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194 Digital Signal Processing

The analysis of any sampled signal or sampled data system in the frequency
domain is extremely difficult using s-plane representation because the signal or
system equations will contain infinite long polynomials due to the characteristic
infinite number of poles and zeros. Fortunately, this problem may be overcome by
using the 2-transform, which reduces the poles and the zeros to a finite number in
the 2-plane.
The purpose of the 2-transform is to map (transform) any point s = ± o ± j co in
the s-plane to a corresponding point z (r |_0) in the 2-plane by the relationship
z = esTt where T is the sampling period (seconds)

Table 4.1
o = 0, cos = —
__ T
jto 0 OJ/8 to/4 3<O/8 (D/2 5(0/8 3(0/4 7o)/8 (0,
Z=lLo)T lL0° 11_45° lL90° lLl35° lLl80° ll_225° 11.270° l|_315a lL360°

Under this mapping, the imaginary axis, a = 0 maps on to the unit circle | z | = 1
in the 2-plane. Also, the left hand half-plane a < 0 corresponds to the interior of the
unit circle \z\ = 1 in the 2-plane. This correspondence is shown in Fig. 4.1.

Us) Uz)

Considering that the real part of x is zero, i.e. cr= 0, we have z = eJloT = 1 L
±7'COT, which gives the values of z (in polar form) shown as in Table 4.1.

We know that the Laplace transform gives L[x\t)]

= X(s) = £ x(nDe-nsT
n- 0
198 Digital Signal Processing

(vi) If x ( n ) is two-sided, and if the circle \ z \ = r0 is in the ROC, then the ROC
will consist of a ring in the z-plane that includes the circle | z | = r0. That is,
the ROC includes the intersection of the ROC’s of the components.
(vii) U X ( z ) is rational, then the ROC extends to infinity, i.e. the ROC is bounded
by poles.
(viii) If x ( n ) is causal, then the ROC includes z = °°.
(ix) If x ( n ) is anti-causal, then the ROC includes 2 = 0.
To determine the ROC for the series expressed by the Eq. 4.2, which is called a
two-sided signal z-transform, this equation can be written as
OO -1 o°

£x ( n ) r ~ n = £*(n)2”"+ ^ x ( n ) z ~ n
n = - oo n = - oo /i = 0

= £x(-/i)z+B + £*(ra)z“n /1 = 1 /1 = 0

The first series, a non-causal sequence, converges for | z \ < r2, and the second
series, a causal sequence, converges for \ z \ > rl9 resulting in an annular region of
convergence. Then the Eq. 4.2 converges for r1 < 12 | < r2. provided ri < r2. The
causal, anti-causal and two-sided signals with their corresponding ROCs are shown
in Table 4.2. Some important commonly used 2-transform pairs are given in Table
4.3.

Table 4.2 The Causal, anti-causal and two-sided signals and their ROCs
Signals ROCs
(a) Finite duration signals
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Therefore, 1\2 ) =
X{z
1 HereA the last
-
term( —
1
(l + + -A
z-
\ .-1- )1 1\2
11z
) '
\~ A 2 = 0, (il .- 2ef. )A+j z=
-
(1-z'
(l-z )
1 -1 1
can
)
-1 3/4(l-z-
be) expanded
(l-z-‘)
2 (l-z-
(l-z- )
1-i - 1 \ 2
1 1-l2\ 2as)
2
Taking inverse 2
-transform,
A 2 - 2 2Aj + A3 =( l0-wez - 1 get)2 z-Transforms 219

S ^(«)=r4(-i)"+4
o l v i n4.23
Example
1/4
g we get, A, =
+
^”i“(")
A = \ and -j, 2
A3 = ^ L4 4 2J
4 4 2

Therefore, X(z) = —+ 1/2

Determine the inverse 2-transform of the following X (z ) by the partial fraction


expansion method

F(2) = X (z ) = *+-- ------


2z2 - 7z + 3
if the ROCs are (a) \z \ > 3, (b) \z \ < 1/2 and (c) 1/2 < 12 |
<3. So lu tion We desire the partial fraction expansion of X (z )lz y which is
X (z ) z + 2 2+2

z(2z2 - 7z + 3) 2 Z ( Z - ± \ Z - 3 )

AJ + A2
2
-
z+2 2
where, A0 = zF (z)|* = 0 = =

2 z - - 3 ) 3
2=0

A, = [ z - - ± | F ( z ) z = 2i 2 2+( 2 - 3 ) 22 = -l
_ 1 2
* "

AO=(2-3)F(2)|2 = 2 +2 1
= 3
3
22
RL

Bahan dengan hak cipta


z-Transforms 221

A 2= F ( Z ) | Z - - ' ) | ,= --- —I
1
2 SA-J 3(z -1) 2

1 1

X (z ) 2 +— 2

u
(-D
f* I*
XU) = -2 ----
u-

(a) When the ROC is \z \ > 1, the signal x (n ) is causal and both terms are causal.

Therefore, x(n) = ^(l)n u (n ) - ^ u(n)


Z Z V
o/
1
2 w(n)

(b) When the ROC is \z \ < —, the signal x (n ) is anti-causal, i.e. the
3
inverse gives negative time sequences. Therefore,

x(n ) =
Mr u (-n - 1)

(c) Here the ROC — < \z \ < 1 is a ring, which implies that the signal
3 ‘
x (n ) is two-sided. Therefore one of the terms corresponds to a
causal signal and the other to an anti-causal signal, obviously the

given ROC is the overlapping of the regions \z \ > - and \z | < 1.


3
The partial fraction expansion remains the same; however because of ROC, the
pole at 1/3 provides the causal part corresponding to positive time and the pole
at 1 is the anti-causal part corresponding to negative time. Therefore, x (n )
becomes

Bahan dengan hak cipta


z-Transforms 222

u (n )

Example 4.25

ir
Find x (n ) using (i) long division and (ii) partial fraction for X (z ) given by

Bahan dengan hak cipta


223 Digital Signal Processing

2+3 z -l
X (z ) =
Also verify the results in each case for 0 < n < 3.
So lu tion
(i) Long Division Method
2 + 3 z ~'

X (z ) 1 + — Z_1 + — Z~2 - — Z~3


=
8 8
2 + i 2-i_Z2-2+il2-^
8 32
1+ - Z _ I + -Z~2 - — Z -3
2 + 3z -l
8 8
2 + —z“i + —z-2 - —z-3

— z"1 -— z-2 + — z-3


2 4 4
- Z_1 + - z-2 + — z"3 - — Z-4
2 8 16 16

- - Z -2 + — Z -3 + — z -4
8 16 16
_Zz-2 + 352-3_JL2-4+JL2-S
8 32 64 64

ii
32
Therefore, X (z) =2 + ^ 2“1 - ^ 2 2 + ^ 2 3
2 8 32
„1 7 41
Taking inverse 2-transform, we get x(n) = ’2’ 8 ’ 3 2 ’

(ii) Partial Fraction Expansion Method


2 + 32“1
X(z) =

1+ z '1 l + i 2 - i 1 - i z -1

2 + 3 -l 8
2
5
*‘l —l
Bahan dengan hak ci|
224 Digital Signal Processing

2 + 3 z*1 8
Ao =
3
z_1 = - 2

2 + 3z -l II
A,=
15

_HD+_ 14
15
)
X (z ) +

SL
Hence x(/i) = Z“‘ "[X(z)]
1+*"1=I42 - > +—f-—Y +— u (n )

When n = 0, x(0) = 2 When


2
f—T
n = 1, x(l) = 1/2 When n = 2, 5 3V. 2.1 15V4/
x(2) = - 7/8 When /1 = 3,
x(3) = 41/32
9 1 _7 41 ’2’
Therefore, x(n) = 8* 32
T
Therefore, the results obtained in each case are identical.
4.4.3 Residue Method
This method is useful in determining the time-signal x (n ) by summing residues of
[X { z )z n ~ J] at all poles. This is expressed mathematically as
x(n ) = V residues of [X (z ) z n !] (4.14)
all poles
XU)
where the residue for a pole of order m at z = a is
m -1
Residue = ---------- lim m —[(z ~ «) m
X (z )z " _
'] (4.15)
( m — 1 ) ! z- d2
io
Using the residue method, determine x(/i) for X (z ) =

E xamp le 4. 26

(z-l)(z-2)
So lu tion X (z ) has two poles of order m = 1 at z = 1 and at z = 2. The
corresponding residues can be obtained as follows:

Bahan dengan hak cipta


z-Transforms 225

For poles at z = 1
Residue = -rr lim ^ ^ (2-1)
2-2 n-
0! z -* i dz ( z - l ) ( z1- 2 )

^{z-2 Bahan dengan hak cipta


226 Digital Signal Processing

For poles at z = 2
|0
i zz n- 1
Residue = — lim (z - 2)
0! z->2 dz° (z - l)(z - 2)

= lim = 2-2n_1 = 2
z -*2
Therefore,JC(/I) = (— 1 + 2”) u(n )

E xamp le 4. 27 Using the residue method, find the inverse


z-transform of

X (z ) = 1 , ROC: |z| >0.5


(z - 0.25) (z - 0.5)
So lu tion
Let X 0 (z ) = X (z )z n "1
~n — 1
(z - 0.25) (z - 0.5)
For convenience let us write this as

*o<z> =
z ( z - 0.25) ( z - 0.5)
This has poles at z = 0, z = 0.25 and z = 0.5 By
the residue theorem
x(n ) = Re su [X 0 (z )i + R e s [ Xu 0 ( z ) l + R es [ X 0 ( z ) l z
=0 z = 0.25 z = 0.5 ”

.
= R es + R es n
z = o z(z - 0.25) (z - 0.5) z = 0.25 z(z - 0.25) (z - 0.5)
n
+ R es
z = 0.5 z(z - 0.25) (z - 0.5)
(a) For pole at z = 0
R esX 0 (z ) = zX Q (z ) | z =
0z =0

(z - 0.25) (z - 0.5) 2 = 0
0 =

(b) For pole at z = 0.25


R es X 0 (z ) = (z - 0.25)X0(z) |J = 025
2 = 0.25
(z - 0.25) z"
z(z - 0.25)(z - 0.5) 2 = 0.25

Bahan dengan hak cipta


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z-Transforms 227

Yiz)
z (z - e~T)(z - e~2T)
Expressing the above as partial fractions YU) _
Ax t A2
-T T
, where Al and A2 are constants.
z-e z - e -2
The constants A j and A2 are evaluated as Ax = - — a n d A 2 =
1
1-e' 1-e

-T
YU) _ 1-e 1-e
Thus, -T
+
-2 T
z-e z-e
2
So, YU) = -
1 -e-T Z _e-T 1 _eT Z _e-2T

Taking inverse z-transform, we


get
y(nT) = (e-'‘T) + —Kr(e-2nT
1
l-e~J 1-e
Substituting T = 1/50, we get
yin) = 50.5(0.980)" - 49.5(0.961)"
Thus the required output response is
yin) = 50.5(0.980)"- 49.5(0.961)"

s-domain approach:
The above problem can be solved in the s-domain as given below. The analog
transfer function His) is given by

H(s)= *

The given input function is x(t) = e 2t Taking


Laplace transform, we get

Xis) =
s+2
Hence, in the s-plane the output function is given by

Yis) = XisyHis) =
is + 2) is + 1)
In order to have (ideally) the same frequency response at co = 0 for both digital and
analog systems, it is necessary to multiply the digital

Bahan dengan hak cipta


228 Digital Signal Processing

transfer function by a factor of T (as w increases, the two frequency response


functions generally diverge). That is,

5'

YT(Z) = Y(s)
_1
(s + 2)(s + 1)
Expressing as a partial fraction,

s+1s+2
y(s) = _I

Hence, Y(z) = — Yis) = - — -------------------- —


T T s +1 s + 2
Using Table 4.3, we have

y(nT) = ± [enT - e-2nT)

Substituting T = 1/50, we get


yin) = 50le-n/50 - e-™50]
= 50[(1.0202)"n - (1.0408)"1]
= 50(0.980)n - 50(0.961)*
Thus the required output function is
yinT) = 50(0.980)n - 50(0.961)n
Note: Due to the approximation (itroduced by the multiplication of T) in the s-
domain approach, we get a squence that closely matches the sequnce abtained
through 2-domain approach. Determination of value of T which gives the same
squence as that in 2-domain is left as an exercise to the students.
REVIEW QUESTIONS

4.1 How is z-transform obtained from Laplace transform?


4.2 Describe the relationship between Laplace transform and 2- transform.
4.3 Explain z-plane and s-plane correspondence.
4.4 Define z-transform.
4.5 Explain the use of z-transform.
4.6 State the properties of convergence for the z-transform.
4.7 State and explain the properties of z-transform.
4.8 Discuss the properties of two-sided z-transform and compare them with one-
sided z-transform.

Bahan dengan hak cipta


z-Transforms 229

4.9What is the condition for z-transform to exist?


4.10 Explain the shift property of z-transform.
4.11 With reference to z-transform, state the initial and final value theorems.
4.12 What is the z-transform convolution in time domain?
4.13 Describe the process of discrete convolution.
4.14 Define and explain (i) convolution and (ii) correlation
4.15 Define and explain cross-correlation and auto-correlation of sampled
signals.
4.16 Define inverse z-transform.
4.17 Explain various methods of finding inverse z-transform.
4.18 Describe the signals shown in Fig. Q4.18 using sets of weighted, shifted, unit
impulse functions.
x(n) x(n)
<

2 2
2
0.5 1

.1 •11
- 2 - 1 0 1 2 3 0 1 2 1 3
*
- 1
-0.5
Fig. Q4.I8

Ans: (a) x(n) = 0.58(n + 1) + 28(n) + 28(n - 1) + 8(n - 2)


(b) x(n) = S(n +2)- 8(n + 1) + 28(n) + 8(n - 2) - 0.58(n - 3)
4.19 Derive the z-transform offfnT) = sin conT
4.20 Derive the z-transform of f(nT) = cos conT
4.21 Find the z-transform of t 2 e~ at.
Ans: X(z) = 2[z* -z*3 (l-e"1 z'1)]/(1 - e~az~')
4.22 Obtain the z-transform of x(n) = - anu(-n - 1). Sketch the ROC.

Ans: X(z) =-------- —^, | z\ < \ a\


1 -az 1
4.23 Find the z-transform of the following discrete-time signals including the
region of convergence.

(i) x(n) = e~ 3n u(n - 1) Ans: X(z) = - — f a t, ROC: \ z \ > | e"31

(ii) x(nT) = (nT)2 u(nT) Ans: X(z) = --------- ,,, ROC: \z\ > 1
(1-z-1)2’ ''
(Hi) x(nT) = 0 for n < 0
= 1 for 0 ≤n ≤5

Bahan dengan hak cipta


230 Digital Signal Processing

- 2 for 6 <n ≤ 10 = 3 for n


>10
Arts: X(z) =1 + z~ 1 + z'2 + z~3 + z~ 4 + z~ 5 +
2( Z - 6 + Z - 7 + Z - 8 + . . . + Z - I 0 ) +
n
3(z- +*-■« +

4.24 Determine the z-transform of the following sequences

(a) u(n - 4) Ans: -- ---- | z | > 2

(b) e Jn*'4u(n) AnS'. ------ r—r-j r


1 _ eJ**/4z-i
(c) S(n - 5) Ans: z~5

(d) U X u ( - n ) Ans: —1—, | z | < 4


\3J l-3z 3

(e) 3nu(n - 2) Ans: 9z 2


,
1 - 3z
4.25 Find the z-transform of the sequence x(n) = na,lu(n)
Ans: X(z) = az 1 .. .

