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Index

A with gradient algorithm expected value 825


950--951 for discrete-time signals
Accumulator 55 with normalized algorithm 829-830
Adaptive arrays 900-902 949-950 moments 825
Adaptive filters 880--959 Adaptive line enhancer power 826
applications of 880--902 895-896 time-averages 830--833
for antenna arrays Adaptive noise cancelling
900--902 896-897 B
for channel equalization Akaike information criterion
883-887 (AIC) 997 Backward predictor 578, 841
for echo cancellation Algorithms 4 Bandlimited signals 266
887-891 Chirp-z 544-549 Bandpass filter 327, 332-334
for interference suppres- FFT 511-537 Bandpass signal 266
sion 891-895 Goertzel 542-544 Bandwidth 265-267,660
for linear predictive Remez 686-691 Bartlett's method (see Power
coding 897-900 Aliasing, frequency-domain spectrum estimation)
for noise cancellation 20, 389-394 Bessel filter 726-727
896-897 time-domain 391 Bilinear transformaiion
for system identification Alternation theorem 684 712-717
882-883 Amplitude 14 Binary codes 405
for system modeling Analog signals (see Signals) Blackman-Tukey method
880-882 (see Power spectrum
Analog-to-digital (AID)
direct-form FIR 902-927 estimation)
converter 5,19,
properties of 925-927 Burg algorithm (see Power
401-410
with FAEST algorithm spectrum estimation)
Autocorrelation, of determinis-
946 Butterworth filters 717-720
tic signals 116-128
with fast RLS algorithm of random signals 323-326,
923-925, 945-947 826 c
with MSE algorithm Autocovariance 826 Canonic form 112
903-907 Autoregressive (AR) process Capon method (see Power
with RLS algorithm 836,987 spectrum estimation)
907-927 autocorrelation of 837 Cauchy integral theorem
with square-root (LDU) Autoregressive-moving 156
algorithms 921-923 average (ARMA) Causality, implications of
lattice-ladder filters 927-954 process 837, 987 65~59
properties of 951-954 autocorrelation of 837 Causal signals 85
with a priori LS algorithm Averages, autocorrelation Causal systems 66-67, 83-85,
940 826 196-198
with error-feedback autocovariance 826 Cepstral coefficients 835
algorithm 944 ensemble 825-828 Cepstrum 261-262
1077
1078 Index

Characteristic polynomial homogeneous solution Parseval's theorem 479


99 98-101 periodicity 465
Chebyshev filters 720-724 particular solution 101-103 symmetry 465-470
Chirp signal 547 total solution 103-108 table 470
Chirp-z transform algorithm Differentiator 691 time reversal 476
544-549 design of 691-693 relationship to Fourier series
Circular convolution 470-476 Digital resonator 335 461,463
Coding 20-35 Digital sinusoidal oscillator relationship to Fourier
Comb filter 341-344 347-349 transform 461
Conjugate-gradient algorithm Digital-to-analog (D!A) relationship to z-transform
906 converter 5, 20, 36, 462
Constant-coefficient difference 408-440 use in frequency analysis
equations 93-108 Dirichlet conditions, for 488-495
solution of 98-108 Fourier series 228 use in linear filtering
Continuous-time signals 17 for Fourier transform 237 480-488
exponentials 17 Discrete Fourier transform Discrete-time signals 9, 36-52
sampling of 17-22, 384-394 (DFf) 454-461 antisymmetric (odd) 48
sampling theorem for computation 480-488, correlation 116-128
26-31, 384-394 511-536 definition 9, 42
Convolution (linear) 69-80 butterfly 522, 526, 528 exponential 44-45
circular 470-476 decimination -in -freq uency frequency analysis of
properties 80-83 FFf algorithm 524-526 241-259
sum 69 decimination-in-time FFf non periodic 48
Correlation 116-128,827 algorithm 519-526 periodic 11-17
autocorrelation 120, direct 480-488,511-513 random 11
321-326, 826 divide-and-conquer representation of 36
computation 123-125 method 513-536 sinusoidal 14-16
cross-correlation 118, in-place computations symmetric (even) 48
321-323 523 unit ramp 44
of periodic signals 123 radix-2 FFf algorithms unit sample 43
properties 120-123 519-526 unit step 43
Coupled-form oscillator nrdix-4 FFf algorithms Discrete-time systems 53-69
347-349 526-532 causal 66-67, 83-85
Cross-power density spectrum shuffling of data 523 dynamic 60
829 split radix 532-536 finite-duration impulse
via linear filtering response 88-89
537-549 finite memory 60, 88-89
D
definition 456 implementation of 563-601
Dead band 625 IDFf 456 infinite-duration impulse
Decimation 754-760 implementation of FFT response 88-89
Deconvolution 262, 349-354, algorithm 536-539 infinite memory 60, 88-89
358-360 properties 464-480 linear 63
homomorphic 262, 360-362 circular convolution memory less 57
Delta modulation 434 470-476 noncausal 66-67
Difference equations 89-108 circular correlation 478 nonlinear 64
constant coefficient 98-108 circular frequency shift nonrecursive 92-93
solution 98-108 478 recursive 90-93
for recursive systems 93, circular time shift 477 relaxed 56
108 complex conjugate 478 representation 36
from one-sided z -transform linearity 465 shift-invariant 60-61
210-211 multiplication 470-476 stability triangle 202
Index 1079

