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Based on Variables:
1.
2.
3.
4.
5.
S1(t )
S4(t)= S 2(t ) : Multichannel signal
S 3(t )
Ir ( x, y, t )
I(x,y,t)= Ig ( x, y, t ) multidimensional
Ib( x, y, t )
Based on Representation:
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III.
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Based on duration.
1.
2.
3.
4.
5.
IV.
1.
n 0
=1
n=0
2. u (n) =1
n0
=0
n<0
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x(n) =
x(k ) (n k )
k
n
u(n) =
(n k )
k 0
Rect (n/2N) =1
=0
else where.
5.Tri (n/N) = 1- n /N
=0
n N
else where.
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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))
1
t
as t .
Power signal: E = , P 0, P
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)
4.
Half-wave symmetry.
xe(n)=
1
[x(n)+x(-n)]
2
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xo(n)=
1
[x(n)-x(-n)]
2
b. Semi-infinite extent: right sided, if they are zero for t < where = finite
1
x(t )
T 0
dt
1
2
x(t ) dt
P = Lt
To 0
To
Xavg = Lt
x(t )dt
0
x(t) = A cos( 2 fo t + )
x(t) = A e j( 2 fo t + )
P=0.5 A2
P=A2
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E= A2 b
E=
1 2
A b
2
E=
1 2
A b
3
Q.
e - t dt =
Q.
Ex =
1 2
1
A 0.5T + (-A)2 0.5T = 0.5 A2 T
2
2
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Px = 0.5 A2
Q.
1
3
Ey = [ A2 0.5T] 2 =
Py =
1 2
A T
3
1 2
A
3
1
T
x(t )
dt = A2
y (t) = 2 x [-
(t 2)
t 2
] = 2 x[ ]
3
3 3
; = 2/3
Area of symmetric signals over symmetric limits (- , )
Odd symmetry:
x0 (t) dt =0
Even symmetry:
xe (t) dt = 2
xe (t) dt
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1
[x(n)+x(-n)]
2
Xo (n) =
1
[x (n)-x (-n)]
2
Area of any sinc or Sinc 2 equals area of triangle ABC inscribed within the main lobe.
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Even though the sinc function is square integrable ( an energy signal) , it is not
absolutely integrable( because it does not decay to zero faster than
1
)
t
(t) = 0 t 0
( )d
= t=0
=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 .
[ [t- ]] =
(t )
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
x1(t) = x(t)
(t-kts ) =
x(kts) (t-kts)
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(t) =0
t 0
= undefined
t=0
' (t )dt 0
[ [t- ]] =
(t )
' (t )
With =-1
(-t) = - (t)
d
[ x(t ) (t )] = x (t) (t- ) + x (t) (t- )
dt
1=2
x ( ) (t- ) + x (t) (t- ) = x ( ) (t- )
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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) = e-t (t) describe?
x(t) = (t) (-1) (t) = (t) + (t)
2
= 0.5 (t-3) t 1 - 8
d
[cos t ] t 0.5
dt
= 23.1327 Answer.
Operation on Signals:
1. Shifting.
x(n) shift right or delay = x(n-m)
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(-n-2) is x(-n) advanced by two samples.
Or
x(n) is right shift x(n-2), then fold x(-n-2)
x(n) fold x(-n) shift left x(-(n+2)) = x(-n-2)
Ex:
x(n) = 2, 3 , 4 , 5, 6, 7 .
Find 1. y(n)=x(n-3) 2. x(n+2) 3. x(-n) 4. x(-n+1) 5. x(-n-2)
1.
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2.
3.
4.
5.
3. a. Decimation.
Suppose x(n) corresponds to an analog signal x(t) sampled at intervals Ts. The signal
y(n) = x(2n) then corresponds to the compressed signal x(2t) sampled at Ts and contains
only alternate samples of x(n)( corresponding to x(0), x(2), x(4)). We can also obtain
directly from x(t) (not in compressed version). If we sample it at intervals 2Ts (or at a
sampling rate Fs =
1
). This means a two fold reduction in the sampling rate.
2Ts
Decimation
{ 1 , 6, 8}
n 2n
Step interpolation
{ 1 , 2, 6, 4, 8}
Step interpolation
n n/2
{ 1 , 1, 6, 6, 8, 8}
Decimation
{ 1 , 1,2,2,6, 6,4,4,8, 8}
n n/2
n 2n
{ 1 , 2, 6, 4, 8}
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}
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= { 1, ,
5
2 1
, 2 , 3,4,5,3,1,-1, - ,- } Linear interpolation.
3
3 3
4. Fractional Delays.
It requires interpolation (N), shift (M) and Decimation (n): x (n x(n) = {2, 4, 6 , 8}, find y(n)=x(n-0.5) = x (
M
( Nn M )
)=x(
)
N
N
2n 1
)
2
h(n) =g(n-1) = x(
n 1
) = {2, 2, 4, 4 , 6, 6,8,8}
2
2n 1
) = {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(n-1) 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 (n-1) and i/p at the same value of
n or previous value of n.
Ex. y(n) = x(n) + 3 x(n-1)
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x(n k )
y(n) = 1/ (m1+m2+1)
k m1
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
( Linear , Unstable )
y(n) = x(k )
k
n 1
x(k ) + x(n)
= y(n-1) + x(n)
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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/ps till nth sample
Ex:
x(n) = n u(n) ; given y(-1)=0. i.e. initially relaxed.
y(n) =
k 0
x(k ) + x(k )
n
= y(-1) +
x(k ) = 0 +
k 0
n =
k 0
n(n 1)
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)]
TIV
TV
7. Causality.
A system is causal if for every choice of no the o/p sequence value at index n= no
depends only on the input sequence values for n no.
y(n) = x(n) + x(n-1) causal.
y(n) = x(n) + x(n+2) + x(n-4) non causal.
8. Stability.
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For every bounded input x(n) Bx < for all n, there exists a fixed +ve finite value
By such that y(n) By < .
PROPERTIES OF LTI SYSTEM.
x(k ) (n k )
1. x(n) =
Therefore o/p of any LTI system is convolution of i/p and impulse response.
y(no) =
h(k ) x(no k )
k 0
n<0
h( k ) x ( n k )
k 0
h( k ) x ( n k )
k 0
h2(n)
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y (n) =
h( k ) x ( n k )
h(k ) x(n k )
h(k )
S=
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h( k ) x ( n k )
y (n+N) =
h( k ) x ( n k N )
put n-k = m
h(n m) x(m N )
h(n m) x(m)
m=k
h( n k ) x ( k )
= y(n) (Ans)
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Y ( z)
=1-0.5 Z-1
X ( z)
System
Inverse System
y (n) 0.5 y(n-1) =x(n)
Y (z) [1-0.5 Z-1] = X (z)
Y ( z)
-1 -1
[1-0.5 Z ]
X ( z)
Recursive filters
y(n) =
ak x(n-k)
y(n) =
bk y(n-k)
k 1
ak x(n-k)
k 0
ak x(n-k)
k 0
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y(n) =
always.
ak x(n-k)
k 0
1
[x (n+1) + x (n) + x (n-1)]
3
1
[ (n+1) + (n) + (n-1)]
3
h(0) =
1
3
1
3
h(-1) =
h(1) =
1
3
S= h(n) <
therefore Stable.