4.26 Find the two-sided z-transform of


x(n) = (1/3)n n≥0 = (-2)n n < -
2
Ans: X(z) =
2+2
z-l
3
4.27 Use convolution to find x(n) if
X(z) is given by
X(z) =
1

Ans: x(n) = u(n) +

4.28 Find y(n) using the convolution property of z-transform when


x(n) = (1, 2, 3, i, - 2, II and h(n) = {1, 1, U

Ans: y(n) = 2,3, 3,3,3,2,0,21


T
4.29 Convolve the sequences x(n) and h(n) where
x (n) = 0, n < 0 = an, n ≥0

Bahan dengan hak cipta


Ans: r(i)y(n) = === 1,2,3,
v(l)y(n)
x(n)
h(n) b4 \=
(a. ,-.b24,16
1,4,10,20,25,
T an -<0
0, }10,
u (n), if\1,2,3,4
- 9,19,36,
a # b-14,33,0,7,13, -18,16,-7,5,- 3
(a) x2(n) = n •j- f- «|W = j ^
(b) x3 (n) =
SpecifyAns:
the
j|'
bT,n≥0
(ii)
answers
1,2,3,
y(n) = na 4 ~ u(n) (or) nb
n 2
, h3(n) =
4,3,~ 2,1
if (i) an 1*b and 2(ii) aT=n 6.
n ! T i fz-Transforms
u(n),
Note (a - b ) = (a - b) (a ‘ + a" - b + a " b + ... + a b - + b ~ 4.30
n n 4,11,20,30,20,11,4 3 2
a=b
n 2
231
n 1 i

4.31 Compute and sketch the convolution y(n) and correlation r(n) sequences
Compute
for the
theycross-correlation
2following
(n) = sequence
pair of signals . r^ (l) of the sequences
2,-1,3, 7,1,2,-3
T
r2(n) =
r3(n) = 1,-1,2,-2,4,1,-2,5)
T
Ans:
4,11, 20,30,20,11,4

1,4,10, 20,25, 24,16

4.32 Find the autocorrelation sequence of the signal


x(n) = anu(n) for-1 < a < 1

Ans: r^tt) = j a *1 * for - oo < / < oo


1-a
4.33 Find the final value of the signal corresponding to the z-transform

Bahan dengan hak cipta


2z -l
X(z) = Ans: 10
1 2
1-1.8Z- +0.8 z-
232 Digital Signal Processing

4.34 Prove that the final value of x(n) for X(z) = -------------------------------- is
' ' ' (z - l ) ( z - 0.2)
1.25 and its initial value is unity.
4.35 Determine z-transform for the sequences given below
x(0) = 1 x(l) =4.7
x(2) = 0 x(3) = 0
x(4) = 0.75
x(5) = -J2 x(n) =
0, n ≥ 6
Ans: X(z) = 1+4.7z~' + 0 . 7 5 z + -fH z*5
4.36 Determine x(0), x(l)t x(2), x(3), x(4) for the following functions of z.
2z + z -2
(i) X(z) =
-1
, 2 +2
Ans: x(0) = 2, x(l) = Ot x(2) = - 2, x(3) = 1 and x(4) = 2 1

(ii) X(z) =
l-lz"

Ans: x(0) = 1, x(l) = j-, x(2) = x(3) = ~Q and x(4) = ~


2 4 o 16
4.37 Find the inverse z-transform of the following
.2

(i) X(Z) =
(z - 1 ) ( z - 0.2)
Ans: x(n) = [1.25 - 0.25 (0.2)n] u (n) x(n) =
[1, 1.2, 1.24, 1.248 ...}

(ii) X(z) =
(z - 0.5 - j0.5)(z - 0.5 + j0.5)
Ans: x(n) = -j(0.5 + j0.5 )n + 1 - . '0.5 -j0.5)n * 1 z
(Hi) X(z) =
(z - u
n(n- l ) u ( n )
Ans: x(n) =

z(l-e-T)
(iv) X(Z) =
T
(z-l)(z-e ) Ans:
x(n) = (1 - e~nT)u(n)

Bahan dengan hak cipta


X(z) = -l -i with ROC: \z\ <3

Ans: (4)n u(n)


Ans:

4.39 Determine x(n) for X(z) = - -l


ar-a
4.38 Find the inverse of z-transform of ---------------
u(n) z-Transforms 233

1-z
X(z) = 1 - 3z
Ans: x(n) = (- 3)n(1' 1+u(n)
z
4.40 Find the sequence x(n), n >0 given that

Ans: x(n) = 4(-l)n -3\ 1 | u(n)

4.42 Using partial fraction expansion, determine x(n) for X(z) given by / \ v/ \
(z-0.5)
(a) X(z)
=-Jz -W)Tz-)
Ans: x(n) = [2.5 - 1.5(0.8r~s]u(n - 2)
(b) X(z) = 0.5 z
zJ - z + 0.5

-'is)
Ans: x(n) = I -4= | sin sI — Iu(n)
“(¥

4.43 Find x(n) ifX(z) =


1--Z-1

n-3
Ans: x(n) = ( — u(n - 3)

z+3
4.44 Find x(n) ifX(z) =
Tf^
f i y -l7 n-8
i -7) + I — ) u(n - 8)
Ans:x(n) = y—j u(n

4.41 Using partial fraction, find x(n) for

Bahan dengan hak cipta


I — 2for the(ii)
J iregion of convergence 2\z\ > — 1n\ -22
4.45 X(z)
Find
Ans:
(b)
(ii) =
x(n)
wherethe
(a) 2
causal
=x(n)
the
=X(z)
x(n)=\ 2)
ROC=
z••14,11,8,5,2,0
2
0{l) z
is n2
signal
V 3z
l-
4 2
z
(i)--x(n)
4z
|z|
-20n[L)>+1
z for
1
- nX(z)
and
~' = \z\ < —
n - 71.565°l-z
using
- 1
+
\u(n)
1 +
0.5
the z z
-l
long
-
Ans: x(n) -JToi-4=
i f (a)= x(n) is causal and (b) x(n)\ is anti-causal
8 + +

T
+ 6[ n ( n - l )u/ 2] \ ±
division method. 1,4,7,10,13,-
234 Determine
4.48 Digital Signal causalt signal cos
theProcessing x(n), having the z-transform
'n l 4 13_ 40
l y/2) v 4
4.46 Find
Ans:(i)x(n)= r
the inverse z-transform
3’ 9’ 27’ of 81’"

-121,40,13,4,1,0
T
4.47 Using long division, determine the inverse z-transform of
1 + 2z_1
(n)

4.49 Using (i) the long division method, (ii) partial fraction method and ( H i )
residue method, find x(n) and verify the results in each case for n in the range
0 ≤n ≤ 3.

« = T^E

Bahan dengan hak cipta


z-Transforms 235

Ans: (i) x(0) = 1, x(l) = 3.25, x(2) = 0.8125, x(3) = 0.203125


(ii) x(n) = - 125 (n) + 13(0.25)n
x(0) = 1, x(l) = 3.25, x(2) = 0.8125, x(3) = 0.203125
(iii) x(n) = 13(0.25)"
x(0) = 13, x(l) = 3.25, x(2) = 0.8125, x(3) = 0.203125 Here x(0) =
13 is not correct. The correct initial value [x(0) = 1] can be obtained using
the initial value theorem.
2
(b) X(z) = 1.8z(z + 0.1667) z
- 0.4z - 0.05
Ans: (i) x(0) = 1.8, x(l) = 1.02, x(2) = 0.498, x(3) = 0.2502
(ii) x(n) = 2(0.5)" - 0.2( - 0.1)"
x(0) = 1.8, x(1) = 1.02, x(2) = 0.498, x(3) = 0.2502
(iii) x(n) = 2(0.5)" - 0.2(- 0.1)"
x(0) = 1.8, x(l) = 1.02, x(2) = 0.498, x(3) = 0.2502 4.50 For a low pass
RC network (R = IMG and C = lpF) . Determine the output response for n in the
range 0 ≤n ≤3 when the input has a step response of magnitude 2V and the
sampling frequency fs = 50 Hz. Confirm the calculated values using the principle of
discrete-time convolution.
Ans: y(0) = 2, y(l) = 3.96, y(2) = 5.88 and y(3) = 7.77 v

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Chapter

Linear Time Invariant Systems

5.1 INTRODUCTION
The techniques and applications of Digital Signal Processing (DSP) require the
understanding of basic concepts and terminology of discretetime, causal, linear and
time invariant systems. A DSP system can be represented by the block diagram
shown in Fig 5.1. The input signal, x (n), is the system excitation, and y (n) is the
response of the system. The system output y ( n ) is related to the excitation
mathematically as y ( n ) = F Mn)] (5.1)
where F is an operator.

x(n) DSP System


y(n) =F( x(n))
Fig. 5.1 A DSP System

5.1.1 Impulse Response


The unit sample response or simply the impulse response h ( n ) of a linear time
invariant system is the system’s response to a unit impulse input signal [ 5 ( n ) ]
located at/i = 0, when the initial conditions of the
system are zero.
5 (n) n =
= 0 n #
y(n) = h(n)
0

y(n) = h (r>)

Fig. 5.2 Impulse Response of a Linear Time Invariant Causal (LTIC)


System

(5.2)

Bahan dengan hak cipta


Linear Time Invariant Systems 237

If the input to the system is the time shifted version of the 8 function, the output
is a time shifted version of the impulse response.The result shown in Fig. 5.2 is due
to the time invariance of the system. In this linear system, application of a 8( n - k )
at the input will produce a h ( n - k ) at the output as shown in Fig. 5.3.
The shifted impulse sequence is given by
8(n-A) =
x(n)=a8(n-k) • • y(n) = ah(n-k)

Fig. 5.3 Time Shifted Impulse Response


1, n = k 0, n * k
where k is an integer. The graphs of 8 (n) and 8 ( n - 2) are shown in Fig. 5.4.

m
i- -

n
- -1 3 4
1 0 1 2 3
n
Fig. 5.4 (a) Graphical Representation of the Unit-impulse Sequence S(n) and (b)
The Shifted Unit-impulse Sequence S ( n - 2 )

The response of the LTI system to the unit impulse sequence 8 ( n ) i s represented
by h ( n ) f i . e , h ( n ) = F [ 8( n ) ] and hence A ( n - k ) = F [8 ( n - k ) ]

Therefore, y ( n ) = F [ x ( n )l = £ * (« 8 (n - k )
F k = -oo

= X x (k )F [ & (n-A)l
OP

= £ x (k) h (n -k)
k = -o°
= x (n) * h (n)
oo

or, equivalently, y ( n ) = ^ h (k) x (n - k) = h (n) * x (n)


k=-°°

Bahan dengan hak cipta


238 Digital Signal Processing

This expression gives the output response y (n ) of the LTI system as a function
of the input signal x ( n ) and the unit impulse (sample) response h (n) and is
referred to as a convolution sum.
5.1.2 Unit Step Response [u (n)]
The unit step sequence u ( n ) is defined by
0, n < 0
a(n)= (5.3)
The shifted unit step sequence u (n - k ) is given by
'0, n < k

u(n-k) =
1, n≥k

u(n
)
The graphical representations of u (n) and u (n - 2) are shown in Fig. 5.5.
k u{n- 2)

1 ~ -

(a) (b)
;
- 2 -1 0 1 3 n
2
- - 1 0 1 n
Fig. 5.5 (a) The Unit-step Sequence u (n) and
(b) The Shifted Unit-step Sequence u (n - 2).

The step response can be obtained by exciting the input of the system by a unit-
step sequence, i.e, x (n) = u ( n ) . Hence, the output response y ( n ) is obtained by
using the convolution formula as

y ( n ) = Y J h (A) u ( n - k )
A = -oo

Relation Between the Unit Sample and the Unit-step Sequences


The unit sample sequence 8 ( n ) and the unit-step sequence u ( n ) are related as

u(n)= Y 8 (m), 5 (n) = u (n) - u ( n - 1) (5.4)


m=0

5.2 PROPERTIES OF A DSP SYSTEM


The properties of linearity, time invariance, causality and stability of the difference
equations are required for the DSP system to be practically realisable.

Bahan dengan hak cipta


Linear Time Invariant Systems 239

5.2.1 Linearity
If y x (n) and y2 (n) are the output sequences in response to the two arbitrary input
sequences x x ( n ) and x2 (n), respectively, then the condition for the DSP system
#
to be linear is given by
Ffaxx (n) + a2x2 (/i)] = ax F [xx («)] + a2 F [x2 (n)]
= a x yx (n) + a2 y2 (n) (5.5)
where ax and a2 are two arbitrary constants andF (1 denotes a signal processing
operator performed by the DSP system. This condition is often called the principle
of superposition. All linear systems possess the property of superposition.

Example 5.1 Check whether the following systems are linear.


(a) F [x (n)] = a nx (n) + b
(b) F lx ( n ) ) = e x(a)
Solution
(a) F [x (n)] = a nx (n) + b
F [xx (n) + x2 (n)] = an [xx (n) + x2 (/i)] + b F [xx (rc)] + F [x2
(n)] = [a nxx (n) + b] + [an x2 (n) + 6]
Since F [xx (n) + x2 (n)] * F [xx (n)] + F [x2 (/i)], the system is nonlinear,
when 6 * 0 .
(b) Given F [x (»)] = e x<n)
F I*, (n) + x2 (it)] = e x‘<n) + Xs (n) = e x> (n,e x»(n)
+ x2 (n)
x (n)
F [xx (it)] + F [x2 ( n ) ] =- oe l
e + F [x2 (n)], the system is nonlinear.
Since F [xx (n) + x2 (n)l * F \pcx (rc)l

Example 5.2 Check whether the following systems are linear.


1N~x
(a) y ( n ) = — £ x ( n - m )
** m = 0
(b) y (n) = lx (n)]2 Solution
1N~x
(a) Let yx ( n ) = — £ x x (n-m) and

m=0 1

N
then, ~l

y-l (»)= T7 X X n
2< - N- 1 N -1
y ( n ) = yi I n ) + y2 ( n ) = — £ xx (n-m) + — £ 2 (n-m)
x
^ m=0 N m= m=
0 0

Bahan dengan hak cipta


240 Digital Signal Processing

N-1
= ~: X ( H - ™ } + *2 (n-m)]
™ m=0
1 ^-1
= T7
iV
S^
m=0
Therefore, this system is linear.
(b) y ( n ) = [ x x (n) + x 2 (n)]2
= [xx (re)]2 + 2 (re) x2 (re) + [x2 (re)] 2 *
Xj

lxx (re)] 2 + [x2 (re)]2 Therefore, the system is non-


linear.

Example 5.3 Check the following responses of the digital filters for
linearity,
(a) y (re T) = F [x (re T)] = 9 x2 (re T- T)
(b) y (re T) = F [* (re T)\ = (re T)2 x (re T + 2 T)
Solution
(a) For a constant, a, other than unity,
F[ox(n T)] = 9 a2 x2 (re T- T)
and aF[x(n T))=9ax2 (n T - T)
Here, F [ a x ( n T ) ] # a F [ x (nT)\ and hence the filter is non-linear, (b) In
this case,
F [ox, (nT) + b x2 (reT)] = (nT)2 (ax1 (nT + 2T) + b x2 (nT + 2 T ) ]
= a ( n T )2 xj( n T + 2 T ) + b ( n T )2 x 2
( n T + 2T ) = a F [ x x (reT)] + b F [ x 2 (reT)]
Therefore, the filter is linear.

Example 5.4 Test the following systems for linearity (a) y (t)
i = 5 sin x (t) and (b) y (t) = 7 x (t) + 5
»
I
Solution
(a) y (t) = 5 sin x (t)
F ]xj (f)] = yj (t) = 5 sin xx (t)
F ]x2 (t)] = y2 (t) = 5 sin x2 (t)
F [xj (*)] + F [x2 (f)] = 5 [sin xx (t) + sin x2 (f)]
whereas, F [a^ (t) + x2 (t)] = 5 [sin[a^ (t) + x2 «)]]
Here, F [xx (t) + x2 (t)] * F [xx (£)] + F[x2 (f)]
and hence the system is non-linear.
(b) y (f) = 7 x (£) + 5
F [xj (t)] = y, (t) = 7 x, «) + 5
F [x2 (*)] = y2 (t) = 7 x2 (t) + 5

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242 Digital Signal Processing

Therefore,

aF [*, (t)] + bF [x 2 (»] = a +b + ^ {t)

dt
d2
F [a X j (t ) + b x 2 (*)] = —s- [a y 2 (*)+ 6 y2 (*)]
df2
+ [ay2 (£) + fey2(*)l — (a y 1 (t) + b y 2 (*)) + 1

dr dr at
+ by 2 (t) ^ + ay x (t) + b y 2 (t)
Here aF [x x (t)] + bF (x2(*)) * F [a x j (*) + &x2U)J and hence the system is non-
linear.
5.2.2 Time- Invariance
A DSP system is said to be time-invariant if the relationship between the input and
output does not change with time. It is mathematically defined as •
if y (n ) = F [x (n)], then y (n - k ) = F [x (n - k)] = 2-* F [x (n )] (5.6)
for all values of k . This is true for all possible excitations. The operator 2“* represents a
signal delay of k samples.

E xamp le 5. 8 Determine whether the DSP systems described by the


following equations are time invariant.
(a) y (n ) = F \x (n)l = a n x ( n ) .
(b) y (n) = F [x (nj\ = a x ( n - 1) + 6 x (n - 2)
So lu tion
(a) The response to a delayed excitation is
F [x (n - &)] = an [x (n - k)\
The delayed response is y (n - k ) = a (n - k ) [x (n - k )\
Here F [x (n - k)] * y (n - k ) and hence the system is not time invariant, i.e. the
system is time dependent.
(b) Here, F [x (n - k )\ = a x [(n - k ) - 1] + bx [(/1 - k ) - 2]
-y(n-k)
Hence the system is time invariant.

