in JNTU World
BY
2. u (n) =1 n0
=0 n<0
n
u(n) = (k ) = (n) + (n1)+ (n2)…..
k
= (n k )
k 0
=0 else where.
5.Tri (n/N) = 1 n /N n N
=0 else where.
1. Sinc (n/N)= Sa(n /N) = Sin(n /N) / (n /N), Sinc(0)=1
Sinc (n/N) =0 at n=kN, k= 1, 2…
Sinc (n) = (n) for N=1; (Sin (n ) / n =1= (n))
6.Exponential Sequence
x (n) = A n
If A & are real numbers, then the sequence is real. If 0< <1 and A is +ve, then
sequence values are +ve and decreases with increasing n.
For 1< <0, the sequence values alternate in sign but again decreases in magnitude
with increasing n. If >1, then the sequences grows in magnitude as n increases.
7.Sinusoidal Sequence
x(n) = A Cos(won+ ) for all n
If = ejwo
A = A ej
x(n) = A ej n ejwon
= A n Cos(won+ ) + j A n Sin(won+ )
If >1, the sequence oscillates with exponentially growing envelope.
If <1, the sequence oscillates with exponentially decreasing envelope.
So when discussing complex exponential signals of the form x(n)= A e jwon or real
sinusoidal signals of the form x(n)= A Cos(won+ ) , we need only consider frequencies
in a frequency internal of length 2 such as < Wo < or 0 Wo<2 .
V. Deterministic (x (t) = t x (t) = A Sin(wt))
& Nondeterministic Signals. (Ex: Thermal noise.)
VI. Periodic & non periodic based on repetition.
VII. Power & Energy Signals
Energy signal: E = finite, P=0
Signal with finite energy is called energy signal.
Energy signal have zero signal power, since averaging finite energy over infinite
time. All time limited signals of finite amplitude are energy signals.
Ex: one sided or two sided decaying. Damped exponentials, damped sinusoidal.
1
x(t) is an energy signal if it is finite valued and x 2 (t) decays to zero fasten than
t
as t .
Power signal: E = , P 0, P Ex: All periodic waveforms
Neither energy nor power: E= , P=0 Ex: 1/ t t 1 E= , P= , Ex: tn
VIII. Based on Symmetry
1. Even x(n)=xe(n)+xo(n)
2. Odd x(n)=xe(n)+xo(n)
3. Hidden x(n)=xe(n)xo(n)
1
4. Halfwave symmetry. xe(n)= [x(n)+x(n)]
2
1
xo(n)= [x(n)x(n)]
2
Signal Classification by duration & Area.
a. Finite duration: time limited.
b. Semiinfinite extent: right sided, if they are zero for t < where = finite
X rms = P
1 2 1 2
E= A2 b E= A b E= A b
2 3
Q.
e  t dt =
1
0
Q.
1 2 1
Ex = A 0.5T + (A)2 0.5T = 0.5 A2 T
2 2
Px = 0.5 A2
Q.
1 1 2
Ey = [ A2 0.5T] 2 = A T
3 3
1 2
Py = A
3
(t 2) t 2
y (t) = 2 x [ ] = 2 x[ ] 2 x( t + ) ; 5 + =1;  + =1 => = 1/3
3 3 3
; = 2/3
Area of symmetric signals over symmetric limits ( , )
Odd symmetry:
x0 (t) dt =0
Even symmetry:
xe (t) dt = 2
0
xe (t) dt
Area of any sinc or Sinc 2 equals area of triangle ABC inscribed within the main lobe.
10
Even though the sinc function is square integrable ( an energy signal) , it is not
1
absolutely integrable( because it does not decay to zero faster than )
t
(t) = 0 t 0
= t=0 ( )d
=1
An impulse is a tall narrow spike with finite area and infinite energy.
The area of impulse A (t) equals A and is called its strength. How ever its hight at
t=0 is .
2 2
I 2 = cos(2 t ) (2t 1)dt = cos(2 t )0.5 (t 0.5)dt = 0.5 cos(2 t) at t=0.5 = 0.5
4 4
x1(t) = x(t)
k
(tkts ) =
k
x(kts) (tkts)
11
’(t) =0 t 0
1
[ [t ]] = (t )
Differentiating on both sides
1
’ [ [t ]] = ' (t )
With =1
’ (t) =  ’ (t)
d
[ x(t ) (t )] = x’ (t) (t ) + x (t) ’ (t )
dt
= x’ ( ) (t ) + x (t) ’ (t )1
Or
d d
[ x(t ) (t )] = [ x( ) (t )] = x ( ) ’ (t ) 2
dt dt
1=2
x’ ( ) (t ) + x (t) ’ (t ) = x ( ) ’ (t )
12
x (t) ’ (t ) dt =
x ( ) ’ (t ) dt 
x’ ( ) (t ) dt
= 0 x’ ( ) =  x’ ( )
Higher derivatives of (t) obey n(t) = (1)n n(t) are alternately odd and even,
and possess zero area. All are eliminating forms of the same sequence that generate
impulses, provided their ordinary derivatives exits. None are absolutely integrable.
The impulse is unique in being the only absolutely integrable function from among
all its derivatives and integrals (step, ramp etc)
What does the signal x(t) = et ’(t) describe?
x(t) = ’ (t) – (1) (t) = ’ (t) + (t)
2
d
= 0.5 (t3) t 1  8 [cos t ] t 0.5
dt
= 23.1327 Answer.
Operation on Signals:
1. Shifting.
x(n) shift right or delay = x(nm)
x(n) shift left or advance = x(n+m)
2. Time reversal or fold.
x(n+2) is x(n) delayed by two samples.
x(n2) is x(n) advanced by two samples.
Or
x(n) is right shift x(n2), then fold x(n2)
x(n) fold x(n) shift left x((n+2)) = x(n2)
Ex:
x(n) = 2, 3 , 4 , 5, 6, 7 .
13
{ 1 , 2, 6, 4, 8} { 1 , 1,2,2,6, 6,4,4,8, 8} { 1 , 2, 6, 4, 8}
n n/2 n 2n
Since Decimation is indeed the inverse of interpolation, but the converse is not
necessarily true. First Interpolation & Decimation.
Ex: x(n) = { 1 1, 2, 5, 1}
14
4 5 2 1
= { 1, , , 2 , 3,4,5,3,1,1,  , } Linear interpolation.
3 3 3 3
4. Fractional Delays.
M ( Nn M )
It requires interpolation (N), shift (M) and Decimation (n): x (n  )=x( )
N N
2n 1
x(n) = {2, 4, 6 , 8}, find y(n)=x(n0.5) = x ( )
2
g(n) = x (n/2) = {2, 2, 4, 4, 6 , 6, 8,8} for step interpolation.
n 1
h(n) =g(n1) = x( ) = {2, 2, 4, 4 , 6, 6,8,8}
2
2n 1
y(n) = h(2n) = x(n0.5) = x( ) = {2, 4 , 6, 8}
2
OR
g(n) = x(n/2) = {2,3,4,5, 6 ,7,8,4} linear interpolation.
g (n) = h(2n)={3,5,7,4}
Classification of Systems
1. a. Static systems or memory less system. (Non Linear / Stable)
Ex. y(n) = a x (n)
= n x(n) + b x3(n)
= [x(n)]2 = a(n1) x(n)
y(n) = [x(n), n]
If its o/p at every value of ‘n’ depends only on the input x(n) at the same value of ‘n’
Do not include delay elements. Similarly to combinational circuits.
b. Dynamic systems or memory.
If its o/p at every value of ‘n’ depends on the o/p till (n1) and i/p at the same value of
‘n’ or previous value of ‘n’.
15
This system computes the nth sample of the o/p sequence as the average of (m1+m2+1)
samples of input sequence around the nth sample.
If M1=0; M2=5
5
y(7) = 1/6 [ x (7 k ) ]
k 0
n 1
= x(k ) + x(n)
k
= y(n1) + x(n)
16
x(n) = { …0,3,2,1,0,1,2,3,0,….}
y(n) = { …0,3,5,6,6,7,9,12,12…}
O/p at the nth sample depends on the i/p’s till nth sample
Ex:
x(n) = n u(n) ; given y(1)=0. i.e. initially relaxed.
1 n
y(n) = x(k ) + x(k )
k k 0
n n
n(n 1)
= y(1) + x(k ) = 0 +
k 0
n =
k 0 2
5. Linear Systems.
If y1(n) & y2(n) are the responses of a system when x1(n) & x2(n) are the respective
inputs, then the system is linear if and only if
[ x1(n) x2(n)] = [ x1(n)] + [ x2(n)]
17
For every bounded input x(n) Bx < for all n, there exists a fixed +ve finite value
y(n) = [ x(k ) (n k ) ] for linear
k
Therefore o/p of any LTI system is convolution of i/p and impulse response.
y(no) = h(k ) x(no k )
k
1
= h(k ) x(no k ) + h(k ) x(no k )
k k 0
n
If i/p is also causal y(n) = h( k ) x ( n k )
k 0
18
S=
k
h(k ) is necessary & sufficient condition for stability.
19
= h( k ) x ( n k )
k
y (n+N) = h( k ) x ( n k N )
k
put nk = m
= h(n m) x(m N )
m
= h(n m) x(m)
m
m=k
= h( n k ) x ( k )
k
= y(n) (Ans)
Y ( z)
=10.5 Z1 System
X ( z)
Inverse System
y (n) – 0.5 y(n1) =x(n)
Y (z) [10.5 Z1] = X (z)
Y ( z) 1 1
[10.5 Z ]
X ( z)
21
N
always.
y(n) = ak x(nk)
k 0
y(n) – y(n1) = x(n) – x(n3)
Present response depends only on present
i/p & previous i/ps but not future i/ps. It gives
FIR o/p.
1
Q. y(n) = [x (n+1) + x (n) + x (n1)] Find the given system is stable or not?
3
Let x(n) = (n)
1
h(n) = [ (n+1) + (n) + (n1)]
3
1
h(0) =
3
1
h(1) =
3
1
h(1) =
3
S= h(n) < therefore Stable.
22
1
y(1) = [ y(0) – x(0)]=0
a
y(2) = 0
1
Q. y(n) = y(n1) + x(n) for n 0
n 1
=0 otherwise. Find whether given system is time variant or not?
Let x(n) = (n)
h (0) = 1 y(1) + (0) = 1
h(1) = ½ y(0) + (1) = ½
h(2) = 1/6
h(3) = 1/24
if x(n) = (n1)
y(n) = h(n1)
1
h(n1) = y(n) = h(n2) + (n1)
n 1
n=0 h(1) = y(0) = 1 x 0+0 =0
n=1 h(0) = y(1) = ½ x 0 + (0)= 1
n=2 h(1) = y(2) = 1/3 x 1 + 0 = 1/3
h(2) = 1/12
h (n, 0) h(n,1) TV
Q. y (n) = 2n x(n) Time varying
1
Q. y (n) = [x (n+1) + x (n) + x (n1)] Linear
3
Q. y (n) = 12 x (n1) + 11 x(n2) TIV
Q. y (n) = 7 x2(n1) non linear
Q. y (n) = x2(n) non linear
Q. y (n) = n2 x (n+2) linear
Q. y (n) = x (n2) linear
Q. y (n) = ex(n) non linear
Q. y (n) = 2x(n) x (n) non linear, TIV
23
(If the roots of characteristics equation are a magnitude less than unity. It is a
necessary & sufficient condition)
Non recursive system, or FIR filter are always stable.
Q. y (n) + 2 y2(n) = 2 x(n) – x(n1) non linear, TIV
Q. y (n)  2 y (n1) = 2x(n) x (n) non linear, TIV
Q. y (n) + 4 y (n) y (2n) = x (n) non linear, TIV
Q. y (n+1) – y (n) = x (n+1) is causal
Q. y (n)  2 y (n2) = x (n) causal
Q. y (n)  2 y (n2) = x (n+1) non causal
Q. y (n+1) – y (n) = x (n+2) non causal
Q. y (n2) = 3 x (n2) is static or Instantaneous.
Q. y (n) = 3 x (n2) dynamic
Q. y (n+4) + y (n+3) = x (n+2) causal & dynamic
Q. y (n) = 2 x ( n )
If =1 causal, static
<1 causal, dynamic
>1 non causal, dynamic
1 TV
Q. y (n) = 2(n+1) x (n) is causal & static but TV.
Q. y (n) = x (n) TV
Solution of linear constantcoefficient difference equation
Q. y(n)3 y (n1) – 4 y(n2) = 0 determine zeroinput response of the system;
Given y(2) =0 & y(1) =5
Let solution to the homogeneous equation be
yh (n) = n
 3  4 n2 =0
n n1
[ 2  3  4] =0
n2
= 1, 4
yh (n) = C1 1n + C2 2n = C1(1)n + C2 4n
y(0) = 3y(1) +4 y(2) = 15
24
C1+ C2 =15
y (1) = 3y (0) +4 y (1) = 65
C1+4C2 = 65 Solve: C1 = 1 & C2=16
y(n) = (1)n+1 + 4n+2 (Ans)
If it contain multiple roots yh(n) = C1 1n + C2 n 1n + C3 n2 1n
or 1n [C1+ nC2 + n2 C3….]