Discrete-time systems (contin- F prediction error filter 575,


ued) 839
Fast Fourier transform (FFT)
stable (BIBO) 67, 85-88 smoothing 36
algorithms 511-537
static 60 structures· 563-582
application to 511
time-invariant 60-61 Wiener filter 862-873
application to, correlation
unit sample (impulse) 537-539 Filter banks 790-796
response 73-80 critically sampled 794
application to, efficient
unstable 67,85-88 computation of DFT quadrature mirror 798-799
Distortion, amplitude 511-537 uniform DFT 790-796
312 delay 328 application to, linear Filtering 480
harmonic 366 filtering 537-539 of long data sequences
phase 312 485-488
implementation 536-537
Down sampling 52 overlap-add method for
mirror FFT 536
(see also Sampling rate 487-488
phase FFT 536
conversion) 52 overlap-save method for
radix-2 algorithm 519-526
Dynamic range 33, 404, 605 485-487
decimation-in-frequency via DFT 481-488
524-526
Filter transformations
E decimation-in-time 334-338, 727-734
519-526
Echo, far-end analog domain 730-732
radix-4 algorithm 526-532 digital domain 732-734
888 nea~end 888 split-radix 532-536
Echo canceller 888 lowpass-to-highpass 334-338
Fast Kalman algorithms 945 Final prediction error (FPE)
Echo suppressor 888 Fibonacci sequence 210
Eigenfunction 302 criterion 996-997
difference equation 210-211 Final value theorem 209
Eigenvalue 302 Filter 326
Elliptic filters 724-726 FIR filters 660
bandpass 327, 332-334 anti symmetric 660-670
Energy definition 312, 326-329 design 660-701
definition 45 design of IIR filters comparison of methods
density spectrum 238-241, 326-349, 701-727 699-701
254-259 all pass 345-347 differentiators 691-693
partial 382 by pole-zero placement equiripple (Chebyshev)
signal 45-46 329-349 approximation 678-699
Energy density spectrum comb 341-344 frequency sampling
238-241, 254-259 notch 338-341 method 671-678
computation 961-966 resonators (digital) Hilbert transformers
Ensemble 825 335-338 693-699
averages 825-828 design of linear-phase FIR window method 678-691
Envelope delay 328 660-701 linear phase property
Ergodic 830 transition coefficient for 664-670
correlation-ergodic 832-833 1047-1052 symmetric 660-670
mean-ergodic 831-832 distortion 312 FIR filter structures 565-582
Estimate (properties) distortionless 328 cascade form 567-568
asymptotic bias 969 frequency-selective 326 direct fQrm 566-567
asymptotic variance 968 highpass 327,329-332 conversion to lattice form
bias 969 ideal 327-329 581-582
consistent 968 lowpass 327, 329-332 frequency sampling form
(see also Power spectrum nonideal, passband ripple 569-574
estimation) 659 stopband ripple lattice form 574-605, 859
Estimate (properties), variance 660 conversion to direct form
968 transition band 660 579-581
1080 Index