E xamp le 5. 9 Check whether the following systems are linear and


time invariant.
(a) F [x (TZ)] =n [x (n)]2
(b) F [x (TJ)1 = a [x (n)]2 + b x (n )
So lu tion
(a) (i) F [x (TI)1 = n [x (n)]2
Here, F [*j (TI)] = n [xj (TI)]2 and

Bahan dengan hak cipta


Linear Time Invariant Systems 243

F |*2 (/i)l = TI [*2 (TI)]2


Therefore, F [*, (TI)] + F [*2 (TI)] = n [(xj (TI)]2 + (x2 (TI)}2]

Bahan dengan hak cipta


244 Digital Signal Processing

Further, F [x x (n ) + x 2 (n)] = n [x x (n) + x 2 (n)]2


= n [{xjfn)}2 + \x 2 (n )) 2 + 2 x x (n ) x 2 (n )\
Here, F [x x (n ) + x 2 (n)] * F [x x (n)] + F [x 2 (n)] and hence the system is non-
linear.
(ii) F lx (n)] = n [x (n)]2 = y (n)
The response of delayed excitation is F lx (n - k )\ = n [ x ( n - k )]2 The delayed
response is
y ( n - k ) = ( n - k ) [x ( n - k )]2
Here, y ( n - k ) * F [x ( n - A)]
Therefore, the system is not time invariant, i.e. time dependent.
(b) ( i) F[*( re)] =a (x (re)]2 + 6x (re)
Here, F [xj (re)] = a [x 1 (re)]2 + bx j (re) and F lx2’(re)] = a [x 2 (re)]2 + bx 2 (re)
Therefore, F [x; (re)] + F [x 2 (re)] = a [fxj (re)}2 + (x2 (re)]2]
+ 6 [xj (re) + x2 (re)]
Also, F [x! (re) + x2 (re)] = a [x! (re) + x2 (re)]2 + b [xj (re) + x2 (re)]
= a [{xj (re)}2 + lx2 (re))2 + 2x 1 (re) x2 (re)] + bx 1 (re) + bx 2 (re)
Here, F [xx (re)] + x2 (re)] * F [xx (re)] + F [x2 (re)]
Therefore, the system is non-linear.
(ii) F [x (n)J = y ( n ) = a [x (n))2 + b [x (n)]
The response of delayed excitation is F [x ( n - k )\ = a [x (n - k)]2 + b x ( n -
k)
The delayed response is y ( n - k ) = a [x ( n - k )]2 + 6 [x ( n - k)]
Since y (n - k)\ = F [x ( n - k )] 9 the system is time invariant.
5.2.3 Causality
Causal systems are those in which changes in the output are dependent only on the
present and past values of the input and/or previous output values, and are not dependent
on future input values. For a linear time invariant system, the causality condition is given
by
h (n ) = 0 for n < 0

E xamp le 5. 10
Check whether the DSP systems described by the following
equations are causal.
(a) y (n ) = 3 x (n - 2) + 3 x (n + 2)
(b) y (n ) = x (n - 1) + a x (n - 2)
(c) y (n ) = x (- n )
So lu tion
(a) y (n ) = 3 x (n - 2) + 3 x (n + 2)
Here,y (n) is determined using the past input sample value 3 x (n - 2) and future
input sample value 3 x (n + 2) and hence the system is non-causal.
Linear Time Invariant Systems 245

(b) y ( n ) = x ( n - 1 ) + a x ( n - 2 )
Here, y (n ) is computed using only the previous input sample values x (n - 1)
and a x (n - 2). Hence, the system is causal.
(c) y (n ) = x ( - n )
Here, the input sample value is located on the negative time axis and the
sample values cannot be obtained before t = 0 . Hence the system is non-
causal.

Example 5.11 The discrete-time systems are represented by the following


difference equations in which x ( n ) is input and y ( n ) is output.
(a) y ( n ) = 3y 2 ( n - 1) - n x ( n ) + 4 x ( n - 1) - 2 x ( n + 1) and (b) y ( n ) = x
( n + 1) - 3 x ( n ) + x (n - 1); n > 0
Are these systems linear? shift-invariant? causal? In each case, justify your answer.
So lu tion
(i) Linearity
The real condition for linearity is F \ a x (n )] = a F [ x (n)]
(a) F [a x (n)] = ay (n ) = 3a2y2 (n - 1) - anx (n) + 4ax (n - 1) - 2a* (n +
1); aF \x (;i)] = a (y (a)) = 3ay2 (n - l)- anx (n ) + 4a* (n - 1)- 2a* (n +
1)
Here F [a* ( n ) ] * a F [ x ( n ) ] . So, the system is non-linear.
(b) F \ax (/i)l = ax (n + 1) - 3a* (n ) + ax (n - 1) a F [x (rc)] = a x (n +
1) - 3a* (n ) + a x (n - 1)
Here, F [a* (n)] = a F [ x (a)]. So, the system is linear.
(ii) Shift invariance
The necessary condition for time invariance is
y(n-k) = F [x (n -k)]
(a) Now,F I* (n — /e)l - 3y2 (n- k-1 ) - n x (n - k ) + 4* (n- k-1 ) - 2* (n-
k +1 ) y (n - k) = 3y2 (n-k- 1) - (n- k ) x (n - k ) + 4* (n- k-1) - 2* (n-
k +1 )
As y ( n - k ) * F [* (n—A)], the system is shift variant.
(b) F I* ( n - k ) \ = * ( n - k + 1) - 3* ( n - k ) + * ( n - k - 1) y ( n
- k ) = x ( n - k + 1) - 3 * (n - k ) + x ( n - k - 1)
As y ( n - k ) = F [ x ( n - A)], the system is time-invariant.
(iii) Causality
The required condition for causality is that the output of a causal system must be
dependent only on the present and past values of the input.
(a) From the given equation, it is clear that the output y ( n ) is dependent on a
future input sample value* ( n + 1). Hence the system
is non-causal.
(b) Since the outputy (n) of the given difference equation is dependent on a future
input sample value * ( n + 1), the system is non-causal.
Paley-Wiener Criterion
The modulus of the frequency response (or the amplitude response) is an even function of
frequency, whereas the phase angle, i.e. the phase response is an odd function of frequency.
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Linear Time Invariant Systems 253

Hence, the impulse response is h (n ) = a8 (n ) + b


When n = 0, h (0) = ab (0) + b = a + b
When n = 1, h (1) = ab (1) + b = b
Here, h (1) = h (2) = - = A(A) = 6
Therefore,
h in ) = a + b when n = 0
=b when n * 0
The necessary and sufficient condition for BIBO stability is
oo

X i A<A) i < °°
k=0
oo
Therefore, £ |A(A)| = IA CO) | + |Ad)| + |A(2)|
k=0
+ ••• + \h ik )\ + •••
= |a + 61 + 16 | + 161 + ••• + 161 + •••
This series never converges since the ratio between the successive terms is one.
Hence, the given system is BIBO unstable.
(c) y in ) = e-*"’.
If x (n)= 8 («),then y in ) = h in ).
Hence, the impulse response is h in ) = e_8,nl When /1 = 0, hiO ) = e ~ & C o ) = e’1
When n = l, A(l) = e"8 (1) = e° = 1 In general,
h (n ) = e"1 when n = 0
=1 when n * 0 .
The necessary and sufficient condition for BIBO stability is

£ |A(A)| <~.
A=0

Therefore, £ \h (k)\ = |A(0)| + |A(1)| + |A(2)| + ••• + \h (k ) \ k=o


— € 1 + 1 + 1 + 1 ••• + 1 •••
Since the given system never converges, it is BIBO unstable, (d) y (n ) = a x (n +
1) + b x (n — 1).
If x(n ) = 8 (n), then y (n ) = h (n ).
Hence, the impluse response is h (n ) = ab (n + 1) + 65 (n-1).
When n = 0, h (0) = a8 (1) + 65 (-1) = 0
When n = 1, h (1) = a5 (2) + 65 (0) = 6
When n = 2, h (2) = a8 (3) + 65 (1) = 0
In general,
h in ) = 6 when n = 1
otherwise

when n = 1

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254 Digital Signal Processing
=0 otherwise
Linear Time Invariant Systems 255

The necessary and sufficient condition for BIBO stability is


oo

X \Mk)\ < o o * =0

oo

Here, X |A(A)| = |6|


* =0
Hence, the given system is BIBO stable if | b | < .
(e) y (n) = ax (n ) . x ( n - l )
If x (n ) = 8 (n), then y (n ) = h (n ).
The above equation can be changed into h (n) = a 8 (n ). 8 (/i-l) When n = 0, h (0) =
a8 (0). 8 (-1) = 0 When n = l, h ( 1) = a8 (1). 8 (0) = 0 When n = 2, h (2) = a8 (2). 8
(1) = 0
The necessary and sufficient condition for BIBO stability is
oo

, X \h(k)\ <~
k =o

Here, £ | h (k) | = 0.
0
So the given system is BIBO stable .
(f) y (n ) - max. of [x (n), x (n - 1), x.(/i - 2)] .
If x (n ) = 8 (n), then y(n) = h(n ).
The above equation can be changed into h (n) = max .
of [8 (n), 8 (n - 1), 8 (n - 2)]
When n = 0,h (0) = max . of [8 (0), 8 (-1), 8 (-2)] = 1
When n = 1,h (1) = max . of [8 (1), 8 (0), 8 (-1)] = 1
When n = 2,h (2) = max . of [8 (2), 8 (1), 8 (0)] = 1
When n = 3,h (3) = max . of [8 (3), 8 (2), 8 (1)] = 0
In general,
h (n ) = 1, for n = 0, 1, 2 .
= 0, otherwise U'.e., /i > 2).
The necessary and sufficient condition for BIBO stability is
oo

X | h (k ) | < k =o
CD.

Here,\h (k)\ = \h (0 )\ + |A(1)| + |A(2)| + - + |A(« | +-


A = 0
= l + l + l + 0+-- = 3.
So, the given system is BIBO stable.
(g) y (n ) = average of [x (n + 1), x (n), x (n - 1)1.
If x (n ) = 8 (n), then y (n ) = h (/i).

Bahan dengan hak cipta


256 Digital Signal Processing

The above equation can be changed into


h (n ) = average of [8 (n + 1), 6 (n), 8 (n - 1)]
When n = 0, h (0) = average of [5 (1), 8 (0), 8 (-1)] = —
3
When n - 1, h (1) = average of [8 (2), 8 (1), 8 (0)] = ^
o
When n = 2, h (2) = average of [ 8 (3), 8 (2), 8 (1)] = 0 In general,
h (n ) = ± for n = 0, 1 . o
= 0 for n > 1.
The necessary and sufficient condition for BIBO stability is

£ |A(fc)| <-
*=o •

Hence, £ |A(A)| = |A(0)| + |A(1)| + |A(2)| + - + |A(A)| + -


k=0

= A + l+o + ...2
3 3 3
Hence the given system is BIBO stable.
(h) y (n ) = ax (n ) + b x r (n-1)
If x (n) = 8 (n), then y (n ) = h (n ).
The above equation can be changed into h (n ) = a8 (/i) + 682 (n-1) When n = 0, h
(0) = a8 (0) + 682 (-1) = a When n = 1, h (1) = a8 (1) + 682 (0) = b When n = 2, h
(2) = aS (2) + 682 (1) = 0

Hence, £ \h (k)\ = |A(0)| + |A(1)| + |A(2)| + - + |A(A*)| + - k=o


= |a|+ 16 | +0 + 0
Hence, the given system is BIBO stable if | a | +16 | < «>.

E xamp le 5. 19 Check the BIBO stability for the impulse response of a digital
system given by h (n ) = a n u(n )
So lu tion
h (n ) = a" u(n) => h (k ) = a* u (k )

£ |A(A)| = |a |* = |a01 + la1! + |a2| + ■•• + |a*| + • ■ • = -A- ^=0 1 a


Hence, the given system is stable if a < 1, i.e. a lies inside the unit circle of the complex
plane.

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257 Digital Signal Processing

5.3 DIFFERENCE EQUATION AND ITS RELATIONSHIP WITH


SYSTEM FUNCTION, IMPULSE RESPONSE AND
FREQUENCY RESPONSE
In general, a causal LTI system may be characterised by a linear constant
coefficient difference equation given by
N M

X a
ky (n - w=X b
kx (n -w (5.10)
*=0 k=0

where terms a k represent recursive, and b k non-recursive, multiplier coefficients.


The corresponding system function, impulse response and frequency response are
determined as given below.
Taking z-transform of both sides of Eq. 5.10 using the linearity property, we
obtain
N M '

X a
kz (y -*)1 = X b iP I* ( n -
A=0 k=0

Therefore, the system function H (z) is

yu) (5.11)
H(z)
X(z
=
)

Taking inverse z-transform of the system function H (z) given in Eq. 5.11, we
obtain the unit sample response, h (n),
h (n ) = Zrl lH (z ) 1 (5.12)
We know that the output Y ( z ) - H (z). X (z)
If x (n ) is a unit sample response or impulse response X (z) = 1 and hence
Y ( z ) = H (z). Therefore, y (n ) = Z"1 [H (z)]
If // (z) is known, a difference equation characterisation can be determined by
reversing the above operations. First H (z) is expanded in terms of negative
powers of z and made equal to Y (z )/X (z). Then cross multiplying given equation
(5.11) and the inverse transform gives the difference equation shown in equation
(5.10)
Taking Fourier transform of the unit sample response h (n), we obtain the
frequency response given by

(5.13)
H (eh = Xk=-~>
h
M e~j(m

= H (z)

z-Transform Solutions of Linear Difference Equations


The z-transform can be used to determine the solution y (n), for n ≥ 0, of
the linear, time-invariant, constant coefficient difference equation that
258 Digital Signal Processing
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Linear Time Invariant Systems 259

characterises a causal discrete-time system. For finding the solution of an M th


order difference equation, M initial conditions and the input signal x ( n ) must be
known. Using the time delay property of the z-transform, the initial conditions y (-
1), y (- 2) ••• y (- M) are incorporated directly in the solution obtained with 2-
transforms, as given below.
Z\y(n-l)\=z-1Y(z)+y(-l) (5.14a)
Z \ y ( n - 2)] = z-2 Y (z) + z'1 y (-1) + >- (-2) (5.14b)
Z (y (n - 3)] = z~3 Y (z) + z-2 y (- 1) + z_1 y (- 2) + y (- 3) (5.14c)
Here, the negative powers of 2 get decreased and negative arguments ofy(n) are
increased. The initial conditions are not required for the input signal when the 2-
transform of the delayed input signals x ( n - k ) in the difference equation are
computed. The time-delay theorem applies to the input signal in which the input
signal is equal to zero for n < 0 when single-sided 2-transforms are used.
Infinite Impulse Response (HR) System
An LTI system is said to be an infinite impulse response (HR) system if its unit
sample response h ( n ) is of infinite duration. A recursive filter that involves
feedback has an impulse response which theoretically continues for ever, and hence
it is referred to as an HR filter.
Finite Impulse Response (FIR) System
An LTI system is said to be a finite impulse response (FIR) system if its unit
sample response h (n) is of finite duration. Since the number of coefficients must
be finite, a practical non-recursive filter is referred to as FIR filter.
The difference equation given in Eq. 5.10 represents a FIR system if a0 * 0 and
a k = 0 for k = 1, 2, •••, N. Otherwise, it represents either an HR or FIR system.
Substituting a k = 0 for k = 1, 2, •••, N in Eq. 5.10 gives

atf (n - 0) = b k x (n - k)
k=0
Rearranging, we
get
x (n -k)
(5.15)

From this equation, it is clear that the present output sample value depends only
on present and previous inputs.
Comparing this equation to that of convolution, we can recognise b k /a 0 as h
( k ) t the value of the unit sample response h ( n ) at time k . Hence h ( n ) is given
by

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1 2 1
Taking
Example
Example
Solution Given
5.20 yy
z-transform,we
5.21 (z)
(n) ++ -
Determine
h get
( ^z'
n )
y Y
(nH b(z)
-(z)
/a
1)
0++ -
and
, z"
^0y ≤
its
, Y
n (
≤ zm
poles
(n - )
2)
2)
otherwise =and
=
= x
xX (z)
zeros
(n)
(n) ++
+ z"
if
xx (n
(n X- 1)
( z -)1) 4(5.16)
____ A DSP system is described
k 0 8 by the linear difference
=
As this is obviously of finite Y (duration,
z ) = = it represents1 + z -l a FIR system. z (z + 1)
equation H (z) =
y (n) = 0.2 x (n)X ( z- )0.5 * (n - 2) + 0.4+* (n
1+32-1+l2-2 Z
2
—-z 3)+ —
260Given
Digital Signal
that Processing
the digital input sequence {-1, 1, 0, -1)8 is applied to this DSP8
system, determine the corresponding digital output sequence.
Solution Taking e-transform of the given linear difference equation, we get
Y (z) = 0.2 X (z) - 0.5 z'2 X (z) + 0.4 z’3 X (z)
Therefore,

H (z) = = 0.2 - 0.5 z-2 + 0.4 z-3


X (z)
The given input sequence isx (n) = (- 1,1, 0, -1) and its z-transform is X ( z ) = -
1 + z-1 - z-3 Therefore, Y ( z ) = H ( z ) X (z)
= - 0.2 + 0.2 z_1 + 0.5 z ~ - 1.1 z-3 + 0.4 z~* + 0.5 z-5 - 0.4 z'6 Taking
2

inverse z-transform, we get the digital output sequence y (n) = {- 0.2, 0.2, 0.5,
- 1.1, 0.4, 0.5, - 0.4)

z (z +1)

R)K)

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Linear Time Invariant Systems 261

(c) The magnitude response | H (e-740) | is an even function of co and


symmetrical about n.
(d) The phase response LH(e j ( 0 ) is an odd function of co and antisymmetrical
about n.
5.4.2 Frequency Response of an Interconnection of a Systems Parallel
Connection
If there are L number of linear time invariant systems in the time- domain
connected in parallel, the impulse response h ( n ) of the resultant system is given
by

h ( n ) = X h k (n)
k=\
where h k in), k = 1, 2, •••, L is the impulse response of the individual systems. By
using the linearity property of the 2-transform, the frequency response of the overall
system is
L
H (z) = X H k ( z ) = H 1 ( z ) + H 2 ( z ) + - + H L ( Z ) (5.17)

.•
k=l
where z = e J ( 0 . Here H k (e J( * ) is the frequency response to the impulse response
h k (n).
From the above discussion, it is clear that the parallel interconnection of systems
involves additivity in both the time and frequency domain.

