational complexity. 1g D 3 0. The memory to store the metrics 0. starting with s0 D σ 1 . because the corresponding path will not be extended. Considering now s0 D σ 2 . Then the survivor sequence associated with s0 D σ 1 is L. and the trellis diagram is completed.s3 / is i D1. In any case. we obtain 0.s1 D σ 4 / D minf1 C 0.sk /.s1 D σ 2 / D minf1 C 2.213) is obtained for s 1 D σ 1 . expressed as a sequence of symbols rather than states.L 1 CL 2 C1/ . The decision is based on the survivor sequence associated with the minimum among the values of 0.219) In practice. M Ns D M .213) D minf1.s0 D σ 1 / D . 1g D 2 0.215) (8. the ﬁrst iteration is completed. In practice.sk D σ i /. for example the smallest of the metrics 0.218) The associated optimum survivor sequence is a0 D 1 a1 D 1 a2 D 1 a3 D 1 (8. 1g D1 We observe that the result (8. k and 0. where we have 0.sk /.sk proportional to the number of states Ns . σ 3 .208). an