Q. Determine the particular solution of y(n) + a1y(n1) =x(n)
x(n) = u(n)
Let yp (n) = k u(n)
k u(n) + a1 k u(n1) =u(n)
To determine the value of k, we must evaluate this equation for any n 1
k + a1 k =1
1
k=
1 a1
1
yp (n) = u(n) Ans
1 a1
x(n) yp(n)
1. A K
2. Amn Kmn
3. Anm Ko nm + K1nm1 + …. Km
4. A Coswon or A Sinwon K1 Coswon + K2 Sinwon
5 1
Q. y(n) = y(n1)  y(n2) + x(n) x(n) = 2n n 0
6 6
Let yp (n) = K2n
5 1
K2n u(n) = K 2n1 u(n1)  K 2n2 u(n2) + 2n u(n)
6 6
For n 2
5 1
4K = (2K)  K +4 Solve for K=8/5
6 6
8 n
yp (n) = 2 Ans
5
Q. y(n) – 3 y(n1)  4 y(n2) = x(n) + 2x(n1) Find the h(n) for recursive system.
25
=0.5
yh(n) = C (0.5)n
yp(n) = K1 cos 0.5n + K2 sin 0.5n
yp(n1) = K1 cos 0.5(n1) + K2 sin 0.5(n1)
=  K1 sin 0.5n  K2cos 0.5n
yp(n)  0.5 yp(n1) = 5 cos 0.5 n
= (K1 + 0.5 K2) cos 0.5 n (0.5 K1 – K2) sin 0.5n
K1 + 0.5 K2 = 5
0.5 K1 – K2 =0 Solving we get: K1= 4 & K2=2
yp(n) = 4 cos 0.5 n + 2 sin 0.5n
26
27
[Complex exponential and sinusoidal sequences are not necessarily periodic in ‘n’
2
with period ( ) and depending on Wo, may not be periodic at all]
Wo
N = fundamental period of a periodic sinusoidal.
3. The highest rate of oscillations in a discrete time sinusoid is obtained when
w = or 
28
W= Ts
2 f = 2 F Ts => f =1
;
2 F = 4 2 f = /4
29
1 1
F= ; T= 8 ; f= N=8
8 8
7. Discretetime sinusoids are always periodic in frequency.
Q. The signal x (t) = 2 Cos (40 t) + Sin (60 t) is sampled at 75Hz. What is the
common period of the sampled signal x (n), and how many full periods of x (t) does it
take to obtain one period of x(n)?
F1 = 20Hz F2 = 30Hz
20 4 K1 30 2 K 2
f1 = f2 =
75 15 N1 75 5 N 2
The common period is thus N=LCM (N1, N2) = LCM (15, 5) = 15
The fundamental frequency Fo of x (t) is GCD (20, 30) = 10Hz
1
And fundamental period T = 0.1s
Fo
Since N=15
1
1sample  sec
75
15
15 sample  ? => 0.2S
75
30
So it takes two full periods of x (t) to obtain one period of x (n) or GCD (K 1, K2) =
GCD (4, 2) = 2
Frequency Domain Representation of discretetime signals and systems
For LTI systems we know that a representation of the input sequence as a weighted
sum of delayed impulses leads to a representation of the output as a weighted sum of
delayed responses.
Let x (n) = ejwn
y (n) = h (n) * x (n)
=e jwn
h(k ) ejwk
k
jw
Let H (e ) = h(k ) ejwk is the frequency domain representation of the system.
k
y (n) = H (e ) e
jw jwn
ejwn = eigen function of the system.
H (ejw) = eigen value
Q. Find the frequency response of 1st order system y (n) = x (n) + a y (n1)
(a<1)
Let x (n) = ejwn
yp (n) = C ejwn
C ejwn = ejwn + a C ejw (n1)
C ejwn [1aejw] = ejwn
1
C=
[1 ae jw ]
1 1
Therefore H (ejw) = =
[1 ae jw ] 1 a(cos w j sin w)
1
H (e jw ) =
1 2a cos w a 2
aSinw
H (e jw ) Tan 1 ( )
1 aCosw
31
1
Q. Frequency response of 2nd order system y(n) = x(n)  y (n 2)
2
x (n) = e jwn
y (n) ce jwn
p
1 jw( n 2)
c e jwn = e jwn  ce
2
1 1 20 16Cos2 w
c e jwn (1+ e 2 jw ) = e jwn c= c
2 1
1 e 2 jw 5 4Cos2 w
2
Sin 2 w
c tan1
2 Cos2 w
32
UNIT  II
c e
jK n
N
DTFS
Non periodic Periodic Ck =
T
1 N 1 2
Ck = T
0
f (t )e jKot dt 1
x p ( n )e
j
N
nK
N n 0
2nTs
1 jK
k=0 to N1
T
x ( n)e
NTs
T = N Ts
t = n Ts : dt = Ts
NonPeriodic f(t) = Non – Periodic x(n) =
2
1 1
F (w)e X (w)e
jt
d jwn
dw
2
2 0
X(w) = FT of DTS
f (t )e jt dt
33
1
=
x ( n ) x ( n ) x *
( n ) x ( n ) X *
( w) e jwn
dw
E=
n n n 2 0
2
1 jwn
=
2 0 X ( w)
*
n
x ( n ) e dw
2
1
X
*
= 2 ( w) X ( w)dw
0
2
1
=
2 X ( w)
dw
2 2
1
Therefore: E= x ( n )
2 X (w) dw  Parsval’s Theorem
n
N 2
1
P = Lt
N 2 N 1
n N
x ( n) for non periodic signal
N 1 2
1
= N
n 0
x ( n) for periodic Signal
2
1 N 1 1 N 1 N 1
* j N nk
=
N n 0
x ( n) x ( n) x ( n) C k e
*
N n 0 k 0
2
* 1
N 1 N 1 j
x ( n)e
nk
Ck N
=
k 0 N n 0
N 1 2 N 1 2
Therefore P= Ck 0
k E=N Ck
k 0
34
N
1
P = Lt
N 2 N 1 n 0
u 2 ( n)
N 1 1
= Lt Power Signal
N 2 N 1 2
E=
jwon
Ex: x (n) = Ae
N 2
1
P= Lt
N 2 N 1 n N
Ae jwon
1
= Lt A 2 [1 1 ........]
N 2 N 1
A 2 (2 N 1)
= Lt 2 N 1 A 2
it is Power Signal and E=
N
2n
Ex: x (n) = 6 Cos whose period is N=4 x (n) = { 6 ,0,6,0 }
4
1 3 2 1
P=
4 n 0
x ( n )
4
[36 36] 18W
2n
j
4
Ex: x (n) = 6 e whose period is N = 4
2
1 3 1
P=
4 n 0
x ( n )
4
[36 36 36 36] 36Watts
35
DISCRETE CONVOLUTION
u(k) = 0 k<0
u(nk) = 0 k>n
n
u(k )u(n k )
n
k 0
= 1 = (n+1) u(n) = r(n+1)
k 0
36
y(n) = k 0
ak ank = an (n+1) u(n)
y(n) =
k
k
u(k) u(nk) =
k 0
k
= (1
n+1
) / (1 )
The convolution of the left sided signals is also left sided and the convolution of two
right sided also right sided.
n n
Q. x(n) = rect ( )=1 N
2N
=0 else where
h(n) = rect ( n )
2N
37
Q. x(n) = {2,1,3}
h(n) = { 1,2,2,3} Graphically Foldshiftmultiplysum
y(n) =
1 2 2 3
2 2 4 4 6
1 1 2 2 3
3 3 6 6 9
y(n) = { 2,3,5,10,3,9}
Q. x(n) = {4, 1
,3} h(n) = { 2,5, 0
,4}
2 5 0 4
4 8 20 0 16
1 2 5 0 4
3 6 15 0 12
y(n) = { 8,22,11,31,4, 12 } Note that convolution starts at n=3
38
Q)
h(n): 2 5 0 4
x(n): 4 1 3
_________________________________
8 20 0 16
2 5 0 4
6 15 0 12
____________________________________
y(n): 8 22 11 31 4 12
i) 2 5 0 4 ii) 2 5 0 4
3 1 4 3 1 4
___________________ _________________________
y(0) = 8 2 20 y(1) = 2+20 = 22
iii) 2 5 0 4 iv) 2 5 0 4
3 1 4 3 1 4
________________________ _______________________
6 5 0 y(2) = 11 15 0 16 y(3) = 31
v) 2 5 0 4 Vi) 2 5 0 4
3 1 4 3 1 4
________________________ _______________________
0 4 y(4) = 4 12 y(5) = 12
If we insert zeros between adjacent samples of each signal to be convolved, their
convolution corresponding to the original convolution sequence with zeros inserted
between its adjacent samples.
39
y(n) = 8 ,22,11,31,4,12
h(n) = 2 , 0, 5, 0, 0, 0, 4 ; x(n) = 4, 0, 1, 0, 3
40
b. x (n) = Cos n after expansion x(n)={ 1,0.5,0.5,1,0.5,0.5}
3
1
fo = N=6
6
2
1 5 j
Ck =
nk
6
x ( n )e k=0 to 5
6 n 0
1 j k
2 4 5
j k j k j k
jk
x(0) x(1)e x(2)e x(3)e x(4)e x(5)e 3
3 3 3
Ck = 6
For k=0 Co =
1
x(0) x(1) x(2) x(3) x(4) x(5) = 0
6
Similarly
K=1 C1 = 0.5 , C2 = 0 = C3 = C4 , C5 = 0.5
Or
2 2 2
1 j 6 n 1 j 5
C
n j kn
= Cos n e
6
x (n) + e = k e 6
3 2 2 k 0
41
2 4 6 8 10
j n j n j n j n j n
6 6 6 6 6
= Co+C1 e +C2e + C3 e +C4 e +C5 e
1
By comparison C1=
2
2 56 10n
j n j 2 n j
6 6 6
Since e =e =e
1
C5
2
c. x (n) = {1,1,0,0}
nk
1 3 j 2
Ck= 4
4
x ( n ) e k=0, 1, 2, 3
n 0
2k
1 j
1 1e
2
= 4
1
c0 2 ; c1
1
4
1 j ; c2 0 ; c3
1
4
1 j
1
Co & C0 = 0
2
2
c 1
& C1 =
4 4
c 2
0 & C2 undefined
2
c 3
& C3 =
4 4
42
PROPERTIES OF DFS
1. Linearity
DFS x ( n m) e
j
~ N
Ck
3. Symmetry
N
~
x ( n )e
n 0
N
N 1
~
x (n) C
* *
k
~
x (n) C k e
j2
N
nk
DFS
k 0
~ ~ ( n) 1
*
DFS
Re~
x ( n ) DFS
x ( n ) x
C k C *
k C ke
2 2
~ ~ ( n) 1
*
j Imx (n) DFS
~ x ( n ) x
C k C * k C ko
DFS
2 2
~
x ( n)
If is real then
~
x ( n ) ~
x *
( n)
~
x e ( n)
2
43
~ ~x ( n) ~
x * ( n)
x o ( n)
2
DFS
~
x e ( n )
1
2
C k C *
k ReC k
DFS
~
x o ( n )
1
C k
2
C *
k j ImC k
Periodic Convolution
N 1 ~
1
~
x ( m) x 2 ( n m) C k1C k 2
DFS
m 0
If x(n) is real
Ck C * k
Re[Ck ] Re[Ck ]
Im[Ck ] Im[Ck ]
Ck Ck
Ck Ck
PROPERTIES OF FT (DTFT)
1. Linearity
y (n) = a x1 (n) + b x2 (n)
Y (e jw ) = a X1(e jw ) + b X2(e jw )
2. Periodicity
j ( w 2 )
H (e ) = H (e jw )
3. For Complex Sequence
44
H (e ) = jw
[ h
n 
R (n) j h I (n) ] [Cos(wn)  j Sin(wn)]
[ h
n 
R (n) Cos(wn) h I (n)Sin(wn) = HR (e jw
)
[ h (n) Cos(wn) h
n 
I R (n)Sin(wn) = HI (e jw
)
H (e jw ) = H R (e jw ) jH I (e jw )
= H R2 (e jw ) H I2 (e jw ) H (e jw ) H * (e jw )
H I (e jw ) 1
H (e ) tan jw
jw
R
H ( e )
4. For Real Valued Sequence
H (e ) jw
= h(n)e
n
jwn
= h(n)Cos(wn) j h(n)Sin(wn)
n n
= H R (e jw ) jH I (e jw )  (a)
H (e jw
) = h(n)e
n
jwn
= h(n)Cos(wn) j h(n)Sin(wn)
n n
= H R (e jw ) jH I (e jw )  (b)
From (a) & (b)
H R (e jw ) H R (e jw )
45
H I (e jw ) H I (e jw )
Real part is even function of w
Imaginary part is odd function of w
H (e jw ) H * (e jw )
H I (ejw ) 1 H I (e
jw
)
H (e  jw
) tan 1
 jw
tan jw
H (e jw )
H R (e ) H R (e )
Phase response is odd function.