(a)

o- HM H2(Z HL(Z)
x(n) yi(") ) y2(n) Y L -M yin)

(b)
Fig. 5.7 (a) Parallel and (b) Cascade Interconnection
of Linear Discrete Time Systems
Cascade Connection
If the L linear time invariant systems are connected in cascade, the impulse
response of the overall system is

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262 Digital Signal Processing

h i n ) = i n ) * h 2 ( n ) * ••• h L i n )

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Linear Time Invariant Systems 263
Using the convolution property of the z-transform, we get
H (z) = H , (z) H 2 ( Z ) H L (Z) (5.18)
Hence, H (e J t o ) = H x (e j < 0 ) H 2 (e j ( 0 ) ~ H L (e j v t )
Here we observe that the cascade connection involves convolution of the
impulse responses in the time domain and multiplication of the frequency responses
in the frequency domain.
5.4.3 Energy Density Spectrum
In the digital system, the spectrum of the signal at the output of the system is
Y ( z ) = H ( z ) X ( z ) \z=eJ<0 (5.19)
Hence, Y (e/w) = H (e n X (e7<0). This is the desired input-output relation in the
frequency domain, which means that the spectrum of the signal at the output of the
system is equal to the frequency response of the system multiplied by the spectrum
of the signal at the input.
||X ( e n \ 2 (5.20)
Jt0
As the energy density spectra ofx ( n ) is S^ (en = |X ( e ) |2 and the energy
density spectra of y (n) is Syy (eya)) = | Y (e^) |2, we have
Syy (en = IH (en 1 S„ (en
2
(5.21)
5.4.4 Magnitude and Phase Spectrum
The magnitude response is the absolute value of a filter’s complex frequency
response. The phase response is the angle component of a filter’s frequency
response. For a linear time invariant system with a real-valued impulse response,
the magnitude and phase functions posses symmetry properties which are detailed
below. From the definition of z-transform, H (e J < a ), a complex function of the real
variable co can be expressed as

H(en= £ hWe-j™
n = -oo

h (n ) cos co/i - j ^ h (n) sin con


n = -oo

n = -oo
=
H R {en+jHj(e j ( ») = \H(e j < *)\ e**<a>)

••

7<0
where H R (e ) and Hj (e Jia) denote the real and imaginary components of H (e^).
Therefore, | H (e j a ) | = -] H % (e j a > ) +
Hf ( e J a )
<l> (co) = tan
1
Ho (eJa)

Bahan dengan hak cipta


264 Digital Signal Processing

Also, O (co) may be expressed as

O (co)
=
*

703
The quantity | H (e- ) | is called the magnitude function or magnitude spectrum
and the quantity O (co) is called the phase function or phase spectrum.
\H(e j ( *)\ 2 = H(e j < a )H\e j *)
= H H (e- J b ) )
= H ( z ) H ( z ~ 1) \ Z s e J »
Here the functionH (z_1) has zeros and poles that are the reciprocal of the zeros
and poles of H ( z ) . According to the correlation property for the e-transform, the
function H (z ) H (z-1) is the z-transform of the autocorrelation of the unit sample
response. Then it follows from the Wiener- Kintchine theorem that |H (z) |2 is the
Fourier transform of the autocorrelation sequence h ( n ) .
5.4.5 Time Delay
The time delay of a filter is a measure of the average delay of the filter as a function
of frequency. It is defined as the negative first derivative of a filter’s phase
response. If the complex frequency of a filter is H (e / 0 ) ) f then the group delay is
T (CO) = - d<t> (co)
dco
(5.22)
where O (co) is the phase angle of H (e ■/t0).

Example 5.24 Determine the impulse response h (n) for the system
described by the second order difference equation.
y (n) - 3y (n - 1) - 4y (n - 2) = x (n) + 2x (n - 1)
Solution
For the impulse response, x (n) = 5 (n) and hence y (n) = h (n) Therefore, the
given difference equation becomes
h {n) - 3 h (n -1) - 4 h (jn - 2) ■ 5 (») + 2 8 in - D Taking z-
transform and rearranging, we get
H (z) [1 - 3Z”1 - 4z“2] = [1 + 2Z’1]

Therefore, H(z)
= 1 + 2 z-1 1- 3z-1 - 4z~ 2
z(z + 2) z (z + 2) Z 2 - 3 Z -
4 (Z+1)(Z-4)
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Discrete and Fast Fourier Transforms 283

Distributive Law
x(n ) * [hx(n) + h 2 (n )] = x(n ) * h x (n ) + x (n ) * h 2 (n )
The distributive law states that if there are two LTI systems with impulse
responses h x (r i) and h 2 (n) and are excited by the same input x (n), then the
equivalent system has the impulse response as
h (n ) = h x (n ) + h 2 (n )
In general, if there are N number of LTI systems with impulse responses h x (n ) y
h (n),...,hN(n) excited by the same input x(n ), then the equivalent system impulse
2

response is given by
N
h (n ) =
/=1

This is nothing but the parallel interconnection of individual systems.


6.2.4 Circular Convolution (Periodic Convolution)
Consider the two sequences x^/i) and x 2 (n) which are of finite duration. Let X x {k )
and X 2 { k ) be the AT-point DFTs of the two sequences respectively and they are
given by

X x {k) = X *1 (n)e- j 2 * n k , N , k = 0, l ,.... N- 1.


n-0
N-l
X2()fe)= (,n)e~ j 2 * n k ' N , k = 0, 1,.... N- 1. (6.3)
n-0
Let x 3 (n ) be another sequence of length N and its AApoint DFT be X 3 (k )
which is a product of X x (k) and X 2 (k ), i.e.
X 3 (k) = X 1 (k)X 2 (k ) 9 k = 0, 1,..., N- l.
The sequence x 3 (n ) can be obtained by taking the inverse DFT of X 3 (k ), i.e.
x 3 (m ) = ID FT [X 3 (k)]

= -^j! X 3(k)e J 2 n m k , N
™ k=0
(k) X2 (k) eJ2nmk/N
™ k=0

1 N
-1 N-l N-l
y -j2nnk/N rj2nlk/N j2nmk/N
-I ^xl(n)e
n= 0 /=o
Nto
1 N-l N-l N-l

l W S*2(i) y e j2Kk(m - n - l)/N k

n=0 1=0
= 0

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284 Digital Signal Processing

Consider the term within the brackets. It has the form

for a = 1
for a * 1
where, a = e ■/2/r(/n n ~ l ) , N % When (m -n- l) is a multiple ofN, then a = 1,
otherwise a N = 1 for any value of a * 0.
N, l = m -n + p N j N = (m - n )(mod N), p: integer)
Therefore, ]Ta* =
O, otherwise
x 3 (n i) becomes
N-l
x 3 (m ) = ^ t x l (n )x 2 (m-n , (mod N)), m = 0, 1,..., N - 1. (6.4)
n=0

where x 2 ((m -n ) mod N) is the reflected and circularly shifted version of x 2 (m )


and n represents the number of indices that the sequence x in ) is shifted to the
right.
The above equation has a form similar to the convolution sum and it is called
circular convolution. The circularly shifted versions of x im) are given below:
xim ) = [x(0), x (lU x iN -3 ), x (N -2\ x (N -1)]
x (m -1, (mod AO) = [x (N- 1), *<0), *(1)... x(N-3 \ xiN -2)]
x im -2, (mod AO) = U iN -2\ x iN- l\ *(0), *(!)... x iN-3 )]

x im -n , (mod AO) = [*(AT-n), x iN-n + 1),..., x(AO-rc-l)] x im - N, (mod


AO) = (x(0), X (1 ). . J C (N-3 ), x (N -2\ x iN-1 ))
It is noted that a shift of n = N results in the original signal x (m ).
6.2.5 Linear Convolution vs Circular
Convolution
Consider a finite duration sequence x (n ) of length N x which is given as input to
an FIR system with impulse response h in ) of length N 2 y then the output is given
by
yin ) = x in ) * h in )
*i-i •
= ^x (k )h (n - k \ n = 0,1, ... 9 N 1 + N 2 - 1
k = 0

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Discrete and Fast Fourier Transforms 285

N2 - I

or y(n )= ^h (k )x (n - k ), n = 0, l f . .., N 1 + N 2 - l
k=0

where, x(n ) = 0, n < 0 and n ≥ N x h (n ) =


0, n < 0 and n ≥ N 2 In the frequency-domain,
Y(co) = H ((o ) X(co)
If the sequence y(n ) has to be represented uniquely in the frequency-domain by
samples of its spectrum Y( w), then
Y (k) = Y ( ( 0 ) \ O u 2 K k / N t k = 0 , 1,...,AT-1 =
X(coW(co)|<0 = 2„A/w, k = 0,1,N- l
Y (k) = X lk ) H (k ) (6.5)
where X (k) and H (k ) are the AT-point DFTs of the sequence x (n ) and h (n )
respectively.
Since y(n ) can be obtained by
y(n) = IDFT|X(A)ff(*)}, (6.6)
the AT-point circular convolution of x (n ) with h (n ) must be equivalent to the
linear convolution of x(n ) with h (n ).
In general, the linear convolution of two sequences x (n ) and h (n ) with lengths
N l and N 2 respectively will give a sequence with a length of N ≥ N x + N 2 - 1.
When x(n ) and h (n ) have a duration less than N, for implementing linear
convolution using circle convolution, N 2 -1 and N x - 1 zeros are padded (added) at
the end of x (n ) and h (n ) respectively to increase their
length to N.
E xamp le 6. 1
Compute (a) linear and (b) circular periodic
convolutions of the two sequences x x (n ) = { 1, 1, 2, 2 ) and x 2 (n ) = 11, 2, 3,
4). (c) Also find circular convolution using the DFT and IDFT. So lu tion (a)
Using matrix representation given as follows, the linear convolution of the two
sequences can be determined.
1 1 2 2
*2(nr\

I
y/ i / 2 / 2 /
2 2/2 / 4/ y
3 3/ 3/ 6/ 6/

4
y/ 4 / 8/ y
Hence, x3(n) = x x (n ) * x 2 (n ) = 11,
3, 7, 13, 14, 14, 8)

Bahan dengan hak cipta


286 Digital Signal Processing

(b) Similarly, circular convolution of the two sequences is shown in Fig. E6.1.
N-1
x3(m) = ^x l (n)x 2 (m - n> (mod AO), m = 0,1, ...» N — 1.
n=0

For m = 0
3
*3(0)= (mod 4))
/1 = 0

x 2 (-n , (mod 4)) is the sequence 22(n) folded. The folded sequence is obtained by
plotting x 2 (n ) in a clockwise direction.
x2(0, (mod 4)) = *2(0)
*2 (-It (mod 4)) = X 2 ( 3 )
*2(-2, (mod 4)) = x 2 (2 )
*2(-3, (mod 4)) = x 2 (l)
s3(0) is obtained by computing the product sequence, i.e. multiplying the
sequences andx2(-n, (mod 4)), point by point and taking the sum, we get *3(0) =
15.
For m = 1:
3
x3(l)= ( n ) x 2 ( l - n y (mod 4)) n - 0
x 2 (l - n, (mod 4)) is the sequence x2(- n , (mod 4)) rotated counter clockwise by
one unit in time. From the product sequence, the sum is x3(l) = 17.
For m = 2:
3
x3(2)= ^x x (n ) X2(2- n t (mod 4))
n = 0

X2(2 - n> (mod 4)) is the sequence x 2 (- n , (mod 4)) rotated counter clockwise by
two units in time. From the product sequence, the sum isx3(2) = 15.
For rn = 3:

23(3) = ^ x x (ti)x 2 (3— /i, (mod 4))


n=0
x2(3 - n> (mod 4)) is the sequence x2(- n, (mod 4)) rotated counter clockwise by
three units in time. From the product sequence, the sum isx3(3) = 13.
Hence, the circular convolution of the two sequences x x (n ) and x 2(n) is
x 3 (n ) = (15, 17, 15, 131

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Discrete and Fast Fourier Transforms 289

Here, N = 4 . Therefore,
3
X 2 (k) = 5>2 (») c--'2’1'1*74 , A = 0, 1, 2, 3.
/1 = 0
For k = 0

X2 (0) = X*2 (») e- j 2 n n ( 0 V 4 = 10 /1 = 0

For A = 1

X2(l)= 'Zx 2 (n)e- J m a /1 0 =

= 1 + 2e~ J * 1 2 + 3e~ j n + 4e ~ j 3 K l 2 = 1 + 2 (-j) + 3


(-1) + 4 (j) = - 2 + j2
For* = 2

X 2 (2) = f ^(n) e-^n /1 = 0


= 1 + 2c'7’' + 3e"-,K + 4 c--'3”
= l + 2(-l) + 3(l) + 4(-l) = -2
For A = 3

X2 (3) = J j x 2 (,n )e - j i 3 K l 2 ) n /1 = 0
= 1 + 2 e- j 3 *' 2 + 3e ~ J 3 n + 4e--'9n/2 = 1 + 2 O) + 3 (-
1) + 4 (- j) = - 2 —2 j X 2 (k )= {10, — 2 + 2 j , - 2, - 2 - 2j}
X 3 (k) = X ^k) X 2 (k )
= (60, (-1 + j) (-2 + 2(/), 0, (-1 -j) (-2 - 2>J}
= { 60, -4 j, 0, 4j )
We know that x 3 (n ) = IDFT (X3(/fe)}
13
Hence, x 3 (n ) = - £X3 (k) e j 2 K n k u , n = 0, 1, 2, 3
4
A=0

= | [60 + (-47) e-'*n/2 + (4j) e i 3 K n ' 2 ]

*3(0) = -j- [60 + (—4«/) + (4j)]= 15 4


*3(1) = \ [60 - 4 j e j K ' 2 + 4j e J M 2 ] = A [60 - 4 j (+ j) + 4j (-j)]

Bahan dengan hak cipta


290 Digital Signal Processing

= - [60 + 4 + 4] = 17 4

X3(2) = i [60 - 4J e j K +4 j e J3K]

= 1[60-4J(-1) + 4J (-1)] = 15
4
9
[60 + (-4j) e m n
X3(3) = | + 4jV *'2]

= 1[60 + (-4J)(-J) + 4JO)]


4
= -[60-4-4] = 13 4
Therefore, x3(n) = [ 15, 17, 15, 13 ]
No te: From the above results, we find that the resulting sequences obtained by
both linear convolution and circular convolution have different values and
length. Linear convolution results in an aperiodic sequence with a length of
(2N- 1), i.e. seven in this case, whereas circular convolution results in a
periodic sequence with a length of N, i.e. four in this case. Circular convolution
will produce the same sequence values as those produced by linear convolution
if three zeros are padded at the end of the two given sequences x^njand x 2 (n ).