FIR filter structures (contin- highest 16


ued) negative 13
normalized 22 IIR filters 701-727
transposed form 586-588
positive 13 design from analog filters
FIR systems 88, 92, 113 701-727
Fixed-point representation relative 22
Frequency analysis 234-251 by approximation of
601-605 derivatives 703-707
Floating-point representation continuous-time aperiodic
signals 234-238 by bilinear transformation
605-608 712-717,727
Flowgraphs 584-586 continuous-time periodic
signals 226-233 by impulse invariance
Folding frequency 25,389-391 707-712
Forced response 95 discrete-time aperiodic
signals 248-251 by matched-z transforma-
Forward predictor 578, tion 717
838-841 discrete-time periodic signals
241-245 least-squares design
Fourier series 18, 226-233, methods 746-747
241-245 dualities 267-268
for LTI systems 300-326 Pade approximaf ~n
coefficients of 226-229,
table of formulas for 269 747-748
241-245
Frequency response 306 pole-zero placement
for continuous-time periodic
computation 317-321 329-349
signals 226-233
geometric interpretation of Prony's (least squares)
for discrete-time periodic
317-321 746-747
signals 241-245
magnitude of 304 Shanks' (least squares)
Fourier transform 234-238,
phase of 304 748-749
248-251
relation to system function IIR filter structures 582-601
convergence of 251-254
inverse 236 314-317 cascade form 588-591
of continuous-time aperiodic to exponentials 301-309 direct form 582-584
signals 234-238 to sinusoids 306-309 lattice-ladder 594-601,
of discrete-time aperiodic Frequency transformations (see 860-862
signals 248-251 Filter transformations) parallel form 591-594
properties 283-291 Full duplex transmission 887 second-order modules 589
convolution 283-284 Fundamental period 15 transposed forms 584-588
correlation 284 Impulse response 106-108
G Initial value theorems 168
differentiation 289-291
frequency shifting 286 Gaussian random variable Innovations process 833-836
linearity 279-281 1045-1046 Interpolation 28-31, 389, 751,
modulation 286-287 subroutine for 1045 760-762
multiplication 288-289 Gibbs phenomenon 254, 669 function 28
of signals with poles on Goertzel algorithm 542-544 ideal 28-31, 389
unit circle 262-265 Granular noise 407 linear 36,427-440
Parseval's theorem 287 Group (envelope) delay 328 Inverse filter 349-350
relationship to Inverse Fourier transform
z-transform 259-261 H 236, 251
symmetry 271-279 Harmonic distortion 366, Inverse system 350-351
table 290 446-447 Inverse z-transform 156-170,
time-reversal 282 High-frequency signal 265 179-193
time-shifting 281 Hilbert transform 658 by contour integration
Frequency 11-16 Hilbert transformer 693-699 156-157, 180-182
alias 15,23 Homomorphic 360, 362 integral formula 156
content 26 deconvolution 360-362 partial fraction expansion
folding 25, 389-391 system 361 184-193
fundamental range 17 Hybrid 888-891 power series 182-184
Index 1081

J Linear prediction filter (see Natural response 95


Joint-process estimate 938--940 Linear prediction) Natural signals 267-268
575 Noise subspace 1017
K Linear predictive coding 897 Noise whitening filter 835
of speech 897-900 Normal equations 846
Kalman gain vector 918
LMS algorithm 905-907 solution of 846--854
L excess mean-square error of Levinson-Durbin algo-
908 rithm 846-850
Lattice filter algorithms properties of 907 Schur algorithm 850-853
927-954 Local loop 888 Number representation
a posteriori form 940 Low-frequency signal 265 601-608
a priori form 940 Lowpass filter 327 fixed-point 601--605
error-feedback form 944 LTI systems 112-116 floating point 605-608
gradient form 950-951 moving average 113 Nyquist rate 28
joint process estimate second order 112-113
938--940
modified form 940-946
structures 108--116
canonic form 112
o
normalized form 949-950 direct form I 108--109 One's complement. 603
properties of 951-954 direct form II 109-113 One-sided z-transform
square-foot form 949-950 nonrecursive 112-116 205-211
Lattice filters 574--579 recursive 112-116 Orthogonality principle
594--601, 858--862, 874 weighted moving average 866-867
ARMA structure 860-862 112 Oscillators (sinusoidal genera-
AR structures 858--860 tors) 347
MA structure 838--841 M CORDIC algorithm for
LDU decomposition 921-923 349
Leakage 489, 964 Maximal ripple filters 685
coupled-form 347-349
Learning curves 911 Maximum entropy method
digital 347
Least squares 746-747 993
Overflow 629--631
filter design 746-747 Maximum-phase system
354--357 Overlap-add method 487-488
Least-squares estimation Overlap-save method 485-487.
907 Mean square estimation 866
903-905 ' Overload noise 407
Levinson-Durbin algorithm Oversampling NO 433-440
846-850 orthogonality principle
866-867 Oversampling D/A 439
generalized 850, 876
Minimum description length
split Levinson 874 p
(MDL) 997
Limit cycle oscillations
Minimum-phase system Paley-Wiener theorem 656
624--629
354-357 Parseval's relations 238 255
Linear filtering 480-488
Minimum variance estimate 287, 479 "
based on DFf 480-488
1012-1015 aperiodic (energy) signals
overlap-add method
Mixed-phase system 354--357 238, 255, 287
487-488
Moving-average filter 304 DFf 479
overlap-save method
Moving-average (MA) process periodic (power) signals
485-487
837,987 230,246
Linear interpolation 427-440
autocorrelation of 837 Partial energy 358
Linear prediction 578,
Moving-average signal 115 Partial fraction expansion (see
838--858
Multichannel signal 8 Inverse z-transform)
backward 841-845
Multidimensional signal 6-9 Periodogram 966-971
forward 838-841
lattice filter for 845 estimation of 966-971
N
normal equations for 846 mean value 968
properties of 855-858 Narrowband signal 266 variance 968
/'
1082 Index