5. FT of a delayed Sequence
FT [h (nk)] = h( n k )e
n
jwn
Put nk = m
= h(m)e
m
jw ( m k )
e jwk
jwk
= e h(m)e jwm = H (e jw )
m
6. Time Reversal
x (n) X (w)
x (n) X (w)
F T [x (n)] = x ( n)e
n
jwn
Put –n = m
x
( m) e jwm
X ( w)
46
7. Frequency Shifting
F T [x (n) e jwo n ]=
n
x (n) e jwo n e jwn
=
n
x (n) e j ( w wo ) n = X (wwo)
8. a. Convolution
x1 (n) * x2 (n) X1(w) X2(w)
n
[x1 (n) * x2 (n) ] ejwn =
n k
[ x1 (k) x2 (nk) ] ejwn
Put nk = m
=
n
x1 (k)
m
[x2(m)] ejw (m+k)
=
n
x1 (k) e jwk
m
[x2(m)] ejwm
= X1(w) X2(w)
1
b. [X1(w) * X2(w)] x1 (n) x2 (n)
2
9. Parsevals Theorem
1
n
*
x1(n) x2 (n) =
2
[X1(w) X2*(w)] dw
dX ( w)
n x (n) j
dw
10.F T of Even Symmetric Sequence
H (e
jw
)=
n
h (n) ejwn
47
1
=
n
h (n) ejwn + h (0) +
n 1
h (n) ejwn
Let n = m
=
m 1
h (m) e jwm
+ h (0) +
n 1
h (n) ejwn
frequency
0 ; H (e jw ) >0
; H (e jw ) <0
11.F T of Odd Symmetric Sequence
For odd sequence h (0) = 0
H (e jw ) =
n 1
h (n) [ejwn  ejwn ]
= j 2 n 1
h (n) Sin (wn) HI (e jw ) is a imaginary valued function
H (e jw ) = HI (e jw ) for HI (e jw ) > 0
=  HI (e jw ) for HI (e jw ) < 0
H (e jw ) For w over which HI (e jw ) > 0
2
= for w over which HI (e jw ) < 0
2
2
1
12. x(0) = 2 X (w)dw
0
Central Coordinates
48
X (0) = x ( n)
n
13. Modulation
X ( w w0 ) X ( w w0 )
Cos (won) x (n)
2 2
FOURIER TRANSFORM OF DISCRETE TIME SIGNALS
X (w) =
n
x (n) ejwn
F T exists if x(n)
n
w
0 1
1
2 3
1

3
49
) = a e jwn
n
jw
H (e
n 0
1
= (ae
n 0
jw n
) =
1 ae jw
Q. x(n) = n n u(n) <1
d 1
j
dw 1 e
n
n u(n) jw
e jw
= (1 e jw ) 2
1
Hint: n
u(n)
n 0
n
e jwn
= (e jw ) n
n 0
=
1 e jw
Q. x(n) = n 0nN
Or
x(n) = n [ u(n) – u(nN)]
= n u(n) – N nN u(nN) Using Shifting Property
1 e jwN
[
N
]
X(w) =
1 e jw
1 e jw
1 (e jw ) N
=
1 e jw Ans
Q. x(n) = n
1 two sided decaying exponential
x(n) = n u(n) + n u(n)  (n) using folding property
1 1 1 2
1 =
=
1 e jw 1 e jw 1 2Cosw 2
Q. x (n) = u (n) Since u (n) is not absolutely summable
50
1
we know that u (t) ( w) jw
1
(w)
Similarly X (w) =
1 e jw +
51
Since X (w) is periodic with period 2 , sample X(w) periodically with N equidistance
2
samples with spacing w .
N
K = 0, 1, 2…..N1
2
2k j
Kn
X x ( n )e N
N n
The summation can be subdivided into an infinite no. of summations, where each sum
contains
N 1 2
2 j
2k
x(n)e
1 Kn
j
............ x(n)e
Kn N
X N
+
N n N n 0
2 N 1 2
j
x(n)e
Kn
N
..................
n N
52
lN N 1 2
j
n
Kn
N
x(n)e
=
lN
l
Put n = nlN
x(n lN )e
N 1 2
j K ( n lN )
N
=
l n 0
2
j
l
N 1 Kn
=
x(n lN )e N
n 0
N 1 2
j
Kn
N
X(k) = xp(n) e
n 0
N 1 2
j Kn
N
We know that xp(n) = Ck e n= 0 to N1
k 0
N 1 2
Ck=
1
N
n 0
xp(n) e
j
N
Kn
k=0 to N1
1
Therefore Ck= X(k) k=0 to N1
N
N 1
2
j Kn
IDFT  xp (n) =
1
N
k 0
X(k) e N
n = 0 to N1
This provides the reconstruction of periodic signal xp(n) from the samples of spectrum
X(w).
The spectrum of aperiodic discrete time signal with finite duration L<N, can be exactly
2k
recovered from its samples at frequency Wk= .
N
Prove: x(n) = xp (n) 0 n N1
53
Using IDFT
N 1 2
1
j Kn
x (n) = X(k) e N
N k 0
N 1 N 1 2
1
j Kn
X (w) = [ X (k) e N
] ejwn
n 0 N k 0
N 1 N 1 2
1 jn ( w
K)
= X (k) [ e N
]
k 0 N n 0
N 1
1
If we define p(w) =
N
n 0
ejwn
wN
N 1 Sin
1 1 e 2
jwN
jw
2
= = e
N 1 e jw w
NSin
2
N 1
2k
Therefore: X (w) = X (k) P(w )
k 0 N
2k
At w = P (0) =1
N
2k
And P (w ) = 0 for all other values
N
N 1 N 1
2k
X (w) = X(k) = X( )
k 0 k 0 N
54
0
xp (n) = x(n LN )
l
a
l
n lN
n
a
= a n a lN 0 n N1
l 0 1 aN
N 1
an N 1
=
1
n 0 1 a
N
e jwn =
1 aN
(ae
n 0
jw n
)
1 1 a N e jwN
Xˆ ( w)
1 a N 1 a e jw
55
2k
j N
2K 1 1 a N
e N
Xˆ N 2K
N 1 a 1 a e j N
1 2K
= j 2k = X
1 ae N N
2k
Although Xˆ (w) X ( w), the samples at Wk= are identical.
N
1 1
Ex: X (w) = & X (k) = 2
j
1 a e jw
k
1 a e N
Apply IDFT
2nk
j
N 1
e
N
1
x (n) = N j
2k using Taylor series expansion
k 0
1 ae N
1 N 1 j 2Nnk r j 2Nkr
= N e
k 0
a e
r 0
1 N 1 j 2k ( nNr )
=
N
a e
r
r 0 k 0
=0 except r = n+mN
x (n) = a
m 0
n mN
= a
n
(a
m 0
N
)m
an
=
1 aN
The result is not equal to x (n), although it approaches x (m) as N becomes .
56
3 2
j
x ( n )e
n
4
X (1) = = 2+2j
n 0
X (2) = 2
X (3) = 22j
DFT as a linear transformation
2
j
Let WN e N
N 1
X (k) =
nk
x(n)W N k = 0 to N1
n 0
N 1
1
x (n) =
N
X (k )W
k 0
nk
N n = 0, 1…N1
X ( 0)
x ( 0)
X (1)
x(1)
Let xN = XN =
x( N 1) X ( N 1)
1 1 1 1
1 W 1 W 2
W N( N 1)
N N
1 W N2 W N4 W N2 ( N 1)
WN =
1 W NN 1 W N2 ( N 1) WN( N 1)( N 1)
57
DFT IDFT
1
XN = WN xN xN = W N* X N
N
x N WN1 X N
1. WNK N WNK
N
1 1 K
W N WN* 2. WN 2
WNK
N
1 1 1 1 1 1 1 1 1 1 1 1
1 W 1 W 2 W 3 1 W 1 W 2 W 3 1 j 1 j
4 4 4 4 4 4
DFT W4 = 1 W 2 W 4 W 6 = 1 W4 W4 W42
2 0 = 1 1 1 1
1
4 4 4
9
1 j 1 j
3 6 3 2
1 W4 W4 W4 1 W4 W4 W4
1 1 1 1 0 6
1 j 1 j 1 2 2 j
X4 = W4 x 4 = 1 1 1 1 2 = 2
1 j 1 j 3 2 2 j
IDFT
0
1 1 1 1 6 1
1 j 1 j 2 2 j
1 * 1
x4 = WN X N = 2 = 2 Ans
4 4 1 1 1 1
1 j 1 j 2 2 j 3
Q.
x (n) = { 1,0.5}
h (n) = { 0.5,1 }
58
Find y (n) = x (n) h (n) using frequency domain. Since y (n) is periodic with period 2.
Find 2point DFT of each sequence.
X (0) = 1.5 H (0) = 1.5
X (1) = 0.5 H (1) = 0.5
Y (K) = X (K) H (K)
Y (0) = 2.25 Y (1) = 0.25
Using IDFT y (0) = 1; y (1) = 1.25
~y (n) h~(n) ~x (n) ~ ~
h (k ) x (n k )
k
~
=
~
x
k
( k ) h (n k )
~
~
y (0) = ~
x (
k
k ) h (k )
~ ~ ~ ~
= x (0)h (0) x (1)h (1)
= 1 * 0.5 + 0.5 * 1 = 1
~
~
y (1) = ~
x (
k
k ) h (1 k )
59
~ ~ ~ ~
= x (0)h (1) x (1)h (0)
= 1 * 1 + 0.5 * 0.5 = 1.25
~ ~
y (2) = ~
x (
k
k ) h (2 k )
~ ~ ~ ~
= x (0)h (2) x (1)h (1)
= 1 * 0.5 + 0.5 * 1 = 1
~
y ( n) = {1, 1.25, 1, 1.25…..}
Q. Find Linear Convolution of same problem using DFT
Sol. The linear convolution will produce a 3sample sequence. To avoid time
aliasing we convert the 2sample input sequence into 3 sample sequence by padding with
zero.