E xamp le 6. 2 Find the response of an FIR filter with impulse


response h (n ) = {1, 2, 4| to the input sequence x (n ) = {1, 2).
So lu tion
Linear Convolution
Given h (n ) = { 1, 2, 4 ) and x(n) = I 1, 2 }
Here N l = 3 and N 2 = 2. Hence N = N x + N 2 - 1 = 4

We know that y(n ) = x (n ) * h (n ) = ' ^ i x (k)h (n - k )


k=-°°
Therefore,

y(0) =
k =-oo
= ... + x(0) M0) + x(l) h (- l) + ... = 0
+ 1 + 0... = !

;y(D= k)
k = ~oo

Bahan dengan hak ipta


Discrete and Fast Fourier Transforms 291

= ... + MO) A(l) + Ml) MO) + M2) M-l) + ...


= ... + (1X2) + (2) (1) + 0 = 4
OG

y(, 2)= Xx (k)h (2 -k )


k=—

= ... + x(0) M2) + Ml) Ml) + M2) M0) + ...


= ... 0 + (1X4) + (2) (2) + 0 + ... = 8

y(3) = X*(MM3 -M
k=~oo

= ... + x(0) M3) + *(1) /1(2) + x(2) Ml) + ...


= ... 0 + (2)(4) + 0 ... = 8 y(n)={ 1,4, 8,8}

Alternate Method
Using matrix representation given below, the linear convolution of the given
two sequences can be determined.

Hence, y(n ) = 11,4,8,8}


To find linear convolution through circular convolution with padding of
zeros:
Given h (n ) = { 1, 2, 4 } and x (n ) = | 1, 2 }
Therefore, N x = 3 , N 2 = 2, N = N x + AT2-1 = 4
Appending zeros to the above sequences, i.e.(N2- 1) and (Nj - 1) zeros are added
at the end of h (n ) and x (n) respectively. Therefore, h (n ) = 1 1,2, 4,0 land
x (n ) = 1 1, 2, 0, 0 }.
N-l
We know that y(n ) = ^x(n) h(m - n> (mod(N)), m = 0, 1, ..., N-l
n=0

Therefore, from Fig. E 6.2, we have


3
y(0 ) = ' £x (n)h (, -n, ( m o d ( 4 ) ) = l
n=0

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304 Digital Signal Processing

= 0 + (1) (4) + (1) (-3) + (0) (-2) + (1) (1) + (1) (0)
+ 0 .. . = 2
When n = -3

y ( - 3)= (k)x(-3 - k)
*=-~

= . .. + x(-2) h(-1) + x(-l) h(-2) + a:(0) h(-3) +


x(l) A(-4) + x(2) A(-5) + . . . =
0 + (1) (- 3) + (1) (-2) + (0) (1) + (1) ( 0 ) + (1) (0)
+ 0...O-5
When n =-4

y(-4) = ^ h ( k ) x ( - 4 - k )
*=—

= . .. + x(-2) A(-2) + x(-l) ft(-3) + x(0) A(-4) +


x(l) A(-5) + x(2) A(-6) + .. . = 0
+ (1) (-2) + (1) (1) + (0) (0) + (1) (0) + (1) (0)
+ 0 + . . . — —1
When n = -5

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(0)0 (0)*

Fig. 6.8 The Flow-Graph of the Decimation-in-time FFT Algorithm for


Discrete and Fast Fourier Transforms 325

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360 Digital Signal Processing

The decimation-in-time FFT flow diagram for N = 12 is shown in Fig.


E6.30.
XAO)
Discrete and Fast Fourier Transforms 361
= X l (k ) + W k l 6 X 2 (k ) + W^X s ik ) + W ftX J k )
1 . where, X l (k ) = %x(4n)W%*
n=0

= *(0) + *(W^ +x(8)W®* +x(12)W}f

X 2 (k) = fx(4n + l)Wi"k n = 0

= x(l) + X(5)W4J + X ( 9)W » * + *(13)W! l k X 3 (k )= £x(4n +

2)W#*
n=0

= x(2) + x(6)W^ +x(10)W«* + x(14)Wjf X 4 (,k )= £x(4n +

3)Wt6"*
n=0

= x(3 ) + x (7 )W % +*(ll)Wf| + x(15)W}g*


Also, X k (k + 4) = X i(k )
Therefore ,
X (0 ) = X x (0 ) + X 2 (0 ) + X3(0) + X 4 (0)
xa)=x1a) + w J x (i) + w%xz( i) + wa16x4a)
6 2

X (2 ) = Xj(2) + Wf6X2(2) + W?fiX3(2) + W %X 4 ( 2) X(3) =


Xx(3) + W?6X2(3) + W?6X3(3) + W?6X4(3) X(4) = X,(0) +
W*6X2(0) + W^6X3(0) + W gX4(0) X(5 ) = X1(1) +
W?6X2(1) + Wj°X3(l) + Wj®X4(l) X(6) = X,(2) + W %X 2 (2 )
+ W gX3(2) + W }jpf4(2) X(7 ) = X y (3 ) + W\ 6X/3) + W
JgX3(3) + W^X4(3) X(8) =Xt(0) + W*6X2(0) + W[®X3(0) +
W ? G X 4 ( 0) X(9)=AT1(1) + W»6X2(1) + W}|.Y3(1) + W^X4(1)
X(10) = Xj(2) + W }°X2(2) + W*“X3(2) + W™X4(2)
Z(11)=X1(3) + Wj^O) + W“X3(S) + WjgX4(S) X(12) =X,(0)
+ Wj|X2(0) + W^X3(0) + W™X 4 ( 0) X(13) = X1(1) +
Wj|X2(l) + W*X 3 ( 1) + W jgX4( 1) ATC14) = Xj(2) + W
\iX 2 (2 ) + W f 6 X 3 ( 2) + W«jr4(2)

Bahan dengan hak cipta


X(15) = Xj(3) + W^X2(3) + W^X3(3) + W?*X4(3)
Figure E6.31 shows the radix-4 decimation-in-time FFT flow diagram for N = 16.
362 Digital Signal Processing

To determine the DFT of the given 16 - point sequence x ( n ) = 10, 1, 0, 1,


0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1)
x.ik) = x(0) + *(4)Wt* + x(8 )W % + *(12)Wg*
=0+0+0+0=0
X 2 (k) = x (l) + x (5 )W 4 1 % + x(9)W®£ +x(13)W}§*
= 1 + U - j) + 1(-1) + l(-j) = -2j
X 3 (k) = x ( 2 ) + X (6)W 4 I + *(10)Wf* + *(14)Wg* = 0
Xj(0) = Xx(l) = X x ( 2 ) = Xj(3) = 0 X2(0) =1 + 1 + 1 + 1 = 4

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408 Digital Signal Processing

h d i7) = 0.041210357 ail) = 0.251848936 hi7) = 0.010378785


h d i8) = 0.018991198 a(8) = 0.139850453 A(8) = 0.002655928

7.4.6 FIR Half-band Digital Filters


A low-pass filter with the following frequency response has a desired filter response
as
for 0 < f < f p shown
in
forf-W-f Fig.7.
5.

(7.71)

Such filters, exhibiting symmetry as shown in Fig. 7.5 are known as half-band
filters. These filters have an odd numbered filter length and are used in applications
like decimation or interpolation of the sampling rate by a factor of two. It can be
seen from the Fig. 7.5 that the frequency response has an odd symmetry around
FI4. This results in zero even filter coefficients. Consequently, the realisation of a
half-band filter requires only half as many multiplies as general FIR filters.
The general properties of half-band FIR filters are listed below.
1. Half-band FIR filters have ripples of equal amplitude in both the passband and
the stopband.
Sp = 5 s = 8
2. The stopband frequency is
f s = 0.5F-f p
3. The transition bandwidth of the filter is then given by
AF = 0.5 F - 2f p

Fig. 7.5 Frequency Response of Half-Rand Filter


Finite Impulse Response (FIR) Filters 409

4. The impulse-response at n = 0 is 0.5, i.e.


410 Digital Signal Processing

5. The frequency response is antisymmetric around the center of the band, i.e.
at f= 0.25F.
\H(0.25F + f)\ = l-\H(0.25F-f)\
At/*= 0.25 F t

\me j u > T )\ = \H{e j * n )\ = |


A

7.5 DESIGN OF OPTIMAL LINEAR PHASE FIR FILTERS


In both frequency sampling and windowing methods of designing FIR filters, there
was a problem with the precise control of the critical frequencies. In the optimal
design method, to be discussed in this section, it is considered as a Chebyshev
approximation problem. It is viewed that the weighted approximation error
between the actual frequency response and the desired filter response is spread
across the passband and the stopband and the maximum error is minimised. This
design method results in passband and the stopband having ripples. The design
procedure is explained using a low-pass filter with passband and stopband edge
frequencies cop and cos, respectively. From Fig. 7.6, the frequency response of the
filter in the passband is
1- 8p < | H(e j f a ) | < 1 + 8^ | (01 < (Op (7.72)
The frequency response in the stopband is
- 8S< |//(e^)|<8s, |co|>cos (7.73)
The term 8p represents the passband ripple and 8, is the maximum attenuation in
the stopband.

Fig. 7.6 Frequency Response Characteristics of


Physically Realisable Filters
There are four different cases that result in a linear phase FIR filter, viz., (i)
symmetric unit impulse response and the length of the filter, M odd, (ii) symmetric
unit impulse response and M even, (iii) antisymmetric unit impulse response and M
odd, and (iv) antisymmetric unit impulse response and M even. The first case is
discussed below in detail and other cases are listed in Table 7.2.
In the symmetric unit impulse response case, h(n) = H(M -1-n). The real-
valued frequency response characteristics \ H (e j { * )\ = \H r (e j { * ) | , given in Eq.
7.14, is
(Af - 3)

+
\H (# J " )\ = 2 XA(ra)coscof^^-ra) (7.74)
v 2 y n=0 v 2 )
Let k = (Af - l)/2 - n. Then Eq. 7.74 can be written as
(Af-1)
2
jm
\H (e )\ = J^ci(k) cos oik (7.75)
A=0
where

a(0) = h (^2^) (7.76)

a ( k ) = 2 h [ ^ ^ - k ) for
The magnitude response for the other cases are similarly converted to a compact
form as given in Table 7.2.

Table 7.2 Magnitude Response Functions for Linear Phase FIR Filters
Filter Type Q(co) P((0)
(Af -1)/2
Case (i) - Symmetric and M odd
1 £a(k) cos c o k * = 0
h(n)= h(M-l-n)

Case (ii) - Symmetric and M even (0 (M/2)- 1


cos — £b(k) cos (x)k
/t(n) = MM-l-n) 2 k = 0
Case (iii) - Antisymmetric and Af odd sin (0 (Af - 3)/2

h ( n ) = - h ( M - 1-n) c(£) cos coA


*=0

Case (iv) - Antisymmetric and M even CO


(M/2)-l
sm —
h ( n ) = - h ( M -1-n) 2 ^ d ( k ) cos co k k = 0

From Table 7.2, it can be seen that the magnitude response function can be written
as given in Eq. 7.77, for the four different cases.
\H(e J < 0 )\ = <?(co)P(co) (7.77)

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430 Digital Signal Processing
T = — tan ^ = - tan - = 0.276 s Qc 2 3 8
CO

Frequency Compressed

T
I

Frequency
expanded

A
l«(«)

Fig. 8.5 Relationship Between co and Q. as

Given in Eq. 8.40 The sampling period is

obtained from the above equation using


Using bilinear transformation,
H(z)=H(s)
S
~T (z-fl)

2 ( z - 1) T
+ 0.1
(z + l)
H(z)
= [f 2 (z - 1)
+ 0.1 +9
T (z + l)

(2I T ) ( z - 1) ( z + 1) + 0-1(2 + l)2 " [


(2I T ) (2 - 1) + 0.1 (2 + l)]2 + 9 (2 + l)2 Substituting T = 0.276 s,
1 + 0.027 z ' 1 - 0.973 2-2

H(,z) =
8.572 - 1184 2_1 + 8.177 z-2

Example 8.8 Apply bilinear transformation to 2

H(s)
= (s + 1) (s + 3)
with T = 0.1s.

Bahan dengan hak cipia


Infinite Impulse Response (IIR) Filters 433
N , the Butterworth response approximates the ideal response. However, the phase
response of the Butterworth filter becomes more non-linear with increasing N .
The design parameters of the Butterworth filter are obtained by considering the
low-pass filter with the desired specifications as given below.
8,< H ( e J l 0 ) \ ≤ 1, 0 < co< coj (8.42a)

H ( e j 0 i ) | < 82, co2 < o) < 7i (8.42b)


The corresponding analog magnitude response is to be obtained in the design
process. Using Eq. 8.41 in Eq. 8.42 and if A = 1, we get

5l 2 1
(8.43a)
l + fQ./Q,.) "
(8.43b)
1 + (Q2/Qc)2" 82

Equation 8.43 can be written in the form


«VQc)2"<i-l (8.44a)

(Q2/Qc)2JV>^-1
Oo (8.44b)

Equality is assumed in Eq. 8.44 in order to obtain the filter order N and the 3 dB
cut-off frequency Qc. Dividing Eq. 8.44b by Eq. 8.44a
(Q2/Q,)2;v = (1/8! - l)/(l/8? - 1) (8.45)
From Eq. 8.45, the order of the filter N is given by

*r_ 1 l°g {[(1/S| - l)/(l/8? - 1)1}


(8.46)
~ 2 log ova,)
The value of N is chosen to be next nearest integer to the value of N as given by
Eq. 8.46. Using Eq. 8.44a, we get
£
Qc = 1 1/2AT
(8.47)
[(1/8?)-l]
The values of and - £ 1 % are obtained using the bilinear transformation or
impulse invariant transformation techniques, Q =

— tan for bilinear transformation or = ~ for the impulse


invariant transformation. •
The transfer function of the Butterworth filter is usually written in the factored
form as given below.
N /2
Bh&c
N = 2,4, 6,... (8.48)
n
H(S) = A = i S +bkncs + ckfrc

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Chapter

Realisation of Digital Linear


Systems

9.1 INTRODUCTION
For the design of digital filters, the system function H ( z ) or the impulse response
h ( n ) must be specified. Then the digital filter structure can be implemented or
synthesised in hardware/software form by its difference equation obtained directly
from H ( z ) or h ( n ) . Each difference equation or computational algorithm can be
implemented by using a digital computer or special purpose digital hardware or
special programmable integrated circuit.
In order to implement the specified difference equation of the system, the
required basic operations are addition, delay and multiplication by a constant. A
chosen structure determines a computational algorithm. Generally, different
structures give different results. The most common methods for realising digital
linear systems are direct, cascade and parallel forms, and state variable realisations.
In this chapter, direct-forms, cascade and parallel forms, ladder form and state
variable realisations of HR filters, direct-form and cascade realisations of FIR
filters, and realisations of linear phase FIR systems are discussed.

9.2 BASIC REALISATION BLOCK DIAGRAM AND THE SIGNAL-FLOW


GRAPH
The output of a finite order linear time invariant system at time n can be expressed
as a linear combination of the inputs and outputs ,
N M
y(n)= ^aky(n-k)+^bkx(n-k) (9.1)
k=\ k=o
where a k and b k are constants with a 0 * ■ 0 and M < N .

Bahan dengan hak cipta


454 Digital Signal Processing

The current output y i n ) is equal to the sum of past outputs, from y i n - 1) to


y i n - N ) , which are scaled by the delay-dependent feedback coefficient a k, plus
the sum of future, present and past inputs, which a^e scaled by the delay-
dependent feed forward coefficient bk.
Taking the z-transform of the output sequence y i n ) given in Eq. 9.1, we get

Y(z)=
M

= 1 Xa h y ( n - k ) + ^bkx(n- k) -n

\k=l k=0
n = - »o

Changing the order of summation M


N r- -n
+
£■x ( n - k ) z - n
Y ( z ) = £aJ ' £ y ( n - k ) z k=0 n = -oo

k = l [n = -«»

= ia** -*y( 2 )+ ^ b ^ X i z )
AT k = l = f ^ b z - k Xk{=z )0
k
Y ( z ) \ 1 - £ a* 2"* k=0
*=i

Therefore, M

k=0 =
Y
H(z) N (9.2)
= (z) i- -k
X(z) k=i

f>(n)aT"
N
The computational algorithm of an LTI digital filter can be conveniently
represented as a block diagram using basic building blocks representing the unit
delay i z -1) or storage element, the multiplier and the adder. These basic building
a
(b) Multiplier

2 *1 (n) + x2ir)

(a) Addition of two sequences x{r) — ax(n)

blocks are shown in Fig. 9.1.

x{n) x{n-
(c) Unit delay
1)
Fig. 9.1 Basic Building Blocks

Bahan dengan hak cipta


Realisation of Digital Linear Systems 455
For example, the difference equation of the first-order digital system may be
written as
y ( n ) = a l y ( n - l ) + x ( n ) + b x x ( n - 1)
The basic realisation block diagram for this equation and the corresponding
structure of the signal flow graph are shown in Figs 9.2 (a) and (b). Here, it is clear
that there is direct correspondence between branches in the digital realisation
structure and branches in the signal flow graph. But in the signal flow graph, the
nodes represent both branch points and adders in the digital realisation block
diagram.
Sink node 1
Source node
1

Fig. 9.2 (a) Basic Realisation Block Diagram Representing a First-order


Digital System and
(b) Its Corresponding Signal Flow Graph.