1
[ y(n) x(n)]
a
1
[ y(n+1) x(n+1)]
a
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y(-1) =
1
[ y(0) x(0)]=0
a
y(-2) = 0
1
y(n-1) + x(n)
n 1
Q. y(n) =
=0
for n 0
1
h(n-2) + (n-1)
n 1
n=0
n=1
n=2
h(2) = 1/12
h (n, 0) h(n,1)
TV
1
[x (n+1) + x (n) + x (n-1)] Linear
3
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(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(n-1) non linear, TIV
Q. y (n) - 2 y (n-1) = 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 (n-2) = x (n) causal
Q. y (n) - 2 y (n-2) = x (n+1) non causal
Q. y (n+1) y (n) = x (n+2) non causal
Q. y (n-2) = 3 x (n-2) is static or Instantaneous.
Q. y (n) = 3 x (n-2) 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
n-2
n-1
- 4 n-2 =0
[ 2 - 3 - 4] =0
= -1, 4
yh (n) = C1 1n + C2 2n = C1(-1)n + C2 4n
y(0) = 3y(-1) +4 y(-2) = 15
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C1+ C2 =15
1
1 a1
yp (n) =
1
u(n) Ans
1 a1
x(n)
yp(n)
1. A
2. Amn
Kmn
3. Anm
Ko nm + K1nm-1 + . Km
4. A Coswon or A Sinwon
K1 Coswon + K2 Sinwon
Q. y(n) =
5
1
y(n-1) - y(n-2) + x(n)
6
6
x(n) = 2n n 0
1
K 2n-2 u(n-2) + 2n u(n)
6
For n 2
4K =
5
1
(2K) - K +4
6
6
yp (n) =
8 n
2
5
Ans
Q. y(n) 3 y(n-1) - 4 y(n-2) = x(n) + 2x(n-1) Find the h(n) for recursive system.
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y(1) = 3 y(0) +2 = 5
C1 + C2 =1
-C1 + C2 =5 Solving C1 =
h(n) = [
1
6
; C2 =
5
5
1
6
(-1)n + 4n ] u(n) Ans
5
5
OR
h(n) 3 h(n-1) -4 h(n-2) = (n) + 2 (n-1)
h(0) = 1
h(1) =3 h(0) + 2 = 5
plot for h(n) in both the methods are same.
Q. y(n) 0.5 y(n-1) = 5 cos 0.5n n 0 with y(-1) = 4
yh(n) = n
0.5
n
n-1
n-1
=0
[ -0.5] =0
=0.5
yh(n) = C (0.5)n
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= rad / sec
w = rad / Sample
F = cycles / sec
f = cycles / Sample
K
N
)n
6
=2 f
6
1
12
N=12 Samples/Cycle ;
Ts
Sampling Period
Q. Cos (0.5n) is not periodic
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Non-periodic
Q. x (n) = 5 Cos (6 n)
2 f = 6 => f = 3
Q. x (n) = 5 Cos
2 f =
6
35
6 n
35
=> f =
3
35
Periodic
=> f =
Q. x (n) = Cos (3 n)
0.01
2
Periodic
for N=2
Periodic
fo = GCD (f1, f2) & T = LCM (T1, T2) ------- For Analog/digital signal
[Complex exponential and sinusoidal sequences are not necessarily periodic in n
with period (
2
) and depending on Wo, may not be periodic at all]
Wo
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Fs
Fs
F
2
2
- Fs 2 F Fs
-
Ts
Ts
- Ts
Therefore - w
5. Increasing the frequency of a discrete- time sinusoid does not necessarily
decrease the period of the signal.
x1(n) = Cos (
n
)
4
N=8
x2(n) = Cos (
3 n
)
8
N=16
2 f = 3 /8
=> f =
3
16
2 f = 2 F Ts
2 F = 4
1
samples/Sec = Hz then digital frequency f = 1
Ts
Ts
=> f =1
2 f = /4
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F=
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1
; T= 8 ;
8
f=
1
8
N=8
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
f1 =
F2 = 30Hz
20 4 K1
75 15 N1
f2 =
30 2 K 2
75 5 N 2
1
0.1s
Fo
Since N=15
1sample ----------
1
sec
75
15 sample ----------- ?
=>
15
0.2S
75
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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 discrete-time 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)
= h( k ) x ( n k )
k
jwn
=e
h(k ) e-jwk
jw
Let H (e ) =
y (n) = H (e ) e
jw
jwn
Q. Find the frequency response of 1st order system y (n) = x (n) + a y (n-1)
(a<1)
Let x (n) = ejwn
yp (n) = C ejwn
C ejwn = ejwn + a C ejw (n-1)
C ejwn [1-ae-jw] = ejwn
C=
1
[1 ae jw ]
Therefore H (ejw) =
H (e jw ) =
1
1
=
[1 ae jw ] 1 a(cos w j sin w)
1
1 2a cos w a 2
aSinw
H (e jw ) Tan 1 (
)
1 aCosw
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1
y (n 2)
2
x (n) = e jwn
(n) ce jwn
c e jwn = e jwn 1
2
1 jw( n 2)
ce
2
c=
1
1
1 e 2 jw
2
20 16Cos2 w
5 4Cos2 w
Sin 2 w
c tan1
2 Cos2 w
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UNIT - II
Discrete Time
Periodic f (t) =
N 1
jkot
Periodic
xp(n) =
c e
k 0
jK
2
n
N
DTFS
Non periodic
1
Ck = T
Periodic Ck =
f (t )e jKot dt
1
N
jK
1
x ( n)e
NTs
2nTs
T
N 1
x
n 0
( n )e
2
nK
N
k=0 to N-1
T = N Ts
t = n Ts : dt = Ts
Non Periodic x(n) =
Non-Periodic f(t) =
1
2
F (w)e
jt
1
2
X (w)e
jwn
dw
Non-Periodic F(w) =
f (t )e jt dt
Periodic X(w) =
x(n)e
jwn
X(w) = FT of DTS
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E=
1
*
x
(
n
)
x
(
n
)
x
(
n
)
x
(
n
)
=
2
n
n
n
Therefore:
jwn
X
(
w
)
x
(
n
)
e
dw
0
n
( w) X ( w)dw
2
X ( w)
dw
X (w)
N 1
N
1
Lt
x ( n)
N 2 N 1
n N
1
x
(
n
)
E=
2
n
1
N
1
= 2
*
jwn
X
w
e
dw
(
)
1
=
2
1
=
2
x ( n)
n 0
N 1
1 N 1
1 N 1
* j N nk
*
x ( n) x ( n) x ( n) C k e
N n 0
N n 0
k 0
* 1
Ck
k 0
N
N 1
Ex:
P=
C
k 0
x ( n)e
2
nk
N
n 0
N 1
Therefore
N 1
N 1
E=N
Ck
k 0
Unit step
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N
1
u 2 ( n)
P = Lt
N 2 N 1 n 0
=
E=
Ex:
N 1 1
Lt
2
N 2 N 1
Power Signal
x (n) =
Ae
jwon
P=
1
jwon
Ae
Lt
N 2 N 1 n N
1
A 2 [1 1 ........]
= Lt
N 2 N 1
=
A 2 (2 N 1)
2
A
Lt 2 N 1
N
Ex:
Ex:
x (n) = n u(n)
x (n) = 3 (0.5)n
E=
Ex:
P=
n 0
x (n) = 6 Cos
E=
n0
9
12 J
1 0.25
whose period is N=4
note: [
n 0
1
]
1
x (n) = { 6 ,0,6,0 }
1 3 2
1
x
(
n
)
4 n 0
4
j
Ex:
2n
4
and
x (n) 9(0.25)n
2
it is Power Signal
x (n) = 6 e
2n
4
whose period is N = 4
P=
1 3
1
x
(
n
)
4 n 0
4
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DISCRETE CONVOLUTION
y(n) =
u(k )u(n k )
u(k) = 0
k<0
u(n-k) = 0
k>n
u(k )u(n k )
k 0
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y(n) =
k 0
y(n) =
u(k) u(n-k) =
k 0
= (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.