For 3 point DFT
X (0) = 1.5 H (0) = 1.5
2
j
2 j
3
X (1) = 1+0.5 e 3
H (1) = 0.5+ e
4 4
j j
3
X (2) = 1+0.5 e H (2) = 0.5+ e 3
60
y(0) = 0.5
y(1) =1.25
y(2) =0.5
y(n) = { 0.5, 1.25, 0.5} Ans
PROPERTIS OF DFT
1. Linearity
If h(n) = a h1(n) + b h2(n)
H (k) = a H1(k) + b H2(k)
2. Periodicity H(k) = H (k+N)
~
3.
h ( n) h(n mN )
m
4. y(n) = x(nn0)
2kn0
j
N
Y (k) = X (k) e
5. y (n) = h (n) * x (n)
Y (k) = H (k) X (k)
6. y (n) = h(n) x(n)
Y (k) =
1
H (k ) X (k )
N
7. For real valued sequence
N 1
2kn
H R (k ) h(n)Cos
n 0 N
N 1
2kn
H I (k ) h(n) Sin
n 0 N
a. Complex conjugate symmetry
h (n) H(k) = H*(Nk)
h (n) H(k) = H*(k) = H(Nk)
61
H (k ) H ( N k )
e. Phase function is odd function
H (k ) H ( N k )
f. If h(n) = h(n)
H (k) is purely real
g. If h(n) = h(n)
H (k) is purely imaginary
8. For a complex valued sequence
x*(n) X*(Nk) = X*(k)
N 1
N 1
X*(k) = x
n 0
*
( n)W N n k
N 1
*
X (Nk) = x
n 0
*
(n)W Nnk = X*(k)
N 1
DFT [x*(n)] = x
n 0
*
(n)W Nnk = X*(Nk) proved
9. Central Coordinates
N 1
1
X (k )
N 1
1
x (0) =
N k 0
x(
N
2
)=
N
(1)
k 0
k
X (k ) N=even
N 1 N 1
X (0) = x ( n)
n 0
N
X( )=
2
(1)
n 0
n
x ( n) N=even
10.Parseval’s Relation
N 1 N 1
2 2
N
x ( n) X (k )
n 0 k 0
N 1
Proof: LHS N x ( n) x * ( n)
n 0
1
N 1 N 1
= N x(n) X *
( k )WNnk
m 0 N k 0
N 1
N 1
= X
k 0
*
( k ) x ( n)W Nn k
n 0
N 1 N 1 2
= X
k 0
*
( k ) X (k) =
k 0
X (k )
x((n)) N x( N n) X ((k )) N X ( N k )
Reversing the Npoint seq in time is equivalent to reversing the DFT values.
x( N
n
DFT n)e N
n 0
Let m=Nn
N 1 j 2k
( N m )
= x ( m)e
n 0
N
m=1 to N = 0 to N1
N 1 j 2k
x ( m) e
m
N
=
m 0
63
N 1 j 2m
( N k )
= x(m)e
m 0
N
= X(Nk)
N 1 j 2k
n 0
l 1 j 2k N 1 j 2k
x(n l )
n
x(n l )
n
N
= N e + N e N
n 0 n l
N 1 j 2k
j 2k
x ( N n l )e
l 1 n
x( N n l )
n N
N
= e +
n 0 n l
Put N+nl = m
= x ( m) e
m N l
N
+
m N
x(m)e N
N 1 2k
j ( m l )
Therefore
x ( m) e
m 0
N
2k 2k
N 1 j j l
x ( m) e
m
N N
=
m 0
e
2k
j l
N
= X(k) e RHS
13.Circular Frequency Shift
64
2l
j n
x ( n )e N
X (k l ) N
2l N 1 2l 2k
n j
x ( n)e
j j n n
N N
DFT x ( n ) e N
= e
n 0
N 1 2n
j ( k l )
= x ( n )e
n 0
N
= X (k l ) N RHS
14.x(n) X(k)
k
{x(n), x(n), x(n)…….x(n)} M X ( )
m
(mfold replication)
n
x( ) { X (k ), X (k ),...... X (k )} (M fold replication)
m
2, 3, 2, 1 8, j2, 0, j2
Zero interpolated by M
{2, 3, 2, 1, 2, 3, 2, 1, 2, 3, 2, 1} {24, 0, 0, j6, 0, 0, 0, 0, 0, j6, 0, 0}
15.Duality
x(n) X(k)
X(n) N x(Nk) 0 K N 1
N 1 2
1
j n
x(n) =
X ( )e N
N 0
N 1 2
1 ( N k )
j
x(Nk) =
X ( )e N
N 0
N 1 2
1 j
k
=
X ( )e N
N 0
65
N 1 2
j
k
N x(Nk) =
X ( )e N
0
N 1 2k
j
X ( n )e
n
N
= = DFT [ X(n) ] LHS proved
n 0
16.Re[x(n)]
X ep (k ) 1
X ep (k ) = X ((k )) N X * ((k )) N
2
j Im[x(n)]
X op (k )
xep (n) Re[X(k)]
xop (n) j Im[X(k)]
xep (n) Even part of periodic sequence = 1 x(n) x((n)) N
2
Proof: X(k) = x
n 0
( n )W nk
N
N 1
X(Nk) = x(n)W
n 0
nk
N X ((k )) N
N 1
X (k) =
* x *
( n)W N nk
n 0
N 1
*
X (Nk) = x
n 0
*
( n )WN
nk
X *
((k )) N
66
X ((k )) N X * ((k )) N 1 N 1
2
x(n) x * (n) WNnk
2 n 0
= DFT of [Re[x (n)]] LHS
N 1 N 1
17. x
n 0
1 ( n) x ( n) = *
2
1 X
N k 0
1 ( k ) X 2* ( k )
Y(k) =
1
N
X 1 (k ) X 2* (k )
X
N 1
1
= 1 (l ) X 2* (k l )
N l 0
X
N 1
1
Y(0) = 1 (l ) X 2* (l )
N l 0
Y(0) =
x 1 (n) x 2* (n)
n 0
N 1 N 1
Therefore X
*
x ( n) x 1 2 ( n) =
1
1 (k ) X 2* ( k )
n 0 N k 0
QUESTIONS
1 Q. (i) {1,0,0,…….0} (impulse) {1,1,1…..1} (constant)
(ii) {1,1,1,……1} (constant) ) {N,0,0,…….0} (impulse)
j 2k
N 1 n
jN2k
1 ( j 2k)
NN
1
e
N
(iii)
n
j 2k
k 0 1 N
1 e N
2nko
(iv) Cos
N
(k k o ) (k ( N k o )
N 2
(Impulse pair)
67
Sol. x(n) =
2
j 2nko j 2n ( N k o )
e N
e N
=
2
We know that 1 N (k )
j 2nKo
x ( n) e N
X ( K Ko)
x(n)
N
(k k o ) (k ( N k o )
2
I. Inverse DFT of a constant is a unit sample.
II. DFT of a constant is a unit sample.
2 Q. Find 10 point IDFT of
X(k) = 3 k=0
=1 1k9
3 Q. Suppose that we are given a program to find the DFT of a complexvalued sequence
x(n). How can this program be used to find the inverse DFT of X(k)?
N 1
X(k) = x
n 0
( n )W nk
N
68
N 1
x(n) =
1
N k 0
X ( k )W nk
N
N 1
N x*(n) = X
k 0
*
( k )W nk
N
1. Conjugate the DFT coefficients X(k) to produce the sequence X*(k).
2. Use the program to fing DFT of a sequence X*(k).
3. Conjugate the result obtained in step 2 and divide by N.
4 Q. xp(n) = { 1
, 2, 3, 4, 5, 0, 0, 0}
1 j 2k jk
x ( n)e
n
8 4
X(k) =
n 0
=1+ e k = 0 to 7
X(0) = 1+1 = 2
j
4
X(1) = 1+ e = 1.707  j 0.707
j
2
X(2) = 1+ e = 1 j
j 3
4
X(3) = 1+ e = 0.293  j 0.707
X(4) = 11 = 0
By conjugate symmetry X(k) = X*(Nk) = X*(8k)
X(5) = X*(3) = 0.293 + j 0.707
69
H(k) =
1
X (k ) X (k ) = 1 [ 24, j16, 0, j16] = {6, j4, 0, j4}
4 4
(vi) c(n) = x(n) x(n)
= {1, 2, 1, 0} {1, 2, 1, 0} = {2,4,6,4}
C(k) = X(k)X(k) = {16, 4, 0, 4}
(vii) s(n) = x(n) x(n) = {1, 4, 6, 4, 1, 0, 0}
S(k) = X(k) X(k) = {16, 2.35 j 10.28, 2.18 + j 1.05, 0.02 + j 0.03, 0.02  j 0.03, 2.18 
j 1.05, 2.35 + j 10.28}
(viii) x(n) 1 4 1 0 6
2
70
1 1
[16 4 4] 6
2
X ( k )
4 4
71
X(k) = x(n)W
n 0
nk
N 0 K N 1
N 1
nk nk
= { Re[x(n)] + j Im[x(n)] } { Re( WN ) + j Im( WN ) }
n 0
N 1 N 1
=
n 0
Re[x(n)] Re( W )  nk
N n 0
Im[x(n)] Im( WN ) +
nk
N 1
n 0
2
Direct evaluation of X(k) requires N complex multiplications and N(N1) complex
additions.
4 N2 real multiplications
{ 4(N1) + 2} N = N(4N2) real additions
The direct evaluation of DFT is basically inefficient because it does not use the symmetry
N
K
WNK N WNnk
nk
& periodicity properties WN 2
WN &
DITFFT:
N N
1 1
2 2
X(k) = x(2n)W
n 0
2 nk
N + x(2n 1)W
n 0
( 2 n 1) k
N
(even) (odd)
N
N
1 1
2
x
2
= x (n)W e
2 nk
N + WN
K
o (n)WN2 nk
n 0 n 0
N N
1 1
2
x
2
=
x (n)W e
nk
N /2 W
+ N
K
o (n)WNnk/ 2
n 0 n 0
K
= Xe(k) + W N Xo(k)
72
Although k=0 to N1, each of the sums are computed only for k=0 to N/2 1, since Xe(k)
& Xo(k) are periodic in k with period N/2
N
K K
For K N/2 WN 2
= W N
X(k) for K N/2
N
K
2
X(k) = Xe(kN/2)  W N Xo(kN/2)
N=8
x(2n) = xe(n) ; x(2n+1) = xo(n)
xe(0) = x(0) xo(0) = x(1)
xe(1) = x(2) xo(1) = x(3)
xe(2) = x(4) xo(2) = x(5)
xe(3) = x(6) xo(3) = x(7)
73
N N
This pt DFT can be expressed as combination of pt DFT.