Advantages of representing the digital system in block diagram form


(i) Just by inspection, the computation algorithm can be easily written
(ii) The hardware requirements can be easily determined
(iii) A variety of equivalent block diagram representations can be easily
developed from the transfer function
(iv) The relationship between the output and the input can be determined.
9.2.1 Canonic and Non-Canonic Structures
If the number of delays in the realisation block diagram is equal to the order of the
difference equation or the order of the transfer function of a digital filter, then the
realisation structure is called canonic. Otherwise, it is a non-canonic structure.

9.3 BASIC STRUCTURES FOR HR SYSTEMS


Causal HR systems are characterised by the constant coefficient difference equation
of Eq. 9.1 or equivalently, by the real rational transfer function of Eq. 9.2. From
these equations, it can be seen that the realisation of infinite duration impulse
response (HR) systems involves a recursive computational algorithm. In this
section, the most important filter structures namely direct Forms I and II, cascade
and parallel realisations for HR systems are discussed.

Bahan dengan hak cipta


456 Digital Signal Processing

9.3.1 Direct Form Realisation of HR Systems


Equation 9.2 is the standard form of the system transfer function. By inspection of
this equation, the block diagram representation can be drawn directly for the direct
form realisation. The multipliers in the feed forward paths are the numerator
coefficients and the multipliers in the feedback paths are the negatives of the
denominator coefficients. Since the multiplier coefficients in the structures are
exactly the coefficients of the transfer function, they are called direct form
structures.
Direct Form I
The digital system structure determined directly from either Eq. 9.1 or Eq. 9.2 is
called the direct form I. In this case, the system function is divided into two parts
connected in cascade, the first part containing only the zeros, followed by the part
containing only the poles. An intermediate sequence w ( n ) is introduced. A
possible HR system direct form I realisation is shown in Fig. 9.3, in which w ( n )

H(z)

W(z)
Ci w

n Hito w{n) Hz(z)


Zeros Poles

represents the output of the first part and input to the second.

moor Y(z
x{n)
)
or
y{n
Fig. 9.3 A Possible HR System direct-form I )
Realisation
w ( n ) = £b k x ( n - k ) • A*o
and
N
y ( n ) = £a*y(n- k ) + w { n )
k=i
Taking 2-transform, we get
M
W(z) = X(z)
k=0
and
N
Bahan dengan hak cipta
Realisation of Digital Linear Systems 457
Y(z) = Y(z) Xa k z ~ k + W ( z )
k=l

Bahan dengan hak cipta


458 Digital Signal Processing

Therefore,
W
Y(z)
( zN )
-k
1 -

A=1
x(n)o y(n
)
1 r!
r1
y{
n
f1

x [ n - 2) y(n-2)

x{n-N+ y{n-N+
1) 1)

x(n- N) y(n-N)

Fig. 9.4 Direct-Form I Realisation of an Nth Order Difference

Example 9.1
The direct form I realisation is shown in Fig. 9.4. The direct form I realisation
requires M + N storage elements. For our convenience, we have assumed that M
= N , for drawing the network.
Develop a direct form I realisation of the difference
equation
y ( n ) = b 0 x ( n ) + b x x ( n - 1) + b 2 x ( n - 2) + b 3 x ( n - 3)
- a x y ( n - 1) - a 2 y ( n - 2 ) - a 3 y ( n - 3)
Solution Taking 2-transform of the given difference equation and simplifying,
we get the transfer function
Y(z)
H(Z) = = b° +6i2-1 +62Z~2 +b3z~3
X(z) 1 + cilz~1 + a2 z~2 + a3 z'3
The corresponding direct form I realisation structure is illustrated in Fig. E9.1.

Bahan dengan hak cipta


\

Fig. E
9.1
Realisation of Digital Linear Systems 459

Direct M
Form and H 2 ( Z ) = X6*2"*
k=0
II
Since

Fig. 9.5 The Decomposition for Direct Form II Realisation


we are dealing with linear systems, the order of these parts can be interchanged.
This property yields a second direct form realisation. In direct Form II, the poles of
H ( z ) are realised first and the zeros second. Here, the transfer function H ( z ) is
broken into a product of two transfer functions//!( z ) and//2(2), where//!( z )has only
poles and H 2 ( z )contains only the zeros as given below:
mz) = H (z). H 2 ( Z )
l

where
The decomposition for direct Form II realisation is shown in Fig. 9.5. An
intermediate sequence u ( n ) is introduced to obtain the output of the filter.
N
u(n) = ^aku(n - k) + x(n)
h = l
M
and y(n)=
k-0

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Fig. E9.16

Realisation of Digital Linear Systems 489

REVIEW QUESTIONS
9.1 How will you obtain the transfer function
M

Y(z) k = 0
H(z) =
X(z) N
-k
k=l
from the linear difference equation
N M
y(n) = Z a k y ( n - k ) + - £)?
k=i k=o
9.2 Distinguish between the methods of realisation namely, block diagram
representation and signal flow graph for implementing the digital filter
transfer function.
9.3 What are the basic building blocks of realisation structures ?
9.4 What are the advantages of representing digital systems in block diagram
form?
9.5 Define canonic and non-canonic structures.
9.6 Compare direct form I and direct form II realisations of IIR systems.
9.7 What are the drawbacks of direct form realisation of IIR systems?
9.8 Explain any two IIR filter realisation methods.

Bahan dengan hak cipta


490 Digital Signal Processing

9.9 How will you develop a cascade structure with direct form II realisation of a
sixth-order IIR transfer function ?
9.10 How will you develop a parallel structure with direct form II realisation of a
sixth-order IIR transfer function ?
9.11 Draw the transposed direct forms I and II structures of an third- order IIR
filter.
9.12 Discuss in detail direct form and cascade form realisations of FIR systems.
9.13 What is meant by a linear phase filter?
9.14 Why do FIR filters have inherent linear phase characteristics?
9.15 What is the necessary and sufficient condition for the linear phase
characteristics in an FIR filter?
9.16 How will you develop direct form realisations of third and fourth order
functions of linear phase FIR systems.
9.17 Draw a block diagram representation for the following systems
(a) y(n) = ay(n - 1) + a x ( n ) - x(n - 1)
(b) yin) = x(n) + x(n - 1) + x (n - 2)

(c) y(n) = -■ y(n - 1) + x (n) - -j- x(n - 1)


z z
9.18 Draw the block diagram representation of the direct form I and II
realisations of the systems with the following transfer functions.

(a) H(z) = —= ------- K ----- ------


5z3 - 4.5z2 1.4z - 0.8
+

z-1 - 3z'2
(b) H(z) =
(10 - z _ i ) (i 0.5z~3 + 0.5z~2)
+

, ,r,
, , 2 + 3z~2 - 1.5z~4
(c) H(z) = ----------------------- ; ----------------- o-

1 - 0.2z-1 0.35z'2
+

9.19 Develop a canonic direct form realisation of


the transfer function
H(z) = 3 + 5Z~! - 8z~2 +4z~5
2 + 3z-1 + 6z~3
and then determine its transpose configuration.
9.20 Obtain the cascade and parallel realisation structures for the following
signals.
(a) y(n) = 4 y ( n - D - 4 y( - 2) + x(n) + ^ x (n - 1)
n

4 8 3
(b) y(n) = - 0 . 1 y ( n - 1) + 0.72 y(n -2) + 0.7 x(n) -0.252 x(n - 2)

Bahan dengan hak cipta


491 Digital Signal Processing

2 { l - z ~ 1 ) { l + ^2z~1 + z ~ 2 )
(c) H ( z ) =
(1 + 0.5z'1 (J 0.9Z'1 + 0.81z
- ~2))

Bahan dengan hak cipta


1 + -l
492( fDigital
H ( z ) Signal
= Processing
z 1 + 0.06z~2
1 - 0.5z~
9.24 The transfer function of a causal UR filter is given by

H(z) 5z(3z - 2)
= ( z + 0.5) {2 z - 1)
Determine the values of the multiplier coefficients of the realisation
structure shown in Fig. Q9.24.

Ans a = 3 5 1 4 , b = -1/2, c = -514, d = 1/2


9.25 Realise the FIR transfer function H(z) = (1 + OMz-1)5 in (i) two different
direct forms, (ii) cascade of first-order sections, (Hi) cascade of one first-
order and two second-order sections, and
(iv) cascade of one second-order and one third-order sections.
9.26 Realise the following system functions using a minimum number of
multipliers.
(a) H(z) = - z-2) (l - ^z- 1
+ z- 2 )

(b) H(z) = 1 + — Z ' 1 + ~ z z 2


+ z~3
9.27 Obtain the system function H(z) for each of the systems shown in the Fig.
Q9.27.
i+i ,+ !
(b) H(z) = 10z~2
Ans (a) H(z) = 8-
^- r 7

1_2z-i\ 1- 0.83Z-1 - 0.76z~2


(1 + 5Z-1) j1 1 - 0.2Z-1 (1 + 1.5z ' 1 ) (1 + 0.8Z-1)

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508 Digital Signal Processing
10.6 COEFFICIENT QUANTISATION IN DIRECT FORM
REALISATION OF FIR FILTERS
The statistical bounds on the error in the frequency response due to coefficient
quantisation (rounding) is given here. The frequency response of a linear phase FIR
filter is given by

(10.33)
For a linear phase FIR filter, h (n) = h ( M - 1 - n ) . The term e = e - j w ( M - m j n a b o v e
expression represents the delay and is unaffected by quantisation. Hence, the
quantisation effect is solely on the pseudomagnitude term M (co). Let { h q ( n ) } be
the sequence resulting from rounding [/i(/i)| to a quantisation step size of 2"B.
Therefore,
hq(n)= h(n) + e(n) (10.34)
and h q ( n ) = h q ( M - 1 - n ) for 0 < n ≤ ( M - l)/2. Let e ( n ) be a random
,-B B
sequence and uniformly distributed over the range and .Let
22
H q ( z ) be the z-transform of (h q (n)) and M q (co) be the pseudomagnitude of the
quantised linear phase FIR filter. The error function is defined to be
' (10.35)
E (e Ji0) = Mq(co) -
M(co)
M- 1 \ ( A/- 3J /2 rf TA 1 \ ”]
j
or E(e ")=e J + 2 ^ e ( n ) cos ^— ------- ------njcoj

where E ( e J 0 i ) is the frequency response of a linear phase FIR filter that has
(e(n)} as the impulse response for the first half and the second half can be obtained
using e ( n ) = e ( M - 1 - n ) . Thus, the filter with its coefficients rounded-off can be
considered as a parallel connection of the ideal filter (infinite precision) with a
filter whose frequency response is E ( e j i a ) e ~ j “ ( M ~ iy2. Since the error e ( n )
due to rounding of the filter
2~ B •

coefficients is always lesser than or equal to -------------- , a bound on \ E ( e J l 0 ) \


can be obtained as shown below.
( M - 3)/2
Af-1
\E(ejw)\ ≤ +2 ^\e(n)\ cos (10.36)
n-0
M — 1
Letting ------------ n = k in the second term of the above expression,

( Af -l )/2
- x; M -1 ,
M- 1 e | ----------- k cos k co |
-2 I
k=l Bahan dengan liak cipta
Effects of Finite Word Length in Digital Filters 509

That is,
Af - 1 2
Y I cos (/2 k
| E(eJ")\
CO) |

= \ (10.37)

or,
. o-fl
ISO?-' )| ≤ M ——
10
(10.38)
1
2

A more useful statistical bound than that given by Eq. 10.38 can be obtained if it
is assumed that the errors due to quantisation of different coefficients are
statistically independent. This statistical bound provides the filter designer a means
of predicting how much accuracy is required for the coefficients to obtain a desired
frequency response without apriori knowledge of the value of the filter coefficients.
From Eq. 10.36,
an expression for the power of the error signal, £7 (eyco )2, is obtained.

<4(o» = W“)2 = e (^~^) + ^ 4e(/i)2 cos2 -n


) c0]
(10.39)
o-2 B
2
For uniform distribution, e ( n ) 2 = o {n) = ------ . Therefore
12
Af - 1
-2 B
1 + 4 Y cos2
f©\2 _ 2
(10.40)
12
(©**)
n=1
Defining a weighting
function W (co) as
Af - 1 “

W ( o) = 1 2 i

1+4 £ cos2 (ton)


(10.41) 2 M - 1
n=1
i *

The standard deviation of the error is given by


°B«0) = (10.42)
The value of the weighting function W(co) lies between 0 and 1, and W ( 0) =
W i n ) = 1 for all Af. Thus,

^ (10-43)
The value of E (&***) is obtained as a summation of independent random
variables, for any ( D . E ( e j t 0 ) is a Gaussian function for large

Bahan dengan hak cipta


510 Digital Signal Processing

values of M . Thus, the mean and variance of E i e J l i > ) gives an excellent


description of E ( e J ( 0 ) .

10.7 LIMIT CYCLE OSCILLATIONS


In Section 10.3, it was assumed that the input sequence traverses several
quantisation levels between two successive samples and so the samples of the
round-off noise sequence were uncorrelated with each other and with the input
signal. These assumptions are invalid in cases such as a constant or zero input to a
digital filter. In such cases, the input signal remains constant during successive
samples and does not traverse several quantisation levels. There are two types of
limit cycles, namely, zero input limit cycle and overflow limit cycle. Zero input
limit cycles are usually of lower amplitudes in comparison with overflow limit
cycles. Let us consider a system with the difference equation
y i n ) = 0.8y ( n - 1 ) + x ( n ) (10.44)
with zero input, i.e., x ( n ) = 0 and initial condition y(- 1) = 10.A comparison
between the exact values of y i n ) as given by Eq. 10.44 using unquantised
arithmetic and the rounded values ofy i n ) as obtained from quantised arithmetic are
given in Table 10.2.

Table 10.2 A Comparison of exact y (n) and rounded y (n)


n y (n)-unquantised y (n)-quantised
-1 10.0 10
0 8.0 8
1 6.4 6
2 5.12 5
3 4.096 4
4 3.2768 3
5 2.62144 2
6 2.0972 2
7 1.6772 2

From Table 10.2, it can be observed that for zero input, the unquantised output
y(n) decays exponentially to zero with increasing n . However, the rounded-off
(quantised) output y i n ) gets stuck at a value of two and never decays further.
Thus, the output is finite even when no input is applied. This is referred to as zero
input limit cycle effect. It can also be seen that for any value of the input condition
| y (- 1) | < 2, the output y i n ) =y (- 1), n ≥ 0, when the input is zero. Thus, the
deadband in this case is the interval [-2, 21.

_____ A digital
Example 10.3 system is characterised by the difference
equation
y in) = 0.9 y in - 1) + x in)
with x in) = 0 and initial conditiony(- 1) = 12. Determine the deadband of the
system.

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512 Digital Signal Processing

l ≤ - ^ - ( 0 < b2 < 1) (10.48)


I-I 02I

OVERFLOW LIMIT CYCLES


Limit cycles can also occur due to overflow in digital filters implemented with finite
precision arithmetic. The amplitudes of such overflow oscillations are much more
serious in nature than zero input limit cycle oscillations. Consider a causal, all-pole
second-order HR digital filter implemented using two’s complement arithmetic with
a rounding of the sum of the products by a single quantiser. The difference equation
describing the system is given by
y ( n ) = Q R [- a 1 y ( n - 1 ) - a 2 y ( n - 2 ) + x (n)l
where Q R [.] represents the rounding operation and y (n) is the actual output of the
filter. The filter coefficients are represented by signed 4-bit fractions. Let a1 = 1A 0 0
1 = - 0.8751d and a2 = 0A 111 = 0.875d and the initial conditions be y (— 1) = 0 A 1
1 0 = 0.75 d and y (— 2) = 1A 0 1 0 = - 0.75d. For zero input, i.e. x ( n ) = 0 and for
n > 0, we get the values for y (n ) as shown below.