Q. x(n) = rect (
n
)=1
2N
=0
else where
h(n) = rect ( n )
2N
y(n)
= x(n) * h(n)
= [u (n+N) u (n-N-1)] * [u (n+N) u (n-N-1)]
n
)
2N 1
n
Tri ( n ) = 1-
for n N
=0
elsewhere.
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Q. x(n) = {2,-1,3}
h(n) = { 1,2,2,3} Graphically Fold-shift-multiply-sum
y(n) =
1
-1
-1
-2
-2
-3
y(n) = { 2,3,5,10,3,9}
,3}
Q. x(n) = {4, 1
h(n) = { 2,5, 0
,4}
20
16
15
12
y(n) = { 8,22,11,31,4, 12 }
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Q)
h(n):
x(n):
_________________________________
8
20
16
15
12
____________________________________
y(n):
22
11
31 4
12
, 5, 0, 4 ; x(n)= 4 , 1, 3
Q. Convolution by sliding step method: h(n) = 2
i)
2 5 0 4
ii)
3 1 4
3 1 4
___________________
_________________________
y(0) = 8
iii)
2 20
2 5 0 4
iv)
3 1 4
6 5 0
2 5 0 4
_______________________
y(2) = 11
2 5 0 4
15 0 16
Vi)
3 1 4
y(3) = 31
2 5 0 4
3 1 4
________________________
0 4
y(1) = 2+20 = 22
3 1 4
________________________
v)
2 5 0 4
y(4) = 4
_______________________
12
y(5) = 12
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Q. h(n) = 2 , 5, 0, 4 ; x(n)= 4 , 1, 3
y(n) = { 8,0,22,0,11,0,31,0,4,0,12}
Q. Compute the linear convolution of h(n)={1,2,1} and x(n) = { 1, -1, 2, 1 ,2, -1, 1, 3,
1} using overlap-add and overlap-save method.
h (n):
x (n):
-1
x1(n):
-1
x2(n):
-1
-1
x3(n):
____________________________________________________________
y1(n) = (h (n)*x1(n))1
y2(n) =
y3(n) =
y(n) =
{ 1
-1
8
4
5
8
1
5
1}
(N2=3)
x1(n):
-1
(N3+N2-1) = 5
-1
x2(n):
x3(n):
y1(n) =
y2(n) =
7 3
1 4
1 5
{1
5 2
y3(n) =
y(n) =
8 5 1
8 5 1}
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fo =
b. x (n) = Cos
fo =
1
6
N=6
j
1 5
x
(
n
)
e
Ck =
6 n 0
2
nk
6
k=0 to 5
j k
j k
j k
j k
1
jk
3
3
3
x(3)e x(4)e
x(5)e 3
x(0) x(1)e x(2)e
6
Ck =
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
x (n)
1 j 6 n 1 j
e
= Cos n e
+
2
3
2
2
n
6
C
k 0
2
kn
6
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= Co+C1 e
2
n
6
+C2e
By comparison C1=
j
Since
C5
2
n
6
4
n
6
+ C3 e
6
n
6
+C4 e
8
n
6
+C5 e
10
n
6
1
2
=e
56
j 2
n
6
=e
10n
6
1
2
c. x (n) = {1,1,0,0}
j 2
1 3
x
(
n
)
e
Ck= 4
n 0
c0
1
2
Co
1
&
2
c
c
c1
1
1 j ;
4
c2 0 ;
c3
1
1 j
4
C0 = 0
2
& C1 =
4
4
k=0, 1, 2, 3
2k
j
1
2
1 1e
nk
4
2
4
& C2 undefined
& C3 =
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PROPERTIES OF DFS
1.
DFS
Linearity
~x2 (n) Ck 2
~
~
a
x
(
n
)
b
x2 (n) aCk1 bCk 2
1
DFS
DFS
2.
Time Shifting
DFS x ( n m) e
3.
1
N
DFS
2mk
N
Ck
Symmetry
~x (n) C
*
DFS
N 1
x ( n )e
~
n 0
~
x (n) C
*
Ck =
2n k
N
~
x (n)
N 1
C
k 0
j2
nk
*
~
~
x
(
n
)
x
( n) 1
*
~
Re
x
(
n
)
DFS
C
k C ke
k
DFS
2
*
~
~
x
(
n
)
x
( n) 1
~
DFS
2
2
~
x ( n)
If
is real then
*
~
~
x
(
n
)
x
( n)
~
x e ( n)
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~
x ( n) ~
x * ( n)
~
x o ( n)
1
*
~
x
(
n
)
C
C
k ReC k
e
k
DFS
2
1
*
~
x
(
n
)
C
k
o
k
DFS
2
j ImC k
Periodic Convolution
N 1 ~
~
x
(
m
)
x
(
n
m
)
2
1
C k1C k 2
m 0
DFS
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
H (e
j ( w 2 )
) = H (e jw )
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jw
H (e ) =
[ h
n -
[ h
n -
[ h (n) Cos(wn) h
I
n -
(n)Sin(wn) = HI (e
jw
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 )
H (e ) tan
jw
H
(
e
)
R
1
jw
jw
H (e )
=
=
h(n)e
jwn
h(n)Cos(wn) j h(n)Sin(wn)
H R (e jw ) jH I (e jw ) -------------------- (a)
H (e
jw
=
=
h(n)e
jwn
h(n)Cos(wn) j h(n)Sin(wn)
H R (e jw ) jH I (e jw ) -------------------- (b)
H R (e jw ) H R (e jw )
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H I (e jw ) H I (e jw )
H (e jw ) H * (e jw )
=>
H (e
- jw
jw
H I (e-jw )
)
-1 H I (e
) tan
tan
H (e jw )
- jw
jw
H R (e )
H R (e )
-1
h( n k )e
FT [h (n-k)] =
jwn
Put n-k = m
h(m)e
jw ( m k )
= e
jwk
h(m)e jwm = H (e jw )
e jwk
6. Time Reversal
x (n) X (w)
x (-n) X (-w)
F T [x (-n)] =
x ( n)e
jwn
Put n = m
jwm
x
(
m
)
e
X ( w)
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7. Frequency Shifting
e jwo n
x(n)
X (w-wo)
F T [x (n)
e jwo n
]=
e jwo n e
x (n)
-jwn
e j ( w wo ) n
x (n)
= X (w-wo)
8. a. Convolution
x1 (n) * x2 (n) X1(w) X2(w)
Put n-k = m
x1 (k)
-jwk
x1 (k) e
[x2(m)] e-jwm
= X1(w) X2(w)
b.
1
[X1(w) * X2(w)] x1 (n) x2 (n)
2
9. Parsevals Theorem
1
*
x1(n) x2 (n) =
2
[X1(w) X2*(w)] dw
n x (n) j
dX ( w)
dw
H (e
jw
)=
h (n) e-jwn
47
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n 1
h (n) e-jwn
Let n = -m
jwm
h (-m) e
m 1
+ h (0) +
n 1
h (n) e-jwn
Therefore =
h (0) + 2
n 1
frequency
; H (e jw ) >0
; H (e jw ) <0
H (e jw ) =
n 1
= -j 2
n 1
H (e jw ) = - H (e jw )
H (e jw ) = HI (e jw )
= - HI (e jw )
H (e jw )
12.