2 4
2k N
Xe(k) = Xee(k) + WN Xeo (k ) k = 0 to 1 (0 to 1)
4
N
2( k ) N
N W 4
Xeo (k ) N N
= Xee(k ) N k= to 1 ( 2 to 3 )
4 4 4 2
74
0
Xe(0) = Xee(0) + W8 Xeo(0) ; xee(0) = xe(0) = x(0)
2
Xe(1) = Xee(1) + W8 Xeo(1) ; xee(1) = xe(1) = x(2)
0
Xe(2) = Xee(0)  W8 Xeo(0) ; xeo(2) = xe(2) = x(4)
2
Xe(3) = Xee(1)  W8 Xeo(1) ; xeo(3) = xe(3) = x(6)
Where Xee(k) is the 2 point DFT of even no. of x e(n) & Xeo(k) is the 2 point DFT of odd
no. of xe(n)
Similarly, the sequence xo(n) can be divided in to even & odd numbered sequences as
xoe(0) = xo(0) = x(1)
xoe(1) = xo(2) = x(5)
xoo(0) = xo(1) = x(3)
xoo(1) = xo(3) = x(7)
75
76
77
= 8 * 3 =24
Each stage no. of butterflies in the stage= 2mq where q = stage no. and N=2m
Each butterfly operates on one pair of samples and involves two complex additions and
one complex multiplication. No. of butterflies in each stage N/2
DITFFT: ( different representation) (u can follow any one) ( both representations are
correct)
N
N
1 1
2
x(2n 1)W
2
x(2n)W 2 nk ( 2 n 1) k
X(k) = N + N
n 0 n 0
N N
1 1
2 2
= x (n)W
n 0
e
nk
N + W Nk x
n 0
o (n)WNnk/ 2
2
K
4 pt DFT Xe(k) + W N Xo(k) k= 0 to N/2 1 = 0 to 3
N
N (K ) N
Xe(k )  W N 2 Xo(k ) k = N/2 to N1 = 4 to 7
2 2
2 pt DFT Xe(k) = Xee(k) + WN2 K Xeo(k) k = 0 to N/41 = 0 to 1
N
2( k )
= Xee(kN/4)  WN 4
Xeo(kN/4) k = N/4 to N/2 1 = 2 to 3
Xo(k) = Xoe(k) + WN2 K Xoo(k) k = 0 to N/41 = 0 to 1
N
2( k )
= Xoe(kN/4)  WN 4
Xoo(kN/4) k = N/4 to N/2 1 = 2 to 3
W82 W41
N=8
X(0) = Xe(0) + W80 Xo(0) ; X(4) = Xe(0)  W80 Xo(0)
X(1) = Xe(1) + W81 Xo(1) ; X(5) = Xe(1)  W81 Xo(1)
X(2) = Xe(2) + W82 Xo(2) ; X(6) = Xe(2)  W82 Xo(2)
X(3) = Xe(3) + W83 Xo(3) ; X(7) = Xe(3)  W83 Xo(3)
78
=
4
Xee(k) = x(4n)W
4 nk 4 nk
N
x(4n)W N =x(0) + x(4) W8
4k
0 n 0
Xee(0) = x(0)+x(4)
Xee(1) = x(0)x(4)
79
80
DIFFFT:
N
1
2 N 1
X(k) = x(n)W
n 0
nk
N + x ( n ' )W n 'k
N put n’ = n+N/2
n1 N / 2
N
1 N
1
2
= x( n)W
2
x(n N / 2)W
nk ( n N / 2) k
N + N
n 0 n 0
N N
1 1
2 N 2
x(n)W x(n N / 2)W
nk k nk
2
= N + W N N
n 0 n 0
N
1
2
= [ x
n 0
( n ) N W nk
+ (1) x(n+ )] N
2
k
N
1
2
N
1
2
N n nk
X(2k+1) = {[ x ( n )  x(n+ )] W N } W N / 2
n 0 2
Let f(n) = x(n) + x(n+N/2)
n
g(n) = { x(n) – x(n+N/2) } W N
81
N=8
f(0) = x(0) + x(4)
f(1) = x(1) + x(5)
f(2) = x(2) + x(6)
f(3) = x(3) + x(7)
82
N
1
4
=
nk
X(4k)
[ f (n) + f(n+ N )] W N / 4
n 0 4
N
1
4
N
1
4
X(4k+1) = [ g ( n
n 0
) + g(n+
N
4
W nk
)] N / 4
N
1
4 nk
X(4k+3) = [{g ( n)  g(n+
nk
N / 2 ] WN / 4
N W
)}
n 0 4
83
84
UNITIV
DIGITAL FILTER STRUCTURE
The difference equation
NP
a k x(nk) + bk
M
y(n) = y(nk)
k N F k 1
NP
a k z k Np NF
(1 C k ) Z 1
k N F
M
Z NF M
1 bk z (1 d
H(z) = k or = A
k 1 k ) Z 1
k 1 k 1
85
1 a1 Z 1 a2 Z 2
H(z) =
1 b1 Z 1 b2 Z 2
Ex:
86
z 5 5 1 Z 2 z 5 1 5Z 2 1
Z 1 Z Z 3
3 12 12 12 3 4 4 4
H(z) = Z 1
Z 2 = Z 1 Z 2
1 1
2 4 2 4
z 1 1
1 Z 1 Z 1 Z 2 1 2
3 4 1 1 1 1 Z Z
= Z 1 Z 2 = z 1 Z Z 1 Z 2
1 3 4 1
2 4 2 4
Ex:
Z Z 1
Z 2 Z
1 Z 2
Z 3
H(z) = = Z 1 Z 1 Z 1
Z 1 Z 1 Z 1
1 1 1 1 1 1
2 4 8 2 4 8
Z
0.65 0.45Z 1
Z 2 1.45 Z 1
= Z 1 Z 1 Z 1
1 1 1
2 4 8
87
z 5 5 1 Z 2
1 5 1 5 2 Z 3
Z Z Z Z
3 12 12 12 3 12 12 12
H(z) = Z 1
Z 2 = Z 1 Z 2
1 1
2 4 2 4
Z 1 Z 2 Z 3 5 2 1 5 1 Z 1 7
1 Z Z
2 4 12 12 3 12 3 3
Z 3 Z 2 Z 1
12 6 3
_______+___________
7 Z 2 Z 1 1
12 12 3
7 Z 2 7 Z 1 7
12 6 3
______________________
5 1 7
Z
4 3
7 5 1
1 Z
Z 3 4
H(z) = Z 2 2
3 1 Z
1 Z
2 4
88
Ex:
Z Z
Z 2 1
H(z) = obtain PSOS
Z 1 Z 1 Z 1
1 1 1
2 4 8
1 Z 1 2Z
1
A
1
B C
Z Z
1
Z = Z 1 Z 1
1 1
Z 1
1 1 1 1 1 1
2 4 8 2 4 8
A = 8/3 B = 10 C = 35/3
D(z) = b Z
N i
i = bo ZN +b1 ZN1 + b2 ZN2 +….. bN1 Z1 + bN
i 0
ROWS COEFFICIENTS
1 bo b1 ……. bN
2 bN bN1 ……. bo
3 Co C1 ……. CN1
4 CN1 CN2 ……. Co
5 do d1 ……. dN2
6 dN2 dN3 ……. do
89
.
.
.
2N3 r0 r1 r2
bo b N i
Ci = bN bi i = 0,1,…N1
co c N 1i
di = c N 1 ci i = 0,1,…N2
i. D(1) > 0
ii. (1)N D(1) > 0
iii. bo bN co c N 1 d o d N 2 ro r2
Ex:
Z4
D(z) = 4Z 3Z 2Z Z 1
4 3 2
H(z) =
4Z 4 3Z 3 2Z 2 Z 1
1 4 3 2 1 1
2 1 1 2 3 4
3 15 11 6 1
4 1 6 11 15
5 224 159 79
D(1) = 4+3+2+1+1 = 11 > 0, (1)4 D(1) = 3 >0
bo b4 co c3 do d2 Stable.
Ex:
1 4Z 2
H(z) = = Ans: Unstable
7 1
1 Z 1 Z 2 4Z 2 7 Z 2
4 2
90
UNITV
Non Recursive filters Recursive filters
Np M
y(n) =
k
ak x(nk) y(n) = ak x(nk) – bk y(nk)
k Nf k 1
For causal i/p sequence It gives IIR o/p but not always.
N
Ex: y(n) = x(n) – x(n3) + y(n1)
y(n) =
k 0
ak x(nk)
NP
91
less severe
9. used in multirate DSP (variable
sampling rate)
IIR FILTER DESIGN
Butterworth, chebyshev & elliptic techniques.
Impulse invariance and bilinear transformation methods are used for translating s
plane singularities of analog filter to zplane.
Frequency transformations are employed to convert LP digital filter design into HP,
BP and BR digital filters.
All pass filters are employed to alter only the phase response of IIR digital filter to
approximate a linear phase response over the pass band.
The system function = H(s)
The frequency transfer function = H(j ) = H(s) / s=j
The power transfer function = H ( j) = H(j ) H*(j ) = H(s) H(s) / s=j
2
To obtain the stable system, the polse that lie in the left half of the splane are assigned to
H(s).
BUTTERWORTH FILTER DESIGN
1
The butterworth LP filter of order N is defined as HB(s) HB(s) = 2N
s
1
j c
Where s = j c
1
H B ( j c ) = H B ( j c ) db = 3dB ‘s
2
or
2
It has 2N poles
2N
s
1 =0
j c
2N
s
= 1
c
j
92
j
j
= e (e c )
2N j 2N
j
e j 2 m
2N
2
= c e e 2
1 N 2 m
2N j
2N
S = c e
1 N 2 m
j
2N
Sm = c e 0 m 2N 1
1
Vo ( s ) CS 1
Vi ( s ) 1 1 RCS
R
CS
1 1 1
c
RS s 2
1
c 1
c
Poles that are let half plane are belongs to desired system function.
1
H B ( j) =
2
2N
1
c
For a large , magnitude response decreases as N, indicating the LP nature of this
filter.
93
2N
= 10 log10(1 )
c
As
= 20 N log10
= 20 N dB/ Decade = 6 N dB/Octane
As N increases, the magnitude response approaches that of ideal LP filter.
The value of N is determined by Pass & stop band specifications.
log(10 6 1)
=N> s
2 log( )
c
Since c is not given, a guess must be made.
94
The specifications call for a drop of 59dB, In the frequency range from the edge of the
pass band (1404 ) to the edge of stop band (8268 ). The frequency difference is equal to
s 10 2N
> c => 1470.3 > c
c <1470.3
Let c =1470.3
At this c it should satisfy pass band specifications.
2N
p
H B ( jp) = [ 1
2
]1 > 0.794 (= 1dB)
c
= 0.59
This result is below the pass band specifications. Hence N=4 is not sufficient.
Let N=5
6
c < s X 10 2N
= 2076.8
10
1404 1
In the pass band H B ( jp) = [ 1
2
] = 0.98
2076
Since N=5
c = 2076
95
S1 = 2076
j144
S2, 3 = 2076 (cos (4 /5) j sin(4 /5)) = 2076 e
j108
S4, 5 = 2076 (cos (3 /5) j sin(3 /5)) = 2076 e
2076 5
HB(s) = s 2076 s 2 3359s (2076 ) 2 s 2 1283s (2076 ) 2
d 2
96
N
3. The phase response curve approaches for large , where N is the no. of poles of
2
butterworth circle in the left side of splane.
Advantages:
1. easiest to design
2. used because of smoothness of magnitude response .
Disadvantage:
Relatively large transition range between the pass band and stop band.
Other procedure
Avo
When c = 1 Avs = 2N
w
1
wo
2 Avo
H B (s) = 2N
s
1
j
If N is odd
S2n =1 = e j 2k
Sk = e j 2k / N k=0,1,2….(2N1)
1
T(s) = ( N 1) / 2
where k =k
N
( s 2 2Cos k s 1)
k 1
97
k 1
1
2n
1
10 log = K1 10 10 1
1 2 N c
1
c
k 2
2
2n
1
10 log = K2 10 10
1
2 2 N c
1
c
k1
1 10 1
2n
Dividing k 2
10
2
10 10 1
10k1
10 1
log 10 k 2
10 10 1
n=
1
2 log 10
2
choosing this value for n, results in two different selections for c . If we wish to satisfy
our requirement at 1 exactly and do better than our req. at 2 , we use
1 2
c = 1
or c = 1
for better req at 2
k1
2n
k2
2n
10 10
1 10 10
1
End
CHEBYSHEV FILTER DESIGN
98
2 2 S
1
Defined as Hc(S) Hc(S) = 1 C N
jp
Two features of Chebyshev poly are important for the filter design
1. C N (x) 1 for x 1
1
2 1
H c ( j) 2 1 for 0 p
99
2. x 1, C N (n) Increases as the Nth power of x. this indicates that for >> p , the
magnitude response decreases as N, or 6N dB Octane. This is identical to Butterworth
filter.
Now the ellipse is defined by major & minor axis.
Define = 1 1 2
N1 1
N
Minor r = p
2
N1 1
N
Major R = p N = Order of filter.
2
SP = r Cos +j R Sin
Ex:
Pass band:
1< H ( j) dB 0
2
for 0 1404
Stop band:
H ( j) dB < 60
2
for 8268
10 log 1 2
1
> 1dB 1dB = 0.794
= 0.508
100
1
s 10 6 1 2
Since 5.9 CN(5.9) > = 1969
p
2
Evaluating
C3(5.9) = 804 C4(5.9) = 9416 therefore N = 4 is sufficient.
Since this last inequality is easily satisfied with N=4 the value of can be reduced to as
small as 0.11, to decrease pass band ripple while satisfying the stop band. The value =0.4
provides a margin in both the pass band and stop band. We proceed with the design with
=0.508 to show the 1dB ripple in the pass band.
Axes of Ellipse:
=0.5081 + (1+0.5082)1/2 = 4.17
1404 1 1
R= 4.17 4.17 702 (1.43 0.67) 1494
4 4
2
1404
1 1
Chebyshev filter poles are closer to the j axis, therefore filter response exhibits a
ripple in the pass band. There is a peak in the pass band for each pole in the filter, located
approximately at the ordinate value of the pole.
101
Exhibits a smaller transition region to reach the desired attenuation in the stop band,
when compared to Butterworth filter.
Phase response is similar.
Because of proximity of Chebyshev filter poles to j axis, small errors in their
locations, caused by numerical round off in the computations, can results in significant
changes in the magnitude response. Choosing the smaller value of will provide some
margin for keeping the ripples within the pass band specification. However, too small a
value for may require an increase in the filter order.
It is reasonable to expect that if relevant zeros were included in the system function, a
lower order filter can be found to satisfy the specification. These relevant zeros could serve
to achieve additional attenuation in the stop band. The elliptic filter does exactly this.