Table 10.3
n y (n-1) y (n - 2) -a, y (n - 1)- a 2 y (n-2) y (n) = Q R [.] y(n)
m decimal
y (-1) = 0A110 y (-2) = iAoio - 0.625
0 1A010100 1A011
1 y <0)= lA0ll y (-1) = 0A110 10A110011 0A110 + 0.75

2 y (1) = 0A110 y (0)=1A011 1A001101 1A010 -0.75


3 y (2)= 1A010 y (1) = 0A110 10A101100 0A110 + 0.75
4 y (3)= 0A110 y ( 2 ) = 1A010 1A010100 1*011 - 0.625

For n = 1, the sum of two products has resulted in a carry bit to the left of the
sign bit that is automatically lost, resulting in a positive number. The same thing
happens for n = 3 and also for other values of n . It can be noted from the above
table that the output swings between positive and negative values and the swing of
oscillations is also large. Such limit cycles are referred to as overflow limit cycle
oscillations.
The study of limit cycles is important for two reasons. In a communication
environment, when no signal is transmitted, limit cycles can occur which are
extremely undesirable. For example, in a telephone no one would like to hear
unwanted noise when no signal is put in from the other end. Consequently, when
digital filters are used in telephone exchanges, care must be taken regarding this
problem. The second reason for studying limit cycles is that this effect can be
effectively used in digital waveform generators. By producing desirable limit cycles
in a reliable manner, these limit cycles can be used as a source in digital signal
processing.

Bahan dengan hak cipta


Effects of Finite Word Length in Digital Filters 513
10.8 PRODUCT QUANTISATION
When two B-bit numbers are multiplied the product is 2B-bits long. This product
must be rounded to B-bits in all digital processing applications. The output of a
finite word length multiplier can be expressed as
Q [a- x (n)] = oti x (n) + e (n) (10.49)
where a* x (n ) is the product which is 2B-bits long and e ( n ) is the error resulting
from rounding the product to B-bits. The fixed-point, finite word length multiplier
can be modelled as shown in Fig. 10.7.
The analysis of product round-off noise is similar to A/D conversion noise. The

Fig. 10.7 Product Quantisation Noise


Model
product quantisation noise can be considered as a random process with uniform
probability density function and zero mean. The average power (variance) is given
by

(10.50)

The effects of rounding due to multiplication in cascaded HR sections are discussed


here. Let h (n ) be the system response and e Q (n ) be the response of the system to
the input error e (n ). Then, as discussed in Section 10.4, the output noise power is
given by

(ra)
a\o = oe X
2 h
* (10.51)
/1 = 0

Using Parseval’s relation (see Example 9.1), we get


2
a \ 0 = <f H (z) H (z"1) z”1 dz (10.52)
2Kj J
c

Equation 10.51 or 10.52 can be used in determining the output noise power
depending on whether h ( n ) or H ( z ) is given. Consider a cascaded second-order
HR structure shown in Fig. 10.8.
As shown in Fig. 10.8b, and error signal e ^ n ) is added at the sum node for each
multiplier. For such a case, the output noise power is given by
oo

°2ek = °2e X h2
>
(fl)
<10-53)
/1 = 0

Bahan dengan hak cipta


Effects of Finite Word Length in Digital Filters 515
Draw the product quantisation noise model of the system and determine the overall
output noise power.
Solution The product quantisation noise model is shown in Fig. E10.2

Fig. E 10.2 Figure for Example 10.5


The transfer function of the system is H ( z )
= H x ( z ) H2 ( Z )
The noise transfer function seen by e x ( n ) is H ( z ) and by e 2 ( n ) is H 2 ( z ) . The
overall output noise power is given by
—.2 _a -2 .+-.2 ° err
“ ol °o2
where, from
Eq. 10.52,
e :j H U ) H U _ 1 ) z'1 dz
9 a 2
2
7C j
= ------------------------? ------------------------ d2
1
X x z dz
2 n j i (1 — 0.9 2 ) (1 — 0.8 2 )
_1 1- 1
(1- 0.9 2) (1 - 0.8 z )
2 n j l ( z - 0.9) (2 - 0.8) (1 - 0.9 z ) (1 - 0.8 z )
x
a
ol = ° e
I\ where

/, = —| -------------------------------- ------------------------ d2
1
2 ;t j l (z - 0.9) (2 - 0.8) (1 - 0.9 2) (1 - 0.8 z )
The integral /j can be solved by the method of residues/
/j = sum of the residues at the poles within the unit circle.

Bahan dengan hak cipta


Effects of Finite Word Length in Digital Filters 517
Example 10.6 Repeat Example 10.5 using Eq. 10.51.
Solution Given

H x ( z ) = ------- ---- T and H 2 (z) = ------------ ---- r


1
1 - 0.9 z_1 2
1-0.82-1

From Eq. 10.51,


°lr = °2ol + <*o2
where
oo

o i l = o ' j Z h 2 ( n ) and <? = X h \ (n)


o2

Bahan dengan hak cipta


Effects of Finite Word Length in Digital Filters 518
n=0 n=0

Let us first determine h ( n ) and h2(n).


1
H(z)
= (1 - 0.9 z_1) (1- 0.8 z ~ l ) 1z^ .- '

( z - 0.9) (z - 0.8)

Therefore
H{z 8
) (z - 0.9) (z - 0.8) (z - 0.9) (z - 0.8)
and
8
H(z)
= (1 - 0.9 z-1) (1-0.8 z-1)
Taking inverse z-transform,
h (TI) = [9 (0.9)" - 8 (0.8)"] u (n)
and
h 2 ( n ) = [9 (0.9)" - 8 (0.8)"]2 u ( n )
Similarly,
——r 1 = (0.8)" u (TI) _l-
h 2 (/I) = Z - 1 [ H 2 (Z)1 = z-1 0.8z_1 J

and
h% (TI) = (0.8)2" u (TI)

Therefore,
OO

o^i = o2e X h2 (71) = (T2 X [9 (0.9)" - 8 (0.8)"]2


n=0 n=0

= a \ ^ (81 (0.81)" + 64 (0.64)n - 144 (0.72)n)


n=0

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<%a = & 522 Digital Signal Processing

Find the output noise power from the digital system.


]
Ans: o 2
e0 = <7? '(1-
P?
1-P2

Hj (z) = —7 and H9 ( Z ) = ------------ ,


1 2
1-0.5 z- 1-0.25Z-'
Ans : a2err = a2oI + <rf2; a2oI = 2.90a2; a2o2 = 3.16 a2

H(z) =
10.10 Find the output noise power of the digital filter whose system function is
1
1 - 0.999 z-3
Ans: a2eo = 500.25 a2
10.11 Explain coefficient quantisation.
10.12 Discuss coefficient quantisation in HR filters.
10.13 Discuss coefficient quantisation in FIR filters.
10.14 What is meant by limit cycles in recursive structures?
10.15 What is dead band of a filter ?
10.16 A digital system is characterised by the difference equation y (n) = 0.95 y
(n - 1) + x (n)
Determine the deadband of the system when x (n) = 0 and y ( - 1 ) = 13.
Ans. [- 10, 10]
10.17 Explain product quantisation.
10.18 Two first order low-pass filters whose system functions are given below
are connected in cascade. Determine the overall output noise power.
1 2
(z) =

and Ho (z) =
3 + bjb-2
-i -i
26
- ; \_a-b
l - b l2b
z 2 )(i-bi)(i-b‘ 2 )\
10.19 Determine the variance of the round-off noise at the output of the two
Ans: c?errrealisation
cascade = aii + o^9;of(£;
o2>the=u olfilter
of. with system function
l~bf
H ( z ) = Hj ( z ) ■ H 2 ( Z )

where
1,„,*1

Bahan dengan hak cipta


Chapter

Multirate Digital Signal


Processing

ll.l INTRODUCTION
Multirate digital signal processing is required in digital systems where more than
one sampling rate is required. In digital audio, the different sampling rates used are
32 KHz for broadcasting, 44.1 KHz for compact disc and 48 KHz for audio tape. In
digital video, the sampling rates for composite video signals are 14.3181818 MHz
and 17.734475 MHz for NTSC and PAL respectively. But the sampling rates for
digital component of video signals are 13.5 MHz and 6.75 MHz for luminence and
colour difference signal. Different sampling rates can be obtained using an up
sampler and down sampler. The basic operations in multirate processing to
achieve this are decimation and interpolation. Decimation is for reducing the
sampling rate and interpolation is for increasing the sampling rate. In digital
transmission systems like teletype, facsimile, low bit rate speech where data has to
be handled in different rates, multirate signal processing is used. Multirate signal
processing finds its application in (i) Sub-band coding (speech, image) (ii) Voice
privacy using analog phone lines (iii) Signal compression by subsampling (iv)
A/D, D/A converters, etc.
There are various areas in which multirate signal processing is used. To list a
few,
(i) Communication Systems
(ii) Speech and audio processing systems
(iii) Antenna systems and
(iv) Radar systems
The various advantages of multirate signal processing are
(i) Computational requirements are less
(ii) Storage for filter coefficients are less

Bahan dengan hak cipta


524 Digital Signal Processing

(iii) Finite arithmetic effects are less


(iv) Filter order required in multirate application are low and
(v) Sensitivity to filter coefficient lengths are less
While designing multirate systems, effects of aliasing for decimation and
pseudoimages for interpolators should be avoided.

1 1 . 2 SAMPLING
Let £(/) be a continuous time varying signal. The signal x ( t ) is sampled at regular
interval of time with sampling period T as shown in Fig. 11.1. The sampled signal
x ( n T ) is given by
x ( n T ) = x ( t ) |, =nT = x (n T ), -oc < n < °o (11.1)
x(t)

x(n)

0
Fig. 1 1 . 1 Sampling of a Continuous Time Signal

•u-
A sampling process can also be interpreted as a modulation or multiplication
process, as shown in Fig. 11.2.
The continuous time signal x ( t ) is multiplied by the sampling function s ( t )
which is a series of impulses (periodic impulse train); the resultant signal is a
discrete time signal x ( n ) .
x ( n ) = x ( t ) s(£)|, = , - oo < n < °o
nT (11.2)

Sampling Theorem
According to the sampling theorem, a band-limited signal x ( t ) having finite energy,
which has no frequency components higher than f h hertz, can be completely
reconstructed from its samples taken at the rate of 2f h samples per second.
The sampling rate of 2 f h samples per second is the Nyquist rate and its
reciprocal 1/2 f h is the Nyquist period.

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540 Digital Signal Processing

With this property, the number of multiplications can be reduced by a factor of


two. If N is even,

t'
y(n)= 2^ h ( k ) [ x ( n - k ) + x ( n -(AT- 1 -A))}
k=0

11.5.2 HR Direct Form Structure


Consider an HR filter with the difference equation represented by,
D N-l
y{n)= X a*y(n-A)+ £ b kx(n-k)
(11.25)
k=l k=0
Figure 11.13 shows the signal flow graph for the HR filter. The system equation
of the HR filter is given by
N-l
Ybkz-k
Y(z)
N(z)_ ^
H(z) = (11.26)
xiz)
~ ' D M *=i

r(n) bo
f -------------- ► ---------------- >• ----------- <
x(n --------------- « ■ y(n)
)
r1

*1

r(n-1)

j> .

a
2 1
tfe
W \\

[ > .................... (

r(n- 2)
/•’

d b
N 1 - N- 1

Bahan dengan hak cipta


Multirate Digital Signal Processing 541
Fig. 1 1 . 1 3 The Signal Flow Graph for an HR Filter

Bahan dengan hak cipta


542 Digital Signal Processing

where N(z) and D(z) are numerator and denominator polynomials oiH(z)
respectively. The signal flow graph in Fig. 11.13 assumes D = N - 1. Feed forward
part of the structure represents the numerator and feedback branches represents the
denominator.
I 1.5.3 Structures for FIR Decimators and Interpolators
Figure 11.14 shows the model of an M to 1 decimator. Let x(n) be the input signal
which is sampled with the sampling frequency of F. Let y(m) be the output signal
which is obtained by reducing the original sampling rate by a factorM. The filter
h(n) also operates at the sampling rate F.
Out of every M input samples only one sample will be available at the output
and the rest M - 1 samples will be discarded by the M to 1 sampling rate
compressor. All multiplications and summations are performed at the rate F. By
applying the commutative operations of the network, all multiplications and
additions can be done at the low sampling rate of F/M. Hence the total
commutation rate is reduced by M.
The difference equation is given by,
N-\
y(m) = £h ( n ) x ( M m - n ) (11.27)
n=0

Figure 11.15 shows the model of 1 to L interpolators. The output signal y(m) is
obtained by increasing the original sampling rate by a factor L .
h ( M ) is realised using the transposed direct form FIR structure. By applying
the commutative property, the new structure has L times less computation when
compared with the structure before commutation.

11.6 POLYPHASE DECOMPOSITION


Polyphase decomposition results in reduction of computation complexity in FIR
filter realisation.
The 2-transform of a filter with impulse response h(n) is given by
H(z) = h(0) + z~ 1 /J(1) + 2~2 M2) + ... (11.28)
Rearranging the above equation we get,
H ( z ) = MO) + 2'2 M2) + z ~ 4 M4) + ...
+ z_1 ( h (1)) + 2~2 M3) + z ' 4 M5) + ...) (11.29)

Type I Polyphase Decomposition


2
H(z) = E0 (Z 2
) + 2"1 E x (2 ) (11.30)
where E 0 ( z 2) and E x ( z 2 ) are polyphase components for a factor of two.

Bahan dengan hak cipta


Multirate Digital Signal Processing 543

x(n ) y {m
)

y {m )
= F/M

Fig. 1 1 . 1 4 Direct Form Representation of M to I Decimator

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The
For Flength
filter ofRthe
M ,overall
( z ) , lengthsNGare
==
= 107—
_ 56
-201og
-20
107 =10•Illl
filter
log
calculatedin
14.610 cascade
53,500
V(O.OOl)
V(0.001)
as 72010is-(0.001)
(0.001)
follows:150For-- 13
K 13
10
G(z), 56+ ( 10x 107m ) +o2 ==VL56,000
8 KK- L 5 K
1128
10
R M . F - 56- 14.6
The filter u
10
length in cascade realisation has increased but the number of
562 Digital Signal Processing
multiplications per second (computerised complexity) can be reduced.
11
Total number of multiplications per second is & M , G + RM. F = 1,09,500
Three-stage Realisation
The decimation filter F ( z ) can be realised in the cascade form i?(z) S ( z 5 ) . The
spec S ( z ) , 5p = 0.0005, 5S = 0.001
ifica
tions are given A f = x 5 = 0.285 as follows
For 10,000
570 R ( z ) ,N8= 12
p = 0.0005, Ss = 0.001

A f= =0.113
For

10,000
N =30
The three-stage realisation is shown in Fig. Ell.6(c).

Of A ^; >• 15 - S(z) \2 G(^


L
)

10 KHz 10 KHz 2 KHz 2 KHz 1 KHz 500 Hz

Fig. E11.6(c) Frequency Response for a three-stage Decimation

Bahan dengan hak cipta


A
h(m) Fig.2 11.28
in dB
Passband
*(") 3 Stopband
The5Comb
ti 5Filter6for o =5 5mf ,
2 f , N t= y(m)
The number ofh multiplications per second is given by R
A

= 12 x = 12,000
M. s
Multirate Digital Signal Processing 563

RM n = 30 x = 60,000
5
The overall number of multiplications per second for a three-stage realisation is
given by
^A/.G + R M . S + R M . R = 1*25,500
The number of multiplications per second for a three-stage realisation is more
than that of a two-stage realisation. Hence higher than two- stage realisation may
not lead to an efficient realisation.
Comb Filters
The impulse response of a comb filter (FIR filter) is given by,

Bahan dengan hak cipta


566 Digital Signal Processing

The passband edge is co^, and the stopband edge is cos around n / M , where M
is an integer. Let the transfer function H k ( z ) and the impulse response h k ( n ) be
defined as

A*(n) = h 0 (n) W^*n, k = 0, 1 ....... M - 1


_ 2rt
M
where, WM=e .