1
x(0) = 2
for HI (e jw ) < 0
for HI (e jw ) > 0
HI (e jw ) > 0
HI (e jw ) < 0
X (w)dw
Central Co-ordinates
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X (0) =
13.
x ( n)
Modulation
Cos (won) x (n)
X ( w w0 ) X ( w w0 )
2
2
X (w) =
x (n) e-jwn
x(n)
F T exists if
Ex: h (n) =
=0
for -1 n 1
otherwise
1
Sol:
jw
H (e ) =
1 jwn
e
3
n 1
1 jw
1
e 1 e jw = 1 2Cos ( w)
3
3
w
0
1
3
1
3
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Ex:
JNTU World
h (n) = an u (n)
jw
H (e
) = a
n 0
=
Q.
e jwn
(ae
jw n
) =
n 0
x(n) = n n u(n)
n
n u(n)
<1
1
d
jw
dw 1 e
e jw
(1 e jw ) 2
Hint:
Q.
1
1 ae jw
x(n) =
u(n)
jwn
n 0
(e jw ) n
n 0
1
1 e jw
0nN
Or
x(n) = n [ u(n) u(n-N)]
= n u(n) N n-N u(n-N) Using Shifting Property
1
e jwN
N
[
]
jw
jw
X(w) =
1 e
1 e
=
Q.
Q.
1 (e jw ) N
1 e jw
Ans
x(n) =
x(n) =
1
1
1 =
=
1 e jw 1 e jw
1 2
1 2Cosw 2
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( w)
Similarly X (w) =
1
jw
1
1 e jw
(w)
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x ( n )e
jwn
Since X (w) is periodic with period 2 , sample X(w) periodically with N equidistance
samples with spacing w
2
.
N
K = 0, 1, 2..N-1
2
j
Kn
2k
N
X
x
(
n
)
e
N
The summation can be subdivided into an infinite no. of summations, where each sum
contains
2k
X
............ x(n)e
N
n N
1
2 N 1
x(n)e
2
Kn
N
2
j
Kn
N
N 1
x(n)e
2
Kn
N
n 0
..................
n N
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lN N 1
x(n)e
n
lN
2
Kn
N
Put n = n-lN
x(n lN )e
N 1
x(n lN )e
l
xp(n)
n 0
k 0
Ck
N 1
Therefore
1
N
2
Kn
N
2
Kn
N
N 1
Ck=
n 0
N 1
X(k) =
2
K ( n lN )
N
n 0
N 1
xp(n)
n 0
Ck=
e
j
1
X(k)
N
2
Kn
N
2
Kn
N
n= 0 to N-1
k=0 to N-1
k=0 to N-1
N 1
1
N
k 0
X(k)
2
Kn
N
n = 0 to N-1
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
recovered from its samples at frequency Wk=
Prove: x(n) = xp (n)
2k
.
N
0 n N-1
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Using IDFT
x (n) =
1
N
N 1
X(k) e
n 0
N 1
X (k) [
k 0
2
Kn
N
k 0
N 1
X (w) =
1
N
N 1
X (k) e
2
Kn
N
] e-jwn
k 0
N 1
1
N
jn ( w
2
K)
N
n 0
If we define p(w)
1
N
N 1
e-jwn
n 0
wN
N 1
jw
1 1 e
2
2
=
=
e
w
N 1 e jw
NSin
2
N 1
2k
Therefore: X (w) = X (k) P(w)
N
k 0
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(
)
N
k 0
k 0
jwN
x(n) = an u(n)
Ex:
Sin
0<a<1
2k
. k=0, 1..N-1, determine
N
1
1 ae jw
1
X (wk) =
1 ae
2
k
N
k=0, 1, 2 N-1
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x(n LN )
xp (n) =
= a n a lN
l 0
n lN
l
n
a
1 aN
0 n N-1
=0
0 n N-1
otherwise
N 1
N 1
X ( w) x (n)e jwn
n 0
x
n 0
N 1
an
e jwn
=
N
n 0 1 a
X ( w)
(n)e jwn
1
=
1 aN
N 1
(ae
jw n
n 0
1 1 a N e jwN
1 a N 1 a e jw
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2k
j
N
N
N
a
e
1
1
N
2K
N 1 a 1 a e j N
2K
= X
N
1
=
1 ae
j 2k
N
2k
are identical.
N
1
Ex:
X (w) =
&
1 a e jw
X (k) =
1
1 a e
2
k
N
Apply IDFT
2nk
j
N
1
e
x (n) = N
2k
j
k 0
1 ae N
N 1
=0
1
= N
j 2Nnk r j 2Nkr
e
a e
k 0
r 0
1
=
N
N 1 j 2k ( nNr )
a e
r 0
k
N 1
except
x (n) =
n mN
m 0
r = n+mN
= a
(a
)m
m 0
an
=
1 aN
The result is not equal to x (n), although it approaches x (m) as N becomes .
Ex:
X (k) =
x ( n)e
2k
n
4
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X (0) =
x(n) = 0+1+2+3 = 6
n 0
X (1) =
x ( n )e
2
n
4
= -2+2j
n 0
X (2) = -2
X (3) = -2-2j
DFT as a linear transformation
Let
WN e
2
N
N 1
x(n)W
X (k) =
n 0
x (n) =
Let
WN =
1
N
nk
N
N 1
X (k )W
k 0
x ( 0)
x(1)
xN =
x( N 1)
k = 0 to N-1
nk
N
n = 0, 1N-1
X ( 0)
X (1)
XN =
X ( N 1)
1
1
1
1 W 1
2
W
N
N
1 W N2
W N4
1 W NN 1 W N2 ( N 1)
W N( N 1)
W N2 ( N 1)
( N 1)( N 1)
WN
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DFT
IDFT
1
W N* X N
xN =
N
XN = WN xN
x N WN1 X N
1.
Ex:
1
N
WNK N WNK
1
WN*
N
2.
WN
N
2
WNK
x (n) = {0, 1, 2, 3}
1
1
1 1
1 W 1 W 2 W 3
4
4
4
DFT W4 = 1 W 2 W 4 W 6 =
4
4
4
3
6
9
1 W4 W4 W4
1
1
1 1
1 W 1 W 2 W 3
4
4
4
2
0
1 W4 W4 W42
3
2
1
1 W4 W4 W4
1
1 0
1 1
1 j 1 j 1
X4 = W4 x 4 = 1 1 1 1 2
1 j 1 j 3
1
1
1 1
1 j 1 j
= 1 1 1 1
1 j 1 j
2 2 j
= 2
2 2 j
IDFT
x4
1 *
= WN X N
4
1
1
1 1
1 j 1 j
1
4 1 1 1 1
1 j 1 j
1
2 2 j
2
= 2 Ans
3
2 2 j
Q.
x (n) = { 1,0.5}
h (n) = { 0.5,1 }
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Find y (n) = x (n) h (n) using frequency domain. Since y (n) is periodic with period 2.
Find 2-point DFT of each sequence.
X (0) = 1.5
H (0) = 1.5
X (1) = 0.5
H (1) = -0.5
Y (1) = -0.25
Using IDFT
y (0) = 1;
y (1) = 1.25
~ ~
h (k ) x (n k )
~
y (0)
=
~
~
x
(
k
)
h
(n k )
~
~
(
)
x
k
h
(k )
~
~
~
~
x (0)h (0) x (1)h (1)
= 1 * 0.5 + 0.5 * 1 = 1
~
y (1)
~
~
x
(
k
)
h
(1 k )
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~
~
~
~
x (0)h (1) x (1)h (0)
~
y (2)
~
~
x
(
k
)
h
(2 k )
~
~
~
~
x (0)h (2) x (1)h (1)
= 1 * 0.5 + 0.5 * 1 = 1
~
y ( n)
aliasing we convert the 2-sample input sequence into 3 sample sequence by padding with
zero.
