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Documentos de Cultura
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.
= 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
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
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.
(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
u (n )
Example 4.25
ir
Find x (n ) using (i) long division and (ii) partial fraction for X (z ) given by
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 ~'
- - 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 ’
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 )
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:
For poles at z = 1
Residue = -rr lim ^ ^ (2-1)
2-2 n-
0! z -* i dz ( z - l ) ( z1- 2 )
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 )
*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 =
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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
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)= *
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
5'
YT(Z) = Y(s)
_1
(s + 2)(s + 1)
Expressing as a partial fraction,
s+1s+2
y(s) = _I
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
(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
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
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"
(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)
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
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)
“(¥
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
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
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.
y(n) = h (r>)
(5.2)
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)
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
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
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.
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
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
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
Therefore,
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
d£
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. 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.
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Linear Time Invariant Systems 253
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
when n = 1
X \Mk)\ < o o * =0
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.
£ |A(fc)| <-
*=o •
= 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
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 )
X a
ky (n - w=X b
kx (n -w (5.10)
*=0 k=0
X a
kz (y -*)1 = X b iP I* ( n -
A=0 k=0
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)
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
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
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
h i n ) = i n ) * h 2 ( n ) * ••• h L i n )
H(en= £ hWe-j™
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)
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)
Bahan dengan hak c a
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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
= -^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
n=0 1=0
= 0
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
N2 - I
or y(n )= ^h (k )x (n - k ), n = 0, l f . .., N 1 + N 2 - l
k=0
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)
(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:
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
For A = 1
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 + 4 + 4] = 17 4
= 1[60-4J(-1) + 4J (-1)] = 15
4
9
[60 + (-4j) e m n
X3(3) = | + 4jV *'2]
y(0) =
k =-oo
= ... + x(0) M0) + x(l) h (- l) + ... = 0
+ 1 + 0... = !
;y(D= k)
k = ~oo
y(3) = X*(MM3 -M
k=~oo
Alternate Method
Using matrix representation given below, the linear convolution of the given
two sequences can be determined.
= 0 + (1) (4) + (1) (-3) + (0) (-2) + (1) (1) + (1) (0)
+ 0 .. . = 2
When n = -3
y ( - 3)= (k)x(-3 - k)
*=-~
y(-4) = ^ h ( k ) x ( - 4 - k )
*=—
2)W#*
n=0
3)Wt6"*
n=0
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408 Digital Signal Processing
(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
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
+
\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 ( 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)
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«(«)
2 ( z - 1) T
+ 0.1
(z + l)
H(z)
= [f 2 (z - 1)
+ 0.1 +9
T (z + l)
H(,z) =
8.572 - 1184 2_1 + 8.177 z-2
H(s)
= (s + 1) (s + 3)
with T = 0.1s.
5l 2 1
(8.43a)
l + fQ./Q,.) "
(8.43b)
1 + (Q2/Qc)2" 82
(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
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Chapter
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.
Y(z)=
M
= 1 Xa h y ( n - k ) + ^bkx(n- k) -n
\k=l k=0
n = - »o
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)
x{n) x{n-
(c) Unit delay
1)
Fig. 9.1 Basic Building Blocks
H(z)
W(z)
Ci w
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
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)
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.
Fig. E
9.1
Realisation of Digital Linear Systems 459
Direct M
Form and H 2 ( Z ) = X6*2"*
k=0
II
Since
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
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.
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)
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
+
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)
2 { l - z ~ 1 ) { l + ^2z~1 + z ~ 2 )
(c) H ( z ) =
(1 + 0.5z'1 (J 0.9Z'1 + 0.81z
- ~2))
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.
(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 •
( 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.
W ( o) = 1 2 i
^ (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
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.
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
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.
(10.50)
(ra)
a\o = oe X
2 h
* (10.51)
/1 = 0
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
/, = —| -------------------------------- ------------------------ 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.
( 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
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<%a = & 522 Digital Signal Processing
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
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
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
t'
y(n)= 2^ h ( k ) [ x ( n - k ) + x ( n -(AT- 1 -A))}
k=0
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
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.
x(n ) y {m
)
y {m )
= F/M
A f= =0.113
For
10,000
N =30
The three-stage realisation is shown in Fig. Ell.6(c).
= 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,
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
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 -
E[(z) = £el(n)z~n
n=0
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i
Index
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.
^ Packed with numerous solved examples (221), 576 review questions and practice
exercises.
http://highered.mcgraw-hill.com/sites/007463996x
ISBN 0-07-NLWL-K
Tata McGraw-Hill Publishing
Company Limited 7 West Patel
Nagar, New Delhi 110 008
:
Vlslf our Mrbiltr al
wwwtattrocgrawhill com