IMPULSE INVARIANCE METHOD
H(z) = h( n) Z
n 0
n
e ST ) = h(n )e
STn
H(z) (at z =
102
If the real part is same, imaginary part is differ by integral multiple of 2 , this is the
T
biggest disadvantage of Impulse Invariance method.
sa sa
Let HA(S) = =
s a b
2 2
s a jb s a jb
hA(t) = e atCosbt for t 0 s1 = ajb
=0 otherwise s2 = a+jb
h (nTs) = e anTsCos(bnTs) for n 0
1 e aTsCos (bTs) Z 1 1 e aTsCos(bTs) Z 1
H(z) = =
1 2e aTsCos (bTs) Z 1 Z 2 (1 e ( a jb )Ts Z 1 )(1 e ( a jb )Ts Z 1 )
The pole located at s=p is transformed into a pole in the Zplane at Z = e pTs , however, the
finite zero located in the splane at s= a was not converted into a zero in the zplane at Z =
e aTs , although the zero at s= was placed at z=0.
103
k
=
(1 1.69Z 0.743Z )(1 1.707 Z 1 0.88Z 2 )
1 2
1 Z 1
S=
Ts
Or
Using forwarddifference mapping based on first order approximation Z = e sTs 1+STs
Z 1
S=
Ts
Using backward difference mapping is based on first order approximation
Z 1 e sTs 1 STs
Z 1 1 Z 1
S =
ZTs Ts
d 2x d dx
/t=nTs = / t nTs
dt 2
dt dt
104
2
1 2Z 1 Z 2 1 Z 1
S 2
=
Ts 2 Ts
k
1 Z 1
k
Therefore S =
T
1 Z 1
Therefore H(z) = Ha(s) /s= using backward difference
T
1 0.5(1 STs)
Z= = 0.5 +
1 STs 1 STs
1 1 jTs
= =
1 jTs 1 Ts 1 2Ts 2
2 2
0.5(1 STs)
Z  0.5 =
(1 STs)
Using Forwarddifference
Z 1
S= Z=1+STs
Ts
u+jv = 1+ ( j) Ts
if =0 u=1 and j axis maps to Z=1
If >0, then u>1, the RHSplane maps to right of z=1.
If <0, then u<1, the LHSplane maps to left of z=1.
105
Bilinear Transformation
Provides a non linear one to one mapping of the frequency points on the jw axis in s
plane to those on the unit circle in the zplane.
This procedure also allows us to implement digital HP filters from their analog
counter parts.
2 Z 1 2 1 Z 1
S= =
Ts Z 1 Ts 1 Z 1
{Using trapezoidal rule y(n)=y(n1)+0.5Ts[x(n)+x(n1)]
H(Z)=2(Z1) / [Ts(Z+1)] }
2 Z 1
To find H(z), each occurrence of S in HA(s) is replaced by
Ts Z 1
STs
1
And Z = 2
STs
1
2
1/ 2
2 Ts 2 j tan 1
Ts
Ts 1 e 2
j 1 2
e jw 2
Ts = 1/ 2
j 1 2 Ts
2
j tan 1
Ts
2
2
1 e
2
Ts
j 2 tan 1 Ts
e jw
e 2
w=2tan1
2
106
The entire j axis in the splane  <j < maps exactly once onto the unit circle 
w such that there is a one to one correspondence between the continuoustime and
discrete time frequency points. It is this one to one mapping that allows analog HPF to be
implemented in digital filter form.
As in the impulse invariance method, the left half of splane maps on to the inside of the
unit circle in the zplane and the right half of splane maps onto the outside.
2 w
In Inverse relationship is tan
Ts 2
w w3
Sin
w
2 8 ....
For smaller value of frequency
2 2 = 2 w
2
Ts w Ts w Ts
Cos 1 ....
2 4
107
The mapping is linear for small and w. For larger freq values, the non linear
compression that occurs in the mapping of to w is more apparent. This compression
causes the transfer function at the high freq to be highly distorted when it is translated to
the wdomain.
Prewarping Procedure:
When the desired magnitude response is piece wise constant over frequency, this
compression can be compensated by introducing a suitable prescaling or prewarping to the
freq scale. scale is converted into * scale.
2 Ts
* = tan
Ts 2
We now derive the rule by which the poles are mapped from the splane to the zplane.
1
Let HA(s) = S=Sp
S Sp
1 Ts(1 Z 1 )
H(z) = =
2 1 Z 1
Sp 2 SpTs1 2 SpTs Z 1
Ts 1 Z 1 2 SpTs
2 SpTs
A pole at S=Sp in the splane gets mapped into a zero at z= 1 and a pole at Z =
2 SpTs
Ex:
Chebyshev LPF design using the Bilinear Transformation
Pass band:
1< H ( j) dB 0 for 0 1404 =4411 rad
Stop band:
H ( j) dB < 60 for 8268 rad/sec =25975 rad/s
2 Ts 4
And s* = tan = 2*10 tan(0.4134 ) = 71690 rad/sec
Ts 2
108
Pass band:
1< H ( j*) dB 0 for 0 * 4484 rad/s
Stop band:
H ( j*) dB < 60 for * 71690rad/sec
= 0.508
H c js * = 1 2C N2 <106
2
p *
s *
Since 16
p *
CN2(16)
10 6 1
<
2
1
10 6 1 2
CN(16) < 2
= 1969
(0.508)
C3(16) = 16301
N = 3 is sufficient
Using Impulse Invariance method a value of N=4 was required.
=4.17
N1 1
N
4484
1
1
4484
1 1
3
r= 4.17 3
4.17 2216
2
2
Since there are three poles, the angles are &
3
S1 = r cos + j Rsin = 2216
2 2
S2,3 = 2216 Cos j 5001 Sin = 1108 j 4331 = 4470 e j104.4
3 3
109
4.43 *1010
Hc(s) =
( s 2216)( S 2 2223s 4470 2 )
Pole Mapping
At S=S1
2 (2216 *10 4 )
In the Zplane there is zero at Z = 1 and pole at Z = 0.801
2 (2216 *10 4 )
1 Z 1 1 2Z 1 Z 2
H(z) = 4.29 * 103
1 0.801Z 1 1 1.638Z 1 0.81Z 2
Pole Mapping Rules:
Hz(z) = 1CZ1 zero at Z=C and pole at Z = 0
1
Hp(z) = pole ar Z=d and zero at z=0
1 dZ 1
C and d can be complexvalued number.
Pole Mapping for LowPass to Low Pass Filters
Applying low pass to low pass transformation to Hz(z) we get
c 1
1 Z
Z
1
1 c
HLZ(Z) = 1c = (1+c )
1 Z 1 1 Z 1
c
The low pass zero at z=c is transformed into a zero at z=C1 where C1 =
1 c
And pole at z=0 is Z=
Similarly,
1 Z 1
HLP(Z)=
1 d 1 d Z 1
1 d
d
Pole at z=d => Z=
1 d
Zero at z=0 => z =
1 Z 1 2Z 2Z
1 1 2
H(z) = K
1 0.622Z 1 1.07Z 0.674Z
1 1 2
110
K=
1 (1)(0.356)3 0.029
(1 0.801* 0.356)(1 (0.819 j 0.373)(0.356))(1 (0.819 j 0.373)(0.356))
111
UNITVI
( )
Phase Delay: p
d ()
Group Delay: g
d
If p = g =constant and independent of frequency are called as constant time delay or
linear phase filters.
() o
o
p Changes with frequency
g =  =constant.
Type 1 Sequence
N 1
Center of Symmetry M= integer value
2
N 3
jnT
2
N 1
T
2
Amplitude spectrum is even symmetric about w=0 & & both H(0) & H( ) can be non
zero.
Type 2 Sequence
112
h(n) = h(N1n)
N 1
Center of Symmetry M= halfinteger value
2
N2 1
jMT
H(w) = 2 h(n)CosT (n m) e
n 0
The Amplitude spectrum is even symmetric about w=0 & odd symmetric about w= &
both H( ) is always zero for type 1 & 2 : Constant phase delay and group delay.
Type 3 Sequence
N 1
M= integer value
2
N23
2 h(n) SinT ( M n) e jMT
H(w) = j
n 0
113
It shows generalized linear phase of MT and constant group delay of M. The
2
Amplitude spectrum is odd symmetric about w=0 & w= and H(0) & H( ) are always zero.
(Generalized means () may jump of at 0 if H(ejw) is imaginary.
Type 4 Sequence
N2 1
2 h(n) SinT ( M n) e jMT
H(w) = j
n 0
Generalized linear phase and constant group delay of M. The Amplitude spectrum is odd
symmetric about w=0 & even symmetric about w= and H(0)=0 always.
For N=even, even Symmetry h(n) = h(N1n)
N 1
) = h ( n )e
jnT
jT
H( e
n 0
N
1 N 1
+
2
= h( n)e
jnT jnT
h( n)e
n 0 N
n
2
Let N1n = m
N
1
2 0
= h( n)e jnT
+ h ( N 1 m) e jT ( N 1 m )
n 0 m
N
1
2
114
N N
1 1
2 2
== h( n)e
n 0
jnT
+ h ( m) e
m 0
jT ( N 1 m )
N
1 N 1 N
1
2 jT N 1
h ( n )e jnT 2 jT
+ h ( m) e
2 jT ( N 1 n ) 2
= e e
n 0 n 0
N
1 N 1 j ( nT 2T ( N 1) j ( T ( N 1 n )
T
( N 1)
2 jT
e e 2
= 2 h ( n )e
n 0
2
2
N
1 N 1
2 jT N 1
= 2 h ( n )e cos T n
2
n 0 2
N
N 1 1
jT 2 N 1
=e
2
2h(n) cos T n Magnitude
2
n 0
N 1
T Linear Phase
2
Poles & Zeros of linear phase sequences:
The poles of any finitelength sequence must lie at z=0. The zeros of linear phase
sequence must occur in conjugate reciprocal pairs. Real zeros at z=1 or z=1 need not be
paired (they form their own reciprocals), but all other real zeros must be paired with their
reciprocals. Complex zeros on the unit circle must be paired with their conjugate (that form
their reciprocals) and complex zeros anywhere else must occur in conjugate reciprocal
quadruples. To identify the type of sequence from its polezero plot, all we need to do is
check for the presence of zeros at z= and count their number. A type2 seq must have an
odd number of zeros at z=1, a type3 seq must have an odd number of zeros at z=1 and
z=1, and type4 seq must have an odd number of zeros at z=1. The no. of other zeros if
present (at z=1 for type=1 and type2 or z=1 for type1 or type4) must be even.
115
FIR Filters
Fs Fs
Fourier series Method F
2 2
2Fs 2Fs
2F
2 2
s s
2 2
1. Frequency response of a discretetime filter is a periodi function with period s
(sampling freq).
2. From the F.S analysis we know that any periodic function can be expressed as a linear
combination of complex exponentials.
Therefore desired freqency response of a discrete time filter can be represented by F.S as
116
H (e jT
) h( n)e
n
jnT
T = sampling period
The F.S coefficient or impulse response samples of filter can be obtained using
1 s / 2
h (n) =
s s / 2
H (e jT )e jnT d
clearly if we wish to realize this filter with impulse response h(n), then it must have finite
no. of coefficient, which is equivalent to truncating the infinite expansion of H (e jT ) , which
leads to approximation of H (e jT ) , which is denoted by
m
H 1 (e jT
) h (
n m
n ) e jnT
.
N 1
We choose M= , in order to keep ‘N’ no of samples in h(n).
2
M
H1(z) = h( n) Z
n M
n
However, this filter can’t be physically realizable due to the presence of +ve powers of Z,
means that the filter must produce an output that is advanced in time with respect to the i/p.
N 1
this difficulty can be overcome by introducing a delay M= samples.
2
M
Therefore M
H(z) = Z H1(z) = Z M
h( n) Z
n M
n
H(z) = b Z
i 0
i
i
be the transfer function of discrete filter that is physically realizable.
Properties:
1. N=2M+1, impulse response coeff, bi = 0 to 2M.
2. h(n) is symmetric about bM
Ex: M=4
117
H (e jT ) H1 (e jT )
This implies that magnitude response of the filter we have desired approximates the
desire magnitude response. The time delay of H(ejw) is a constant M. thus sinusoids of
different frequencies are delayed by the same amount as they are processed by the filter, we
have designed. Consequently, this is a linear phase filter, which means that it does not
introduce phase distortion.