For 3- point DFT
X (0) = 1.5
X (1) = 1+0.5
X (2) = 1+0.5
H (0) = 1.5
2
3
H (1) = 0.5+
4
3
2
3
H (2) = 0.5+
4
3
2
3
4
3
+ 0.5
+ 0.5
4
3
8
3
Compute IDFT
y(n) =
j
1 2
Y ( k )e
3 k 0
2kn
3
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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 )
4. y(n) = x(n-n0)
j
Y (k) = X (k) e
2kn0
N
1
H (k ) X (k )
Y (k) =
N
7. For real valued sequence
N 1
H R (k ) h(n)Cos
n 0
N 1
2kn
N
H I (k ) h(n) Sin
n 0
a.
2kn
N
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i.
N
2
DFT
N
need to be computed in order
2
HR (k) = HR (N-k)
c.
H (k ) H ( N k )
e.
H (k ) H ( N k )
f.
If h(n) = h(-n)
If h(n) = -h(-n)
x(n)W
n 0
N 1
X*(k) =
n 0
nk
N
( n)W N n k
N 1
X (N-k) =
x
n 0
N 1
DFT [x*(n)] =
x
n 0
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9. Central Co-ordinates
1
x (0) =
N
N 1
X (k )
x(
k 0
N 1
1
N
)=
N
2
(1)
X (k ) N=even
k 0
N 1
X (0) =
N 1
x ( n)
N
X( )=
2
n 0
(1)
x ( n)
N=even
n 0
10.Parsevals Relation
N 1
x ( n)
n 0
N 1
X (k )
k 0
N 1
Proof:
N x ( n) x * ( n)
LHS
n 0
1
= N x(n)
m 0
N
N 1
N 1
k 0
N 1
X
k 0
X
k 0
( k )WNnk
N 1
( k ) x ( n)W Nn k
n 0
N 1
N 1
( k ) X (k)
X (k )
k 0
x((n)) N x( N n) X ((k )) N X ( N k )
Reversing the N-point seq in time is equivalent to reversing the DFT values.
DFT
x( N n)
N 1
x( N
n)e
j 2k
n
N
n 0
Let m=N-n
N 1
x ( m)e
m=1 to N = 0 to N-1
n 0
N 1
j 2k
( N m )
N
x ( m) e
j 2k
m
N
m 0
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N 1
x(m)e
j 2m
( N k )
N
m 0
= X(N-k)
x ( n l ) N X ( k )e
N 1
j 2k
l
N
j 2k
n
N
n 0
l 1
x(n l )
n 0
j 2k
n
N
l 1
x( N n l )
N 1
x(n l )
+
j 2k
n
N
n 0
n l
j 2k
n
N
N 1
x ( N n l )e
j 2k
n
N
n l
Put N+n-l = m
N 1
x ( m) e
j 2k
( m l )
N
2 N 1l
m N l
x(m)e
j 2k
( m l )
N
m N
Therefore
x ( m) e
x ( m) e
2k
m
N
m 0
= X(k)
2k
( m l )
N
m 0
N 1
2k
l
N
2k
l
N
RHS
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x ( n )e
2l
n
N
X (k l ) N
2l
j
n
N
x
(
n
)
e
=
DFT
N 1
=
14.x(n)
x ( n )e
N 1
x ( n)e
2l
n
N
2k
n
N
n 0
2n
( k l )
N
X (k l ) N
n 0
RHS
X(k)
k
(m-fold replication)
n
x( ) { X (k ), X (k ),...... X (k )}
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)
0 K N 1
X(n) N x(N-k)
x(n) =
x(N-k) =
N 1
1
N
1
N
X ( )e
2
n
N
1
N
N 1
X ( )e
N 1
2
( N k )
N
X ( )e
2
k
N
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N 1
X ( )e
N x(N-k) =
2
k
N
N 1
X ( n )e
2k
n
N
n 0
16.Re[x(n)]
1
X ep (k ) = X ((k )) N X * ((k )) N
2
X ep (k )
j Im[x(n)]
X op (k )
N 1
Proof: X(k) =
nk
x
(
n
)
W
N
n 0
N 1
X(N-k) =
x(n)W
n 0
N 1
x
X (k) =
*
n 0
nk
N
X ((k )) N
( n)W N nk
N 1
X (N-k) =
*
nk
*
x
(
n
)
W
X
((k )) N
N
n 0
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X ((k )) N X * ((k )) N 1 N 1
x(n) x * (n) WNnk
2
2 n 0
= DFT of [Re[x (n)]]
N 1
N 1
17.
n 0
Let y(n) =
LHS
*
2
( n) x ( n) =
1
N k 0
( k ) X 2* ( k )
1
X 1 (k ) X 2* (k )
Y(k) =
N
1
=
N
X
N 1
l 0
(l ) X 2* (k l )
X
N 1
1
Y(0) =
N
l 0
(l ) X 2* (l )
x
Y(0) =
n 0
(n) x 2* (n)
N 1
x ( n) x
Therefore
n 0
*
2
N 1
( n)
1
=
N
X
k 0
(k ) X 2* ( k )
QUESTIONS
1 Q.
n
(iii)
1
1 e
j 2k
N
2nko
(iv) Cos
N
N 1
jN2k
e
k 0
j 2k
N
1 ( j 2k)
1 N
N
(k k o ) (k ( N k o )
2
(Impulse pair)
67
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Or
j 2nko
N
e
Sol.
x(n) =
j 2nko
N
e
2
We know that 1 N
x ( n) e
II.
j 2n ( N k o )
N
(k )
j 2nKo
N
X ( K Ko)
N
(k k o ) (k ( N k o )
2
x(n)
I.
e
2
j 2nko )
N
k=0
1k9
x(n) =
1
10 (k )
5
1
(n)
5+
Ans
3 Q. Suppose that we are given a program to find the DFT of a complex-valued sequence
x(n). How can this program be used to find the inverse DFT of X(k)?
N 1
X(k) =
nk
x
(
n
)
W
N
n 0
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N 1
x(n) =
nk
X
(
k
)
W
1
N k 0
N 1
N x*(n) =
*
nk
X
(
k
)
W
N
k 0
xp(n) = { 1
, 2, 3, 4, 5, 0, 0, 0}
x(n) = {1, 1, 0, 0, 0, 0, 0, 0}
1
X(k) =
x ( n)e
j 2k
n
8
n 0
=1+
n = 0 to 7
jk
4
Find DFT.
k = 0 to 7
X(0) = 1+1 = 2
X(1) = 1+
X(2) = 1+
X(3) = 1+
e
e
e
j
4
j
2
j 3
4
= 1.707 - j 0.707
= 1- j
= 0.293 - j 0.707
X(4) = 1-1 = 0
By conjugate symmetry X(k) = X*(N-k) = X*(8-k)
X(5) = X*(3) = 0.293 + j 0.707
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X(k) = { 2 , 1.707 - j 0.707, 0.293 - j 0.707, 1-j, 0, 1+j, 0.293 + j 0.707, 1.707 + j
0.707 }
6 Q.
Y(k) = X(k) e
= 4, j2, 0, -j2
IDFT x(n)
=
x(n) e
j 2
ln
N
jn
2
1
X (k ) X (k ) = 1 [ 24, -j16, 0, j16] = {6, -j4, 0, j4}
4
4
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1
4
1
2
X
(
k
)
[16 4 4] 6
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III UNIT:
FFT
N 1
X(k) =
x(n)W
n 0
N 1
nk
N
0 K N 1
n 0
N 1
N 1
nk
nk
nk
N
Re[x(n)] Re( W ) -
n 0
nk
Im[x(n)] Im( WN ) +
n 0
N 1
nk
n 0
additions.