Phase 12 computational require- of WeIch estimate 982-983


maximum 354-357 ments 984-985 Quantization 20, 31-35,
minimum 354-357 performance characteris- 403-406
mixed 354-357 tics 981-984 differential 433
reSponse 306 WeIch 975-977,982-983 differential predictive 434
Pisarenko method 1015-1019 parametric (model-based) dynamic range 33, 404, 605
Poles 170 methods 985-1009 error 31, 34, 613-639
complex conjugate 189-192, ARMA model 988, in AID conversion 403-406
204-205 999-1001 in filter coefficients 613-620
AR model 989 level 32, 403
distinct 173-198
AR model order selection resolution 32, 605
location 173-198
996-997 rounding 32, 608-612
multiple-order 187-188
Burg method 990-994 step size 32, 605
Polyphase filters 766-767 least-squares 994-995 truncation 32, 608-612
for decimation 768 MA model 990, 997-999 Quantization effects 31, 405,
for interpolation 773 maximum entropy method 549, 601
Power 46 993 fixed-point numbers
definition 46 model parameters 601-605
signal 47 988-990 one's complement 603
Power density spectrum modified Burg 991 sign-magnitUde 603
229-233 relation to linear predic- table of bipolar codes
definition 230 tion 988-990 406
estimation of (see also Power sequential least squares two's complement
spectrum estimation) 995-996 603-605
230 Yule-Walker 990 floating-point numbers
periodic signals 229-233, USe of D FT 971-973 605-608
245-248 Prediction coefficients 839 in AID conversion 34-35,
random signals 828-830 Prediction-error filter 574, 839 406-408
rectangular pulse train properties of 855-858 in computation of DFT
232-233 Principal eigenvalues 1020 549-555
POwer spectrum estimation Probability density function direct computation
963 824-825 549-551
Capon (minimum variance) Probability distribution FFf algorithms 551-555
method 1012-1015 function 1041-1044 in filter coefficients 613-620
direct method 963 Prony's method 746-747 limit cycles 624-629
eigenanalysis algorithms Pseudorandom sequences dead band 625
1019-1028 145 overflow 629-631
ESPRIT 1'022-1025 Barker sequence 145 zero-input 625
MUSIC 1021 maximal-length shift register scaling to prevent overflow
order selection 1025-1026 sequences 144-145 629-631
Pisarenko 1015-1019 statistical characterization
experimental results Q 631-639
1001-1009 Quantizer 403
Quadrature mirror filters
from finite data 966-973 midrise 404
788
indirect method 963 midtread 404
for perfect reconstruction
leakage 964 resolution 403-405
798-799
nonparametric methods uniform 404
for subband coding 788
973-985 Quality 981-984
Bartlett 974-975 of Bartlett estimate 982
R
Blackman-Thkey of Blackman-Tukey estimate Random number generators
977-981, 983-984 983-984 1041-1046
Index 1083

Random processes 323-326, Reflection coefficients 575, of bandpass signals 779-781