Ex:
Design a LPF (FIR) filter with frequency response
H (e jT ) 1 for c
s
= 0 for c
2
1 c jnT
s c
h(n) = e d
2 c
s 0
= Cos (nT )d
2 SincnT
=
s nT
118
2 1
SincnT SincnT
= 1 n
2Fsn.
Fs
bi = h(iM)
2M
H(z) =
i
bZ i
i 0
s 2Fs 1
w= T T
2 2 Fs
Ex:
Design LPF that approximate following freq response.
H(F) = 1 0 F 1000Hz
=0 else where 1000 F Fs/2
When the sampling frequency is 8000 SPS. The impulse response duration is to be
limited to 2.5ms
Ti = 2MT
2.5 *10 3
M= 10 N=21
1
2*
800
119
1 c jnT
s c
h(n) = 1.e d
1 Fc j 2FnT 1 Fc j 2FnT 2 Fc
2Fs Fc Fs Fc Fs 0
= 1.e 2dF = e dF Cos(2FnT )dF
1 1
= Sin 2FcnT Sin (0.25n )
n n
________________________________________________________________
OR
1
w = T = 2 *1000 *
8000 4
Hc(w) = 1 w
4
= 0 else where
1
1.e dw = 1 Sin (0.25n )
4 jwn
2
4 n
h(0) = 0.25 h(6) = 0.05305
h(1) = 0.22508 h(7) = 0.03215
h(2) = 0.15915 h(8) = 0
h(3) = 0.07503 h(9) = 0.02501
h(4) = 0 h(10) = 0.03183
h(5) = 0.04502
bi = h(i10)
20
H(z) = b Z
i 0
i
i
FIR HPF
1 c
1.e jnT d e jnT d
s / 2
h(n) =
s s / 2 c
120
1 e jnT c e jnT s / 2
s jnT
= s / 2 c
jnT
jcnT s
j nT
s
j nT
1 e e 2
e 2 e jcnT
= s jnT
jcnT jcnT
s
j nT j nT
s
2 1 e e e 2
e 2
= s nT
2j 2j
2 s nT
=
2FsnT Sin c nT Sin 2
1
=
1
sin c nT Sinn = sin c nT
n n
FIR BPF
2 u 1
sin u nT sin l nT
s l
h(n) = cos nT d =
n
Ex:
Desing a BPF for H(f) = 1 160 F 200Hz
=0 else where
Fs = 800SPS
Ti = 20 ms
Ti 20 *10 3
M= 8 N = 17
2T 1
2*
800
121
H(z) = b Z
i 0
i
i
The abrupt truncation of infinite series is equivalent to multiplying it with the rectangular
sequence.
WR(n) = 1 n M
=0 else where
h (n) h(n)WR (n)
H (e jw ) H (e jw ) *WR (e jw )
1
H (e
j
= )WR (e j ( w ) )d
2
122
4
Main lobe width = & it can be reduced by increasing N, but area of side lobe will
N
be constant.
For larger value of N, transition region can be reduced, but we will find overshoots &
undershoots on pass band and non zero response in stop band because of larger side lobes.
So there overshoots and leakage will not change significantly when rectangular window is
used. This result is known as Gibbs Phenomenon.
The desined window chts are
1. Small width of main lobe of the fre response of the window containing as much as of
the total energy as possible.
2. Side lobes of the frequency response that decrease in energy as w tends to .
3. even function about n=0
N 1
4. zero in the range n
2
123
Let us consider the effect of tapering the rectangular window sequence linearly from the
middle to the ends.
Triangular Window:
2n N 1
WT (n) 1 n
N 1 2
=0 else where
In this side lobe level is smaller that that of rectangular window, being reduced from 13
8
to 25dB to the maximum. However, the main lobe width is now . There is trade off
N
between main lobe width and side levels.
General raised cosine window is
2n N 1
W(n) = (1 )Cos for n
N 1 2
=0 else where
If =0.5 Hanning Window
If =0.54 Hamming Window
2n 4n
WB(n) = 0.42 + 0.5 Cos 0.08Cos Blackman Window
N 1 N 1
Kaiser Window
2n
2
Io 1
N 1
Wk (n) for n
N 1
Io( ) 2
=0 else where
is constant that specifies a freq response trade off between the peak height of the side
lobe ripples and the width or energy of main lobe and Io(x) is the zeroth order modified
Bessel function of the first kind. Io(x) can be computed from its power series expansion
given by
2
1 x k
Io(x) = 1+ k! 2
k 1
124
=1+
(1!) 2 + (2 !) 2 + (3 !) 2 +…..
If we let K1,W1 and K2,W2 represent cutoff (pass band) * stop band requirements for the
digital filter, we can use the following steps in design procedure.
1. Select the window type from table to be the one highest up one list such that the stop
band gain exceeds K2.
2. Select no. of points in the windows function to satisfy the transition width for the type
2
of window used. If Wt is the transition width, we must have Wt = W2W1 k .
N
where K depends on type of window used.
K=1 for rectangular , k=2 triangular…..
2
Therefore N K
w2 w1
125
=0 else where
N = 2M+1 = 21
WH(0) = 1 WH(6) = 0.39785
WH(1) = 0.97749 WH(7) = 0.26962
WH(2) = 0.91215 WH(8) = 0.16785
WH(3) = 0.81038 WH(9) = 0.10251
WH(4) = 0.68215 WH(10) = 0.08
WH(5) = 0.54
Next these window sequence values are multipled with coefficients h(n), obtained in ex1,
to ontain modified F.S Co eff h’(n).
h’(0) =0.25
h’(1) =0.22
h’(2) =0.14517
h’(3) =0.0608
h’(4) =0
h’(5) =0.02431
h’(6) =0.02111
h’(7) =0.0086725
h’(8) =0
h’(9) =0.00256
h’(10) =0.00255
126
2M
H’(z) = b' Z
i 0
i
i
Ex:
Find a suitable window and calculate the required order the filter to design a LP digital
filter to be used A/DH(Z)D/A structure that will have a 3dB cutoff of at 30 rad/sec and
an attenuation of 50dB at 45 rad/sec. the system will use a sampling rate of 100 samples
/sec
Sol:
The desired equivalent digital specifications are obtained as
1
Digital ….. w1 wc cT 30 0.3 k1 3dB
100
1
w2 2T 45 0.45 k 2 50dB
100
1. to obtain a stop band attenuation of 50dB or more a Hamming window is shosen
since it has the smallest transition band.
2. the approximate no. of points needed to satisfy the transition band requirement (or the
order of the filter ) can be found for w1 =0.3 rad &w2 = 0.45 rad, using Hamming
window (k=2), to be
2 2.2
N k =26.65
w2 w1 0.45 0.3
N = 27 is selected
127
the attractive property of the Kaiser window is that the side lobe level and main lobe
width can be varied continuously by simple varying the parameter . Also as in other
window, the main lobe width can be adjusted by varying N.
we can find out the order of Kaiser window, N and the Kaiser parameters to design
FIR filter with a pass band ripple equal to or less that Ap, a minimum stop band attenuation
equal to or greater than As, and a transition width Wt, using the following steps:
Step 1 :
10 0.05 Ap 1
s 10 0.05 As
, p 0.05 Ap
10 1
1 p 1 p
Ap = 20 log 10 As = 20 log 10 => As = 20 log s
1 p s
1
100.05Ap =
1
1 100.05Ap = 1
Therefore: solving above eq for , we get
10 0.05Ap  1
10 0.05Ap 1
Step 2:
Calculate As using the shosen values
Aso= 20 log
128
Step 3:
Calculate the parameter as follows for
=0 for Aso 21 dB
= 0.5842(Aso 21)0.4 + 0.07886(Aso 21) for 21< Aso 50 dB
= 0.1102(Aso 8.7) for Aso >50 dB
Step 4:
Calculate D as follows
D = 0.9222 for Aso 21 dB
As 7.95
= for Aso >21 dB
14.36
Step 5:
Select the lowest odd value of N satisfying the inequality
samD
N 1
t
Wsam : Angular Sampling frequency
sam : Analog Freq
t = s p for LPF
= p s for HPF
129
c =
1
p s for LPF & HPF
2
t t
c1 = p1 ; c2 p2 for BPF
2 2
t t
c1 = p1 ; c2 p2 for BSF
2 2
Ex:
Calculate the Kaiser parameter and the no. of points in Kaiser window to satisfy the
following lowpass specifications.
Pass band ripple in the freq range 0 to 1.5 rad/sec 0.1 dB
Minimum stop band attenuation in 2.5 to 5.0 rad /s 40 dB
Sampling frequency : 10 rad/s
Sol:
130
D = 2.566
Step 5:
10(2.566)
N 1 26.66 => N=27
1
2n
2
Io 1
N 1
Wk (n)
Io( )
Io( )
Wk (0) 1
Io( )
2
2
Io 3.9524 1
26
Wk (n) Io(3.94)
10.269
0.9899
=
Io(3.9524) Io(3.9524) 10.3729
131
OBJECTIVE PAPER1
Match the following: For a real valued sequence, the DTFT follow the properties as
9) Re [H (jw) ] a) Real valued function of w
10) Im[ H(jw) ] b) even function of w
11) F.T [even symmetric sequence] c) Imaginary valued function of w
12) F.T [odd symmetric sequence] d) odd function of w
13) x(n) = {4, 1, 3} h(n) = {2, 5, 0, 4} what is the output of the system.
a) {8, 22, 11, 31, 4, 12} b) {8, 22, 11, 31, 4, 12} c) {8, 22, 11, 31, 4, 12}
d) none
14) y(n) = x(n) * h(n) then y1 (n) = {0, 0, x(n), 0 } * { 0, h(n), 0 } is equal to
a) {0, 0, y(n), 0} b) {0, 0, 0, y(n), 0, 0} c) [0, 0, y(n), 0 } d) {0, y(n), 0, 0}
15)If x(n) and h(n) are having N values each, to obtain linear convolution using circular
convolution, the number of zeros to be appended to each sequence is
a) N – 1 b) 2N – 1 c) N d) N + 1
16)W49 = ?
a) – j b) + j c) + 1 d) 1
132
OBJECTIVE PAPER2
5) Consider the system shown in fig. The transfer function Y(Z) / X(Z) of the system is
x(n) y(n)
+ +
Z1
b a
1 1 1 1
a) (1+aZ )/ ( 1+bZ ) b) (1+bZ )/ ( 1+aZ )
c) (1+aZ1)/ ( 1bZ1) d) (1bZ1)/ ( 1+aZ1)
6) A linear discrete time system has the characteristic equation Z30.8 Z=0, the system
a) is stable b) is marginally stable
c) is un stable d) stability cannot be assessed from the given information
akx(n k )
5
8) y(n) =  k 1
b k(yn k ) the minimum no of delay elements
k 2
needed to realize the system is
a) 5 b) 10 c) 8 d) 11
10) To ensure a causal system, the total no of zeros must be less than or equal to the total
number of poles ( T / F )
133
1 2 3 4 5 6 7 8 9 10
a a c c a a a c T
11) The poles or zeros at the origin do effect the magnitude response ( T / F)
12) All poles and zeros of a minimum phase system lie inside the unit circle ( T / F)
14) Find total no of complex multiplications using FFT for N=8: __________
15) Find total no of complex additions using FFT for N=8: __________
16) Find total no of real additions using direct DFT for N=8: __________
11 12 13 14 15 16
F T a 12 24 240
134
OBJECTIVE PAPER3
State TRUE or FALSE
1) u(n) = (n k )
K 0
1 2 3 4 5 6 7 8 9 10 11 12
T F T F 3 1 4 2 B B A a
13) The output of anti causal LTI system is
n
a) y (n) = h( k ) x ( n k )
K 0
b) y (n) = h( k ) x ( n k )
K 0
1
c) y (n) = h( k ) x ( n k )
d) y (n) = h( k ) x ( n k )
135
18) Given g(n) = {1, 2, 3} , find x(n) = g (n / 2), using linear interpolation
19)
h1(n) + h3(n) y(n)
x(n)
h2(n)
In the figure shown, how do you replace whole system with single block
a) [ h1(n) + h2(n) ] * h3(n) b) h1(n)h3(n) * h2(n)h3(n)
c) [ h1(n) + h2(n) ] h3(n) d) none
20 The h(n) is periodic with period N, x(n) is non periodic with M samples, the output
y(n) is
a) Periodic with period N b) Periodic with period N+M
c) Periodic with period M d) none
13 14 15 16 17 18 19 20
C A B B C C A A
136
OBJECTIVE PAPER4
a) 6 b) 10 c) 0 d) none
2) If x(n) = 1, n≤2
0, other wise
Find DTFT
a) sin(5w)/sinw b) sin(4w)/sinw c) sin(2.5w)/sin(0.5w) d) none of the above
a) {8, 22, 11, 31, 4, 12} b) {8, 22, 11, 31, 4, 12} c) {8, 22, 11, 31, 4, 12} d)
none
7) If x(n) and h(n) are having N values each, to obtain linear convolution using circular
convolution, the number of zeros to be appended to each sequence is
a) N – 1 b) 2N – 1 c) N d) N + 1
8)W49 = ?