4
N2
real multiplications
N
2
nk
WN
&
WNK N WNnk
DITFFT:
N
1
2
X(k) =
x(2n)W
n 0
(even)
= x (n)W
e
N
1
2
x(2n 1)W
n 0
N
1
2
2 nk
N
x (n)W
=
n 0
2 nk
N
( 2 n 1) k
N
(odd)
N
1
2
n 0
N
1
2
K
+ WN
x
n 0
(n)WN2 nk
N
1
2
nk
N /2
K
W
+ N
x
n 0
(n)WNnk/ 2
= Xe(k) + W N Xo(k)
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Although k=0 to N-1, 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
K
For K N/2
WN
N
2
K
W
=N
X(k) = Xe(k-N/2) - W N
N
2
Xo(k-N/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)
X(k) = Xe(k) +
W8k Xo(k )
= Xe(k-4) -
k = 0 to 3
W8k 4 Xo(k 4)
k = 4 to 7
X(0) = Xe(0) +
X(1) = Xe(1) +
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N
N
pt DFT can be expressed as combination of pt DFT.
2
4
2k
N
= Xee(k- )4
Xo(k) = Xoe(k) +
N
2( k )
4
N
k = 0 to
Xeo (k
N
-1
4
N
)
4
k=
(0 to 1)
N
N
to
-1
4
2
( 2 to 3 )
WN2 k Xoo(k ) k = 0 to N -1
4
N
= Xoe(k- ) 4
2( k
WN
N
)
4
Xoo(k
N
)
4
N
N
k=
to
-1
4
2
For N=8
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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)
Xo(0) = Xoe(0) +
W80 Xoo(0) ;
Xo(1) = Xoe(1) +
W82 Xoo(1) ;
Xo(2) = Xoe(0) -
W80 Xoo(0) ;
2
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No.
No.
of
of points N
No.
of
Complex
Multiplications
Improvement Factor:
Direct N2
Stages
Speed
N2
FFT
N
Log 2 N
2
N
Log 2 N
2
16
64
12
5.33
16
256
32
32
1024
80
12.8
64
4096
192
21.33
For N=8
No of stages given by= Log2N = Log28 = 3.
No. of 2 i/p sets = 2( Log 2N
-1 )
=4
NLog2 N
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= 8 * 3 =24
Each stage no. of butterflies in the stage= 2m-q 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
1
2
X(k) =
x(2n)W
n 0
N
1
2
x (n)W
n 0
nk
N
2
N
1
2
2 nk
N
x(2n 1)W
W Nk
n 0
N
1
2
x
n 0
( 2 n 1) k
N
(n)WNnk/ 2
K
4 pt DFT Xe(k) + W N Xo(k)
k= 0 to N/2 -1 = 0 to 3
N
(K )
N
N
Xe(k- ) - W N 2 Xo(k- )
2
2
2 pt DFT
k = N/2 to N-1 = 4 to 7
= Xee(k-N/4) - WN
N
)
4
Xeo(k-N/4)
= Xoe(k-N/4) - WN
N
)
4
k = 0 to N/4-1 = 0 to 1
k = N/4 to N/2 -1 = 2 to 3
k = 0 to N/4-1 = 0 to 1
Xoo(k-N/4)
k = N/4 to N/2 -1 = 2 to 3
W82 W41
N=8
X(0) = Xe(0) + W80 Xo(0) ;
Xe(0) = Xee(0) +
W80 Xeo(0) ;
Xe(2) = Xee(0) -
W80 Xeo(0)
2
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Xo(2) = Xoe(0) -
W80 Xoo(0)
Xo(3) = Xoe(1) -
W82 Xoo(1)
N
1
4
Xee(k) = x(4n)W
0
4 nk
N
x(4n)W
=
n 0
4 nk
N
4k
=x(0) + x(4) W8
Xee(0) = x(0)+x(4)
Xee(1) = x(0)-x(4)
x(0)
x(0) x(0)
x(4)
x(2) x(1)
x(2)
x(4) x(2)
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x(6)
x(6) x(3)
x(1)
x(1) x(4)
x(5)
x(3) x(5)
x(3)
x(5) x(6)
x(7)
x(7) x(7)
Other way of representation
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DIFFFT:
N
1
2
X(k)
x(n)W
n 0
N
1
2
= x( n)W
n 0
N
1
2
x(n)W
n 0
nk
N
nk
N
N 1
nk
N
n 'k
x
(
n
'
)
W
N
1
2
x(n N / 2)W
n 0
N
k
2
N
put n = n+N/2
n1 N / 2
( n N / 2) k
N
N
1
2
x(n N / 2)W
nk
N
n 0
N
1
2
N W nk
[
x
(
n
)
=
+ (-1) x(n+ )] N
2
k
n 0
N
1
2
X(2k)
N
1
2
N
{[
(
)
x
n
X(2k+1) =
- x(n+
)] W N } W N / 2
2
n
nk
n 0
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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)
g(0) = [x(0) - x(4)]
W80
W81
W82
W83
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N
1
4
X(4k)
[ f (n) + f(n+ N )] W N / 4
=
4
nk
n 0
N
1
4
X(4k+2) =
N
1
4
nk
N
[
g
(
n
)
W
X(4k+1) =
+ g(n+
)] N / 4
4
n 0
N
1
4
nk
N W
X(4k+3) = [{g ( n) - g(n+
)}
N / 2 ] WN / 4
4
nk
n 0
W84 k
2
4k
W
X(4k+2) = f(0) f(2) + { [ f(1) f(3) ]
8 } W8
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W82
W82
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UNIT-IV
DIGITAL FILTER STRUCTURE
The difference equation
NP
y(n) =
a k x(n-k) + bk
M
k N F
k 1
NP
k N F
H(z) =
y(n-k)
z k
1 bk z
Np NF
or = A
Z NF
k 1
(1 C k ) Z 1
M
(1 d
k 1
k 1
) Z 1
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1 a1 Z 1 a2 Z 2
H(z) =
1 b1 Z 1 b2 Z 2
Ex:
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z 5 5 1 Z 2
Z
3 12 12
12
1
2
H(z) =
Z
Z
1
2
4
z 5 1 5Z 2
1
Z 3
1 Z
3 4
4
4
Z 1 Z 2
1
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
Z 1 Z 2
Z 1 Z 2
4
3
2
4
2
4
Ex:
Z Z
Z 2
H(z) =
Z 1 Z 1 Z 1
1
1
1
2
4
8
0.65 0.45Z
Z
Z 2 1.45 Z 1
Z 1 Z 1 Z 1
1
1
1
2
4
8
=Z
Z 1
1
1 Z
Z 3
Z 1 Z 1 Z 1
1
1
1
2
4
8
Z 2
Z 1 Z 2
1
4
32
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1
2
4
1 5 1 5 2 Z 3
Z Z Z
12
12
3 12
Z 1 Z 2
1
2
4
Z 1 7
Z 1 Z 2 Z 3
5 2 1 5 1
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
Z
Z
3
4
2
H(z) = Z
1
Z
3
1 Z
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Ex:
Z Z
Z 2
H(z) =
Z 1 Z 1 Z 1
1
1
1
2
4
8
obtain PSOS
1 Z 1 2Z
1
A
B
Z Z
Z = Z 1 Z 1
1
1
1
1
1
2
4
8
2
4
A = 8/3
B = 10
C
Z 1
1
C = -35/3
N ( z)
H(z) =
D( z )
N
D(z) = b Z
i 0
N i
ROWS
COEFFICIENTS
bo
b1 . bN
bN
bN-1 . bo
Co
C1 . CN-1
CN-1 CN-2 . Co
do
d1 . dN-2
dN-2
dN-3 . do
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.