H k ( z ) = f>* («)*'"
n=0

= 1*0 (»)
n=0n=0

= H0 ( Z W £ ) , A = 0, 1,..., M - \
The frequency response is

Hk(ejt*) = H0 1 -

right, the frequency response o i H k { z ) is obtained as


2
In general, if the frequency response of H 0 ( z ) is shifted by to the
nk
M
shown in Fig. 11.30. If the impulse responses h k ( n ) are complex, then it is not
necessary that \ H k ( e yu) )| to exhibit symmetry with respect to zero frequency.
The M filters H k ( z ) is the uniformly shifted version of H 0 (z). The magnitude
responses \ H k ( e J i a ) \ for the M-band filters are uniformly shifted versions of
\ H 0 ( e J b i ) | . Hence, the filter bank is called uniform filter bank. The Af-band
filters H k ( z ) can be used as analysis filters or synthesis filters.
Polyphase Implementation
The efficient realisation of the uniform filter bank can be developed using
polyphase decomposition.
The low-pass filter H 0 ( z ) can be expressed in polyphase form as

H0(z)= I*"1 £,(*")


1=0

where E t ( z ) are the polyphase components.

E[(z) = £el(n)z~n
n=0

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i

Index

A/D converters, 523 Adaptive Bartlett Method, 594 Bartlett power


equalisation, 633 Adaptive filters, 631 spectrum estimate, 600, 602
Adaptive FIR filter, 631 Adaptive line Bartlett window, 397, 737 Basic
enhancer, 634 Adaptive noise canceling, realization, 453 Basic structures for IIR
634 Advantages of digital system, 455 systems, 455 Basic wavelet, 678
Advantages of FIR filters, 381 Alias free Basics of AR, MA and ARMA models,
QMF filter bank, 575 Alias free 607
realization, 574 Aliasing, 34, 526, 528 Basis function, 607 BIBO stability, 250
Alternation theorem, 412 Amplitude Bilinear transformation, 427 Binomial
spectrum, 15 Amplitude, 15 series, 776 Blackman and Tukey method,
Analog frequency transformation, 446, 596 Blackman window, 397, 739
449 Blackman-Tukey power spectrum
Analog signals 1^ 2 Analog to Digital estimate, 602, 603 Burg method, 620
converters, 28, 29 Analysis filter bank, Butterworth filters, 432
565, 759 Analysis of speech signal, 660
Antenna beamwidth, 672 Anti-aliasing Canonic structures, 455 Capon method,
filter, 34 Anti-aliasing, 528, 548 Anti- 622 Cascade form, 462
causal, 221 Anti-imaging filter, 531, 549
Aperiodic signal, 4, 5, 306 Application to
Radar, 671 Applications of Laplace
transform, 127:
Applications of Multirate DSP, 523
Applications of wavelet transform, 682
Applications to Image processing, 673
Approximation of derivatives method, 418
AR model, 607
AR process and Linear prediction, 617
Area under F(f), 74
Area under fU), 74
ARMA model, 607
Audio compression, 683
Autocorrelation, 374, 586, 694
Averaging modified periodogram, 595
Averaging periodogram, 574, 594

Bandpass filter, 703, 709, 715, 721, 727,


732
Bandstop filter, 710, 716, 722, 729, 734

Bapan dengan hak cipta


Index 803

Cascade realization of FIR systems, 484 Digital representation of speech signals,


Cascade realization of HR systems, 462 638
Causal and non-causal systems, 20 Causal Digital Signal Processing, 2 Digital
system, 20 Causality, 243 Channel signals 1^ 2 Dirac delta function, 34
vocoders, 664 Chebyshev filters, 439, 724 Direct - Form realization of FIR systems,
Chebyshev window, 741 Circular 483
convolution, 283, 310 Circular correlation Direct - Form realization of IIR systems,
of periodic sequences, 310, 375 456
Classification of signals, l y 3 Coding, 663 Dirichlet's conditions, 41, 56 Discrete
Coefficient quantization, 505, 508 Comb convolution, 279 Discrete Fourier
filter, 563 Transform(DFT), 305
Comparison of Fourier and Wavelet Discrete wavelet transform, 677, 681
transform, 682 Complex conjugation, 1A Discrete-Time Fourier Transform, 305
Complex form of Fourier series, 52 Discrete-time periodic signals, 5 Discrete-
Composite Radix FFT, 352 Compression, time signals, 3, 21, 585 Doppler effect,
663 Computation of filter coefficients, 672 Doppler filtering, 672 Down sampler,
649 323 Down sampling, 34 Duality, 65,1A
Computational requirements, 523, 602 Dyadic sampling, 681 Dynamic systems,
Continuous time systems, 3 Continuous 17
Wavelet Transform (CWT), 677
Continuous-time periodic signals, 5 Echo cancellation, 635, 636, 639 Elliptic
Continuous-time signal, 3, 584 filters, 445
Continuous-time wavelet, 678
Convolution integral, 138, 140
Convolution, 66, 144, 208, 279
Correlation, 210, 373 Cosine function,
130 Cross-correlation, 373, 693

D/A converters, 323 Damped Hyperbolic


sine and cosine functions, 131
Damped sine and cosine functions, 131
Data Compression, 682 Data transmission,
636 Dead band, 51Q Decimation filter,
523, 526 Decimation-in-frequency (DIF)
algorithm, 334
Decimation-in-time (DIT) algorithm, 320
Decimators, 547 Definite integral, 780
Definition of inverse z-transform,
196
Denoising, 684 Design of HR filters, 418
Design of optimal filters, 409 Design
techniques for FIR filters,
385
Design using Kaiser window function,
404
Deterministic and Non-deterministic
signals, 4
DFT in spectral estimation, 591
Differentiation, 207 Digital filter banks,
565 Digital filter design, 551 Digital FM
stereo, 670 Digital frequency
transformation, 450

Bahan dengan hak cipta


804 Index
Encoder, 29 Geometric series, 777 Geometrical
Encoding, 36 construction method,
Energy density spectrum, 262, 584 Energy 269
signal, 104 Energy spectrum, 63 Gibbs phenomenon, 388 Gradient
Equivalent structures, 467 Erros in QMF Adaptive Lattice method, 654
filter bank, 573 Estimation of Group delay, 380
autocorrelation, 586 Estimation of Power
density spectrum, 587, 589 Even and odd Half-band filters, 408 Hamming window,
signals, 7 Even functions, 44 Exponential 395, 743 Hanning window, 397, 745 High
form of Fourier series, 52 pass filter, 702, 707, 713, 716, 725, 731
Exponential function, 129 Exponential Hyperbolic sine and cosine function, 130
pulse, 87 Exponential series, 776 Extra HR direct form structure, 540 HR filter
ripple filter, 412 Fast convolution, 368 bank, 577 HR structures for decimators,
Fast Fourier Transform, 319 550
Filter banks with equal pass bands, 579 Impulse function (Unit impulse), £4
Filter banks with unequal pass bands, 580 Impulse invariant method, 423 Impulse
Filter design for FIR decimator and response, 25, 147, 157, 158, 159,236
interpolator, 554 Infinite Impulse Response (HR) filters,
Filter design for HR decimator and 417
interpolator, 555 Filter structures, 539 Infinite Impulse Response (HR) system,
Final value theorem, 138, 144, 164, 212 257, 417
Finger print compression, 684 Finite Infinite summation formulae, 776 Initial
Impulse Response (FIR) system., 257 value theorem, 137, 144, 164, 211
Finite summation formulae, 775 Finite Interpolation, 523, 530, 547 Inverse
word length effects, 496 Finite-Impulse Chebyshev filters, 444 Inverse CWT, 679
Response (FIR) Filters, 380 Inverse Discrete-Time Fourier Transform,
FIR Decimators and Interpolators, Ml 305 Inverse Fourier transform, 63 Inverse
FIR direct form structure, 539 FIR Half- Laplace transform, 128 Inverse z-
band digital filters, 408 FM stereo transform, 196
transmitter, 670 Folding, 21
Forward-Backward Lattice method, 650 Kaiser window, 402, 748 Kirchoffs
Fourier series method, 385 Fourier series, voltage law, 162
40j 385 Fourier transform pair, 63
Fourier transform, 62, 127 Fractional
Ladder structures, 475 Lag window, 597
sampling rate alternation, 533
Laplace transform of periodic functions,
Frequency and Time domain
154
characteristics, 552 Frequency
Laplace transform pairs, 128, 142 Laplace
convolution, 65, 70 Frequency
transform, 127, 193 L-channel QMF filter
differentiation, 64, 73,
bank, 577 Levinson-Durbin algorithm,
143
612 Limit cycle oscillations, 510
Frequency domain, 397 Frequency
Limitations of non-parametric method,
integration, 65, 73,144 Frequency
604
response, 260, 261, 384 Frequency
Linear and non-linear systems, 18 Linear
sampling method, 389 Frequency shifting
convolution verses circular convolution,
(modulation), 65, 72
284
Frequency transformation, 446, 448
Linear convolution, 284, 291, 691 Linear
Frequency warping, 429
difference equations, 24, 256 Linear
differential equation, 127 Linear filtering,
Gate function, 82 Gaussian pulse, 82 631 Linear phase filters, 384 Linear phase
General polyphase framework, 545 Lth band filters, 571 Linear prediction,

Bahan dengan h3k cipta


Index 805

617 Linear systems, 18 Linear time- Parallel realization of HR system, 464


invariance systems, 236 Linear time- Parametric methods, 606 Parseval's
invariant system, 147 Linearity, 64, 65, identity, 58 Parseval's theorem, 64, 65,
143, 203, 239 Logarithmic series, 777 310 Partial fraction expansions, 144, 216
Long division method, 213 Low pass Periodic and aperiodic signals, 4 Periodic
filter, 700, 706, 712, 718, 724, 729 convolution, 283 Periodic gate function,
Lth band filters, 569 99 Periodic samping, 525 Periodic
waveforms, 41 Periodogram, 589, 599
MA model, 607 MacLaurin series, 776 Phase delay, 380 Phase response, 149, 381
Magnitude and Phase responses, 148 Phase spectra, 15
Magnitude and phase spectrum, 262 Pisarenko Harmonic Decomposition
Magnitude response of Chebyshev filters, method, 623
439 Poles of a Butterworth filters, 434 Poles of
Magnitude response of digital filters, 381 a Chebyshev filters, 441 Polyphase
Magnitude response of Elliptic filters, 446 decomposition, 541. 544 Polyphase FIR
Magnitude response of Inverse Chebyshev structures, 547 Polyphase Framework, 545
filters, 445 Magnitude response, 149 Polyphase HR Decimators, 550 Polyphase
Manipulation of signal flow graphs, 536 IIR filter structures for Decimators, 551
Marr wavelet, 684 Matlab Programs, 688 Polyphase parallel decimators, 550
Matrix inversion lemma, 648 Maximal Polyphase Type Decimators, 547
ripple filter, 412 Mean, 588 Polyphase Type Interpolators, 548 Poly-
Minimization of J(hM), 644, 645, 646 Wiener criterion, 244 Power signal, 103
Minimum mean square error criterion, 643 Power spectral estimation, 591, 610 Power
Modeling Voiced and unvoiced speech spectrum, 59, 586 Product quantization,
sounds, 641 513 Properties of a DSP system, 238
Models of vocal organs, 660 Modulation, Properties of convolution, 281 Properties
72 Mother wavelet, 678 Multilevel filter of CWT, 679 Properties of DFT, 308
banks, 578 Multiplication by tn, 143 Properties of Fourier transform, 64
Multiplication of two sequences, 310 Properties of frequency response, 260
Multiplication theorem, 71 Multirate Properties of Laplace transform, 143
Signal Processing advantages, 523 Properties of ROC, 197 Properties of unit
Multistage decimators and interpolators, impulse function, 11
555 Properties of z-transform, 203 Pulse code
modulation, 674, 675 Pulse train signal,
Network transfer function, 146 Non 99
parametric methods, 593 Non-canonic
structures, 455 Non-causal systems, 20 Quadrature Mirror filter bank, 572
Non-deterministic, 4 Non-linear systems, Quality of power spectrum estimator, 599
18 Non-periodic waveforms, 41 Nyquist Quanitization effect in A to D conversion,
filters, 569 Nyquist period, 524 Nyquist 499 Quantization errors, 519, 520
rate, 33, 524 Quantization, 36 Quantized boxcar signal,
36
Odd functions, 44
Odd signals, 7 Radix-2 FFT, 320 Radix-3 FFT, 352
One-sided Laplace transform, 128 One- Radix-4 FFT, 352 Ramp sequence, 689
sided z-transform, 196, 213 Optimal Random signals, 4 Range resolution 672
linear-phase FIR filter, 409 Output noise Rayleigh's energy theorem, 64 Real time
power, 502 Overflow limit cycles, 512 function, 65 Realization of Linear phase
Overlap add method, 369 FIR systems, 485
Overlap save method, 371 Reconstruction filter, 31, 36 Rectangular
pulse, 84 Rectangular window, 394, 736

Bahan dengan hak cipta


806 Index

Recursive least square algorithm, Step response of Parallel R-L-C circuit,


647 ’ 161
Region of convergence (ROC), 128, Step response of Series R-L-C circuit, 160
196 Steps in the RLS algorithm, 650
Relationship between DFT and other Structures for FIR decimators and
transforms, 306 interpolators, 541 Sub band coding, 523,
Remez exchange algorithm, 413, 782 665 Sub sampling, 523 Symmetry, 43, 65
Representation of discrete-time signals in Synthesis filter bank, 565 System
terms of impulse, 280 Representation of modeling, 631
signals, 13 Representation of systems, 24 Taylor series, 776 Thermal noise, 4
Residue method, 223 Response of LTI Thermal noise, 4 Time advance, 213 Time
systems to arbitrary inputs, 280 Rounding convolution, 65,82 Time delay, 143, 149,
errors, 496 Routh-Hurwitz stability 213, 283 Time differentiation, 64, 73, 143
criterion, Time integration, 65, 73, 143 Time
148 periodicity, 144 Time reversal, 64, 74, 204
Time scaling, 64,
Sampler, 29 Time shifting, 64, 71, 205 Time-frequency
Sampling by impulse function, 32 representation, 675 Time-invariance, 242
Sampling function, 29 Time-variant and time-invariant systems,
Sampling instants, 28
20 tn function, 132 Toolboxes, 688
Sampling of continuous time signals,
Transformation of the independent
. 29
Sampling rate conversion, 525 Sampling variable, 21 Transposed forms, 467
theorem, 34 Sampling, 324 Scalar Triangular pulse, 30 Trigonometric
multiplication, 143 Scale change, 143 Fourier series, 41 Trigonometric
Scaling, 66, 207, 518 Sectioned identities, 774 Trigonometric series, 777
convolution, 368 Shift Truncation errors, 496 Two channel
invariance, 244 Shifting, 21 quadrature Mirror filter bank, 572
Short time Fourier transform (STFT), 675, Two-sided Laplace transform, 128 Two-
676 sided z-transform, 196,
Short time spectrum analysis, 660 Signal
design, 673 Signal flow graphs, 535 Unconstrained Least square method, 621
Signal quantisation and encoding, 86 Uniform DFT filter bank, 565 Unilateral
Signal reconstruction, 35 Signal to Noise Laplace transform, 128 Unit impulse train,
102 Unit impulse, 84 Unit pulse function,
Ratio (SNR), 500 Signum function, 21
11 Unit step function, 10, 95, 129 Unit
Sine function, 130 Singularity functions, 9
step response, 238 Unit-impulse function,
Sinusoidal function, 96 Smoothing the 10 Unit-pulse signal, 688 Unit-ramp
periodogram, 596 Sound synthesis, 683 function, 10 Unit-step sequence, 689
Spectral analysis, 584 Spectral estimation, Unstable systems, 20 Up sampler, 523,
585 Spectrum estimation, 604 Speech 531
analysis synthesis system, 662
Speech signal, 658 s-plane poles and Variance, 588 Video
zeros, 147, 150 Stability, 148, 247 Stable compression, 682
and unstable systems, 20 State equations
for Discrete time systems, 28 State matrix,
26 State space structures, 479 State space,
26 State vector, 26 State-variable
technique, 26 Static and dynamic systems,
12 Step and impulse responses, 147,
157, 158

Bahan dengar
DIGITAL SIGnAL
PRDCESSinG
This book comprehensively covers the undergraduate course on Digital Signal
Processing. Computer usage is integrated into the text in the form of problem solving
using MATLAB. Solved examples and critical -thinking exercises and review
questions enhance the reader s comprehension of the concepts.

Salient Features
^ Overview of Signals and Systems concepts.

^ Comprehensive coverage of key topics, including Infinite Impulse Response Filters


(HR), Finite Impulse Response Filters (FIR), effects of finite word length in digital
filters, and multirate digital signal processing.

^ Review of the essential mathematical concepts such as Fourier Analysis, Laplace


Transforms and Z Transforms.

^ A chapter devoted to Applications of Digital Signal Processing in Speech Processing,


Image Processing and RADAR.

^ Packed with numerous solved examples (221), 576 review questions and practice
exercises.

^ A chapter on MATLAB featuring numerous examples illustrating its application to signal


processing.

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ISBN 0-07-NLWL-K
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