a) – j b) + j c) + 1 d) 1
9) DFT [ x* (n) ] = ?
a) X * (K) b) X * (K) c) X * (NK) d) none
11) Both discrete and periodic in one domain are also periodic and discrete in other
domain (T / F)
137
13) Reversing the N point sequence in time is equivalent to reversing the DFT values (T /
F)
OBJECTIVE PAPER5
1. The Fourier transform of a finite energy discrete time signal, x(n) is defined as [ ]
n
x(n) e x(n) e
jn
a) X( )= b) X( )=
n= n=
x(n) e x(n) e
jn jn
c) X( )= d) X() =
n= n=0
N 1 j 2kn N 1 j 2kn
a) x(n) =
1
N
n 0
X(k) e N b) a) x(n) =
1
N
n 0
X(k) e N
j 2kn N j 2kn
c) x(n) =
1
N
n 0
X(k) e k d) a) x(n) =
1
N
n 0
X(k) e N
3. A N – periodic sequence x(n) and its DFT x(k) are known. Then the DFT of x(n) =
(n) will be
a) ej2nk b) 1 c) ej2nok/N d) ej2nk /N [ ]
4. If the length of sequence x(n) is L and h(n) is M then the length of o/p sequence of the
circular convolution is [ ]
a) L+M b) L+M1 c) L if L>M d) 2L if L=M
138
7. The circular shift of an N point sequence is equivalent to linear shift of its periodic
extension [ ]
8. The multiplication of DFT of two sequences is equal to DFT of the linear convolution
of two sequences [ ]
15.To get the result of linear convolution with circular convolution of sequence x(n) &
h(n), the sequences should extended to the length of __________________
e) x1(n) x2*(n)
139
1= 2= 3= 4=
17. Show that the given sequence x(n) = { 1,2,3,2,1,0} for the following conditions
using concentric circles.
a) x(n) b) x(2n) (2M)
OBJECTIVE PAPER6
MULTIPLE CHOICES
1. In Impulse invariant transformation, the mapping of analog frequency to the digital
frequency is
a) one to one b) many to one c) one to many d) none
2. The digital frequency in bilinear transformation is
a) w = 2 tan1(Ts/2) b) w = tan1(Ts/2)
c) w = 2 tan1(Ts) d) w = 2 tan1(/2)
3. Which technique is useful for designing analog LPF
a) Butter worth filter b) Chebyshev filter
c) Both a and b d) none
4. Which filter is more stable?
a) Butter worth b) Chebyshev c) none
5. As increases , the magnitude response of LPF approaches with
a) –20Ndb/oct b) –6Ndb/oct c) –10Ndb/dec d) none
6. Using Impulse invariant technique the pole at S= SP is mapped to Zplane as
a) Z=eSPTs b) Z=e (SPTs) c) Z=eSP (Ts) d) None
TRUE or FALSE
7. The disadvantage of Chebyeshev filter is less transition region
8. The advantage of Butter worth filter is flat magnitude response.
9. for the given same specifications order of the Chebyshev filter is more than
Butterworth filter
10. Poles of Butterworth filter lies on circle.
1 2 3 4 5 6 7 8 9 10
B A C A B B F T F T
140
17.Given allowable ripples in Pass band is –3 dB, the value of is 0.997 (2M)
OBJECTIVE PAPER7
Choose the correct Answer
c) = 2 tan ( T )
1
d) = 2 tan1( /2)
3. Using bilenear transformation for T = 1sec the pole p k is in S Plane is mapped to Z –
plane using [ ]
1 z 1 1 z 1 1 z 1 Z 1
a) S = 2 b) S = c) S = 2 d) S=
1 z 1 1 z 1 1 z 1 Z 1
2 1 1
c) H a () = d) H a () =
1 C N ( / c )
2 2
1 C N ( / c )
2
141
10.In …………………. Window spectrum the width of main lobe is double that of
rectangular window for same value of N [ ]
a) Hamming window b) Kaiser window c) Blackman window d) none
15.For same specifications, the order N of chebyshev filter is less compared to Butter
worth filter. [ ]
20.In FIR filter with constant phase delay the impulse response is symmetric
[ ]
142
OBJECTIVE PAPER8
8. If M & N are the lengths of x(n) & h(n) then length of x(n) * h(n) is [ ]
a) M+ N –1 b) M + N +1 c) max (M,N) d) min (M,N)
11.In a discrete signal x(n), if x(n) =x(n) then it is called symmetric signal [ ]
12.The F.T of the product of two time domain sequence is equivalent to product
of their F.T [ ]
13.The DFT of a signal can be obtained by sampling one period of FT of the signal
[ ]
14.DFS is same as DTFS [ ]
143
OBJECTIVE PAPER9
CHOOSE THE CORRECT ANSWER
1. Power signal is
a) Periodic b) aperiodic c) Continuous d) none [ ]
2. WN nK is
j 2 K j 2 Kn 2 Kn
j 2 nK
a) e N
b) e c) e N
d) e N
[ ]
3. When the sequence is circularly shifted in time domain by ‘m’ samples i.e. x((nm))N
then on applying DFT, it is equivalent multiply sequence in frequency domain by
j 2 Km j 2 Km 2 Km
j 2 Km
a) e N
b) e N
c) e d) e N
[ ]
K 0
10.Appending zeros to a sequence in order to increase the size or length of the sequence
is called ……………………..
11.In Npoint DFT using radix 2 FFT, the decimation is performed …………… times.
12.In 8point DFT by radix 2 FFT, there are …………… stages of computations with
…………………….. butterflies per stage.
144
15.How many multiplications and additions are required to compute Npoint DFT using
radix 2 FFT
16.How many multiplications and additions are required to compute Npoint DFT
145
OBJECTIVE PAPER10
1. How we can calculate IDFT using FFT algorithm. (2M)
5. Find the ZTransform and ROC for the signal x(n) = an u(n).
6. Find the ZTransform and ROC for the signal x(n) =  an u(n1).
8. Z{(n)} = ……………………..
Z
9. Find inverse ZTransform for X(z) = when ROC is Z<1
Z 1
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10.What are the differences and similarities between DIT and DIF algorithms. (2M)
17.Direct form I required less no.of memory elements as compared to Canonic form.[ ]
18.A linear time invariant system with a system function H(Z) is BIBO stable if and only
if the ROC for H(Z) contains unit circle. [ ]
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OBJECTIVE PAPER11
2. What is the relation between analog and digital radiant frequency in Impulse
Invariance design..
3. What is the relation between analog and digital radiant frequency in Bilinear
transformation design.
7. Mention any two techniques to design IIR Filter from analog filter.
8. What are the differences between Chebyshev type I and type II.
148
TRUE OR FALSE
12.Poles of Butterworth filter lies on circle. [ ]
17.Chebyshev, type II filter exhibit equiripple behavior in the pass band and monotonic
characteristic in the stopband. [ ]
18.Chebyshev, type I filter exhibit equiripple behavior in the pass band and monotonic
characteristic in the stopband. [ ]
20.For the given specifications order of the Chebyshev filter is more as compared to
Butterworth filter. [ ]
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OBJECTIVE PAPER12
c. Dynamic system
d. Recursive System
150
8. Find the Convolution Sum Graphically with all the steps3 Marks
x(n)= 2 1 h(n)= 1 1
1 0 0 1
11. Write the necessary condition for the stability of the system
151
OBJECTIVE PAPER13
State TRUE or FALSE
1. In direct –form II realization the number of memory locations required is more than
that of direct form –I realization [ ]
2. An LTI system having system function H(z) is stable if and only if all poles of H(z)
are out side the unit circle. [ ]
5. As the order of Butter worth filter increases than the response is closer to ideal filter
response. [ ]
Answer the following
7. Indicate the poles and zeros of the given system and also check the stability of the
system
z ( z 1)
H(z) = (2M)
( z 0.2)( z 0.4)( z 0.5)
8. Realize the given system function H(z) using direct form –II
3 3.6 z 1 0.6 z 2
H(z) = (2M)
1 0.1z 1 0.2 z 2
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9. Realize the given system function H(z) using cascade form (2M)
1
H(z) =
(1 0.5 z )(1 0.5 z 1 )
1
z
10.Find the inverse ztransform of x(z) = using partial fraction method.
( z 2)( z 3)
(2M)
14.The order of the Butter Worth filter is obtained by using the formula N
______________________
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OBJECTIVE PAPER14
3. The number of multiplications needed in the calculation of DFT using FFT with 32
point sequence = ________________
6. For DIT –FFT algorithm the input is bit reversed and the output is in natural order
[ ]
8. WNNK 1 [ ]
9. WNNK/ 2 1 [ ]
10.In DIT –FFT, the input sequence is divided into smaller subsequences [ ]
11.Calculate the DFT of the sequence x(n)={1,0,0,1} using DIT –FFT (2M)
12.Draw the Butterfly diagram for 8point DFT using DIT –FFT algorithm (2M)
154
14. Write the steps for the calculation of IDFT using DIT –FFT (2M)
155
11. The average power of a discrete time signal with period N is given by ___________
12. The convolution sum of causal system with causal sequence is ____________
156
17. If the impulse response h(n) = 2n u( n) then determine the corresponding system is
causal or stable. (1M)
18. Test the given discrete system for linearity , causality and time invariance
h(n) = n ex(n) ( 2M)
ASSIGNMENT UNIT5
1 (a) Draw the frequency response of Npoint rectangular window.
(b) Design a fifth order band pass linear phase filter for the following specifications.
i. Lower cutoff frequency = 0.4 πrad/sec
ii. Upper cutoff frequency = 0.6 πrad/sec
iii. Window type = Hamming
Draw the filter structure. [4+12]
2) Design a band pass filter to pass frequencies in the range 12 radians/second using
Hanning window N=5. Draw the filter structure and plot its spectrum. [16]
3) (a) Compare the performances of rectangular window, hamming window and Keiser
window
(b) The desired response of a low pass filter is
Hd(ej!) = _ e−j3!, −3π _ ω _ 3π/4
0 , 3π/4 _ ω _ π
Determine H(ej!) for M=7 using a Hamming window. [6+10]
4) (a) Design a linear phase low pass filter with a cutoff frequency of π/2
radians/seconds. Take N=7
(b) Derive the magnitude and phase functions of Finite Impulse Response filter when
i. impulse response is symmetric & N is odd
ii. impulse response is symmetric & N is even. [8+8]
5) (a) Design a low pass filter by the Fourier series method for a seven stage with cutoff
frequency at 300 Hz if ts = 1msec. Use hanning window.
(b) Explain in detail, the linear phase response and frequency response properties of
Finite Impulse Response filters. [8+8]
6) (a) Outline the steps involved in the design of FIR filter using windows.
(b) Determine the frequency response of FIR filter defined by y(n) = 0.25x(n)+ x(n1)+
0.25x(n2). Calculate the phase delay and group delay. [8+8]
7) (a) Define Infinite Impulse Response & Finite Impulse Response filters and compare.
157
(b) Design a low pass Finite Impulse Response filter with a rectangular window for a
five stage filter given: Sampling time 1 msec; fc = 200Hz.Draw the filter structure with
minimum number of multipliers. [6+10]
ASSIGNMENT UNIT7
1) a) What are the advantages of Multirate signal processing?
b) Differentiate between Decimator and Interpolator?
2) Prove that spectrum of down sampler is sum of M uniformly shifted and stretched
version of X(ejw) scaled by a factor 1/M and also discuss the aliasing effect?
3) State and prove any one identity property in down sampler and any one identity
property in up sampler?
4) Let x(n)={1,3,2,5,1,2,2,3,2,1},find
a) Up sample by 2 times and down sample by 4 times
b) Down sample by 4 times and up sample by 2 times c) Justify why these outputs are
not equal.
158