.
.
2N-3
Ci =
di =
r0
r1
r2
bo
b N i
bN
bi
co
c N 1i
c N 1
ci
i = 0,1,N-1
i = 0,1,N-2
i. D(1) > 0
ii. (-1)N D(-1) > 0
iii. bo bN
co c N 1
ro r2
d o d N 2
Ex:
Z4
H(z) =
4Z 4 3Z 3 2Z 2 Z 1
1
1 1
3 4
15 11 6
224 159 79
11 15
bo b4
D(z) = 4Z 3Z 2Z Z 1
co c3
do d2
Ex:
H(z) =
1
7
1
1 Z 1 Z 2
4
2
4Z 2
=
4Z 2 7 Z 2
Ans: Unstable
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UNIT-V
Non Recursive filters
Recursive filters
y(n) =
Np
ak x(n-k)
y(n) =
k Nf
Np
ak x(n-k)
y(n) =
k 0
ak x(n-k)
bk y(n-k)
k 1
bk y(n-k)
k 1
k0
ak x(n-k)
ak x(n-k)
k 0
NP
General TF : H(z) =
Always stable.
k N F
z k
1 bk z k
k 1
FIR filters
IIR filters
tone
generators
filters
for
digital
telephone
3. Realized non recursively.
Realized recursively.
4. stable
Stable or unstable.
H(n) = an u(n) a<1 stable
= 0
a>1 unstable
Less
Less storage
Quantization noise
precision
arithmetic
can
be
made
negligible
8. Co-efficient accuracy problem is
More
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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 splane singularities of analog filter to z-plane.
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 s-plane are assigned to
H(s).
BUTTERWORTH FILTER DESIGN
1
1
j c
2N
Where s = j c
H B ( j c ) =
2
1
2
or
H B ( j c ) db = -3dB s
It has 2N poles
s
1
j c
2N
=0
2N
= -1
S2N = -1 ( jc )2N
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= e (e c )
2N
S = c
2N
2N
= c
2N
j 2N
2
e j 2 m
1 N 2 m
1 N 2 m
j
2N
Sm = c
0 m 2N 1
( 4 2 m )
6
=e
2
3
,e ,e
4
3
,e
5
3
,e
,e
7
3
Vo ( s )
Vi ( s )
c
1
CS
R
RS
1
CS
1
s
c
1
1 RCS
1
1
c
Poles that are let half plane are belongs to desired system function.
H B ( j) =
2
1
1
c
2N
For a large , magnitude response decreases as -N, indicating the LP nature of this
filter.
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H B ( j)
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dB
= 10log10 H B ( j)
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.
for
0 1404
( p = 1404 )
for
8268
( s = 8268 )
Stop band:
H ( j) dB < -60
2
If the c is given
s
H ( js) = [ 1
c
2
2N
log(10 6 1)
=N>
s
2 log(
)
c
Since c is not given, a guess must be made.
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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
log2 8268 = 2.56 octaves.
1404
1 oct ----
- 6N dB
2.56 ------
=>
N=
59
3.8
2.56 X 6
There fore: N =4
s
Now H B ( js) = [ 1
c
2
s 10
2N
s 2 N > 106
6
2N
2N
> 106
c 2 N
c <1470.3
Let c =1470.3
At this c it should satisfy pass band specifications.
p
H B ( jp) = [ 1
c
2
2N
= 0.59
This result is below the pass band specifications. Hence N=4 is not sufficient.
Let N=5
c < s X 10
6
2N
= 2076.8
10
2
1404 -1
In the pass band H B ( jp) = [ 1
] = 0.98
2076
Since N=5
c = 2076
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S1 = -2076
S2, 3 = 2076 (cos (4 /5) j sin(4 /5)) = 2076 e
S4, 5 = 2076 (cos (3 /5) j sin(3 /5)) = 2076 e
j144
j108
2076 5
HB(s) =
2. the magnitude response is maximally flat about =0, in that all its derivatives up to
order N are equal to zero at =0
Ex: c=1, N=1
H B ( j) = (1+ )
2
2 -1
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N
for large , where N is the no. of poles of
2
Avo
w
1
wo
2N
Avo
H B (s) =
Avs =
s
1
j
2N
If n is even S2N = 1 = e j ( 2k 1)
The 2N roots will be Sk= e
Sk = Cos(2k 1)
2N
j ( 2 k 1)
jSin (2k 1)
2N
2N
1
k=1,2,.2N
N /2
(s
k 1
where
2Cos k s 1)
= (2k 1)
2N
If N is odd
S2n =1 = e j 2k
Sk = e j 2k / N
T(s) =
k=0,1,2.(2N-1)
1
( N 1) / 2
k 1
where
( s 2 2Cos k s 1)
=k
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0 20 log H ( j) K1 for 1
20 log H ( j) K 2
for 2
1
= K1
10 log
1 2 N
1
c
2n
1
10 log
2 2 N
1
2n
= K2
k 1
10 10 1
10
k 2
10
k1
10
1
10 1
Dividing k 2
2n
10 10 1
10k1
10 1
log 10 k 2
10 10 1
n=
1
2 log 10
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
c =
10
k1
10
1
2n
or
c =
10
k2
10
1
2n
End
CHEBYSHEV FILTER DESIGN
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2 2 S
Defined as Hc(S) Hc(-S) = 1 C N
jp
CN(x)
2x2-1
4x3-3x
8x4- 8x2 +1
etc..
Two features of Chebyshev poly are important for the filter design
1. C N (x) 1
2 1
for x 1
H c ( j) 2 1
for 0 p
for 0 p
Whereas the frequency value important for the design of the Butterworth filter was the
c , the relevant frequency for the Chebyshev filter is the edge of pass band p .
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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
1
N1
Minor r = p
Major R = p
1
N1
N = Order of filter.
SP = r Cos +j R Sin
Ex:
Pass band:
-1< H ( j) dB 0
2
for
0 1404
for
8268
Stop band:
H ( j) dB < -60
2
> -1dB
-1dB = 0.794
= 0.508
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H c js
2 2 s
<10-6
=
N
jp
10 6 1 2
CN(5.9) >
= 1969
2
s
5.9
Since
p
Evaluating
C3(5.9) = 804
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:
=0.508-1 + (1+0.508-2)1/2 = 4.17
1404
R=
2
1
1
4
4
4.17 4.17 702 (1.43 0.67) 1494
1
1
1404
4
4
r=
4.17 4.17 512
2
poles locations :
S1,2 = 512Cos
7
7
j1494Sin
=
8
8
S3,4 = 512Cos
Hc(S) =
7 5
,
8
8
5
5
j1494Sin
=
8
8
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.
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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
H(z) (at z =
e ST ) = h(n )e
re jw e j)T
r = eT
e jw e jT w T
Z1 = eT e jT
Let S1 = j =>
S2 = j (
STn
2
) =>
T
Z2 = eT e jT j 2 eT e jT
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If the real part is same, imaginary part is differ by integral multiple of 2 , this is the
biggest disadvantage of Impulse Invariance method.
Let HA(S) =
sa
sa
=
2
2
s a jb s a jb
s a b
hA(t) = e atCosbt
=0
for t 0
otherwise
s2 = -a+jb
h (nTs) = e anTsCos(bnTs)
H(z) =
s1 = -a-jb
for n 0
1 e aTsCos (bTs) Z 1
1 2e aTsCos (bTs) Z 1 Z 2
1 e aTsCos(bTs) Z 1
(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 Z-plane at Z = e pTs , however, the
finite zero located in the s-plane at s= -a was not converted into a zero in the z-plane at Z =
e aTs , although the zero at s= was placed at z=0.