824-833 598,845-846,927 polyphase filters for
averages 825-831 Resonator (see Digital 766-767
autocorrelation 826 resonator) Sampling theorem 26-28,
autocovariance 826 Reverse (reciprocal) polyno- 384-394
expected value 825 mial 579, 842 Schur algorithm 850-853
for discrete-time signals backward system function pipelined architecture for
829-830 579,842 853-854
moments 825 Round-off error 608-612, split-Schur algorithm 874
power 826 631-639 Shanks'method 748-749
correlation-ergodic 832-833 Sigma-delta modulation 436
discrete-time 829-830 s Signal flowgraphs 584-588
ergodic 830 Sample-and-hold 402-403, 409 Signals 2-4
jointly stationary 825 Sample function 825 analog 9
mean-ergodic 831-832 Sampling 9, 19, 21, 384-394 antisymmetric 48
power density spectrum aliasing effects 25-26 aperiodic 48
828-829 frequency 21 bandpass 266
response of linear systems frequency domain 449-454 continuous-time 9
323-326 interval 21 deterministic 11
autocorrelation 323-326 Nyquist rate 28 digital 10
expected value 323 of analog signals 21-31, discrete-time 9, 36-52
power density spectrum 384-394 electrocardiogram (EeG)
324-326 of discrete-time signals 8
sample function 825 751-806 harmonically related 17
stationary 825 of sinusoidal signals 22-26 multichannel 8
wide-sense 826 period 21
time-averages 830-832 multidimensional 9
periodic 21 natural 267
Random signals (see Random rate 21
processes) frequency ranges 267-268
theorem 26-28
Rational z-transforms 184-193 periodic 13
time-domain 22-26,384-394
poles 170-173 random 11, 824-833
uniform 21
zeros 170-173 correlation-ergodic
Sampling-rate conversion
Recursive least squares 832-833
762-806
907-954 ergodic 824
(see also Sampling-rate
direct-form FIR algorithms conversion) expected value of 825
907-927 applications of 784-806 mean-ergodic 832-833
fast LS 923-925,945-947 for DFf filter banks moments of 826-830
properties of 925-927 790-796 statistically independent
lattice algorithms 928-954 for interfacing 785 827
a posteriori form 940 for lowpass filters 786 strict-sense stationary
a priori form 940 for phase shifters 784-785 825
error-feedback form for subband coding time-averages 830-833
944 787-788 unbiased 831
gradient 950-951 for transmultiplexing uncorrelated 827
joint process estimate 796-798 wide-sense stationary
938-940 by arbitrary factor 781-782 826
modified form 940-946 by rational factor 762-766 seismic 268
normalized form 949-950 decimation 751-760 sinusoidal 12
properties of 951-954 filter design for 762-775 speech 2-4
square-root form 921-923 interpolation 751,760-762 symmetric 48
Recursive systems 112-116 multistage 775-779 Signal subspace 1020
1084 Index

Sign magnitude representation steady-state 195-196 y


603 transient 104, 195-196
Sinusoidal generators (see zero-input 95 Yule-Walker equations 846
Oscillators) zero-state 94 modified 1000
Spectrum 225-226 Yule-Walker method 990
analysis 226 T
estimation of 226,961-1028
(see also Power spectrum
Time averages 830-833 z
Time-limited signals 266
estimation) 226 Zero-input linear 96
Toeplitz matrix 847,864
Split-radix algorithms 532-536 Zero-input response 95
Transient response 104,
Spread-spectrum signal 892 195-196, 309-311 Zero-order hold 34, 409
Stability of LTI systems Transition band 660 Zero padding 456
196-203
1Tansposed structures 584-588 Zeros 170
of second-order systems Truncation error 32, 608-612
201-203 Zero-state linear 96
Trunk lines 888 Zoom frequency analysis
Stability triangle 202
Two's complement representa- 821-822
Steady-state response tion 603
195-196, 311-312 z-transforms 147
Structures 108-116 definition 147-148
direct form I 108-109
u bilateral (two-sided)
direct form II 109-113 Uniform distribution 407, 147-148
Subband coding 787-789 549-551, 608-612 unilateral (one-sided)
Superposition principle 63 Unit circle 261-265 205-211
Superposition summation Unit sample (impulse) inverse 156-170, 179-193
73 response 106-108 by contour integration
System 3, 53-56 Unit sample sequence 43 156-157, 180-182
dynamic 60 by partial
finite memory 60 v fraction-expansion
infinite memory 60 184-193
Variability 981
inverse 350 by power series 182-184
Variance 549-551,631-639
invertible 350 properties 157-170
relaxed 56 convolution 164-166
System function 177-179,
w correlation 166-167
314-317 Welch method 975-977, differentiation 163-164
of all-pole system 179 982-985 initial value theorem
of all-zero system 177-179 Wide band signal 266 168
of LTI systems 177-179 Wiener filters 862-873, 904 linearity 157-159
relation to frequency FIR structure 864-866 multiplication 167-168
response 314-317 for filtering 863 Parseval's relation 168
System identification 350, for prediction 863
scaling 161-162
358-360 for smoothing 863
System modeling 836 IIR structure 867-871 table of 169
System responses 94 noncausal 871-873 time reversal 162
forced 95 Wiener-Hopf equation 864 time shifting 159-161
impulse 106-108 Wiener-Khintchine theorem rational 170-179
natural (free) 95,212 285 region of convergence
of relaxed pole-zero systems Window functions 668 (ROC) 147-156
170-179 Wold representation 836 relationship of Fourier
of systems with initial Wolfer sunspot numbers 10 transform 259-261
conditions 211-214 autocorrelation 125 table of 169

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