1, 2Ts
Z3,4 = e S
H(z) =
3 , 4Ts
), therefore Ts = 10-4]
Ts
k
(1 2 * 0.862Cos10.2Z 0.862 Z )(1 2 * 0.94Cos 24.8Z 1 0.94 2 Z 2 )
1
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k
(1 1.69Z 0.743Z )(1 1.707 Z 1 0.88Z 2 )
1
1 Z 1
S=
Ts
Or
Using forward-difference mapping based on first order approximation Z = e sTs 1+STs
S=
Z 1
Ts
Z 1 1 Z 1
=
ZTs
Ts
d dx
d 2x
/t=nTs = / t nTs
2
dt dt
dt
Ts
Ts
Ts
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1 2Z 1 Z 2 1 Z 1
S
=
Ts 2
Ts
1 Z 1
Therefore S =
1 Z 1
1
1 STs
= 0.5 +
0.5(1 STs)
1 STs
1
jTs
1
=
2
2
1 jTs 1 Ts 1 2Ts 2
Z - 0.5 =
0.5(1 STs)
(1 STs)
Using Forward-difference
S=
Z 1
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 RHS-plane maps to right of z=1.
If <0, then u<1, the LHS-plane maps to left of z=1.
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Bilinear Transformation
Provides a non linear one to one mapping of the frequency points on the jw axis in splane to those on the unit circle in the z-plane.
This procedure also allows us to implement digital HP filters from their analog
counter parts.
S=
2 Z 1 2 1 Z 1
=
Ts Z 1 Ts 1 Z 1
2 Z 1
Ts Z 1
STs
1
And Z = 2
STs
1
2
1/ 2
e jw
jw
Ts
2 Ts 2
j tan 1
2
Ts
1 e
j 1
2
=
1/ 2
2
Ts
Ts
j tan 1
Ts
j 1 2
2
1
e
2
2
j 2 tan 1
Ts
2
w=2tan-1
Ts
2
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The entire j axis in the s-plane - <j < maps exactly once onto the unit circle w such that there is a one to one correspondence between the continuous-time 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 s-plane maps on to the inside of the
unit circle in the z-plane and the right half of s-plane maps onto the outside.
In Inverse relationship is
2
w
tan
Ts 2
w w3
....
Sin
2 8
2
2
w
2 =
For smaller value of frequency
2
w
Ts
Ts
Ts
w
Cos
1
....
2
4
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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 w-domain.
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 s-plane to the z-plane.
Let HA(s) =
1
S Sp
S=Sp
1
H(z) =
=
2 1 Z 1
Sp
Ts 1 Z 1
Ts(1 Z 1 )
2 SpTs1 2 SpTs Z 1
2 SpTs
A pole at S=Sp in the s-plane gets mapped into a zero at z= -1 and a pole at Z =
2 SpTs
2 SpTs
Ex:
Chebyshev LPF design using the Bilinear Transformation
Pass band:
-1< H ( j) dB 0
for
for
Stop band:
H ( j) dB < -60
2
Ts
4
tan
= 2*10 tan(0.0702 ) = 4484 rad/sec
Ts 2
And s* =
2
Ts
4
tan
= 2*10 tan(0.4134 ) = 71690 rad/sec
Ts 2
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Pass band:
-1< H ( j*) dB 0
for
0 * 4484 rad/s
for
* 71690rad/sec
Stop band:
H ( j*) dB < -60
1dB
= 0.508
s *
H c js * = 1 2C N2
p *
2
Since
<10-6
s *
16
p *
CN2(16)
10 6 1
<
10 6 1 2
CN(16) <
= 1969
2
(0.508)
C3(16) = 16301
N = 3 is sufficient
Using Impulse Invariance method a value of N=4 was required.
=4.17
1
N1
N
1
1
4484
3
4.17 4.17 3 5001
=
Major R = p *
2
2
r=
1
1
4484
3
3
4
.
17
4
.
17
2216
2
2
3
2
2
j 5001 Sin
= -1108 j 4331 = 4470 e j104.4
3
3
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Hc(s) =
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4.43 *1010
( s 2216)( S 2 2223s 4470 2 )
Pole Mapping
At S=S1
In the Z-plane there is zero at Z = -1 and pole at Z =
2 (2216 *10 4 )
0.801
2 (2216 *10 4 )
1 Z 1
1 2Z 1 Z 2
1 0.801Z 1 1 1.638Z 1 0.81Z 2
1
pole ar Z=d and zero at z=0
1 dZ 1
= (1+c )
1 Z 1
Z
1 Z 1
1
HLZ(Z) = 1-c
1 c
The low pass zero at z=c is transformed into a zero at z=C1 where C1 =
And pole at z=0 is Z=
Similarly,
HLP(Z)=
1 Z 1
1 d 1 d Z 1
1 d
1 d
1 Z 1 2Z 2Z
1 0.622Z 1 1.07Z 0.674Z
1
H(z) = K
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K=
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1 (1)(0.356)3
(1 0.801* 0.356)(1 (0.819 j 0.373)(0.356))(1 (0.819 j 0.373)(0.356))
0.029
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UNIT-VI
Phase Delay: p
( )
Group Delay: g
d ()
d
() o
o
Changes with frequency
p
g = - =constant.
Type 1 Sequence
Center of Symmetry M=
N 1
integer value
2
N 3
jnT
H(w) = h(n) 2 h(n)CosT (n m) e
n 0
N 1
T
2
Amplitude spectrum is even symmetric about w=0 & & both H(0) & H( ) can be non
zero.
Type 2 Sequence
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h(n) = h(N-1-n)
Center of Symmetry M=
N 1
half-integer 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
M=
N 1
integer value
2
N23
2 h(n) SinT ( M n) e jMT
H(w) = j
n 0
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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(N-1-n)
N 1
H( e
jT
) = h ( n )e
jnT
n 0
N
1
2
= h( n)e
N 1
h( n)e
+
jnT
n 0
jnT
N
2
Let N-1-n = m
N
1
2
h( n)e
n 0
jnT
h ( N 1 m) e
m
jT ( N 1 m )
N
1
2
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N
1
2
==
h( n)e
n 0
N
1
2
h ( n )e
jnT
h ( m) e
N 1
jT
N 1
jT
n 0
= 2 h ( n )e
N 1
jT
n 0
=e
N
1
2
+ h ( m) e
jT ( N 1 n )
N 1
jT
n 0
2 h ( n )e
N
1
2
jT ( N 1 m )
m 0
n 0
N
1
2
jnT
N
1
2
T
j ( T ( N 1 n )
( N 1)
j ( nT 2T ( N 1)
2
e
e
N 1
cos T n
2
N
N 1 1
jT
2
2
n 0
N 1
----Magnitude
2
2h(n) cos T n
N 1
T Linear Phase
2
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FIR Filters
Fourier series Method
Fs
Fs
F
2
2
2Fs
2Fs
2F
2
2
s
s
2
2
1. Frequency response of a discrete-time 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
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H (e
jT
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h( n)e
jnT
T = sampling period
The F.S co-efficient or impulse response samples of filter can be obtained using
1 s / 2
H (e jT )e jnT d
h (n) =
s s / 2
clearly if we wish to realize this filter with impulse response h(n), then it must have finite
no. of co-efficient, which is equivalent to truncating the infinite expansion of H (e jT ) , which
leads
H 1 (e
to
jT
approximation
H (e jT ) ,
of
which
is
denoted
by
jnT
h
(
n
)
e
n m
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
However, this filter cant 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.
this difficulty can be overcome by introducing a delay M=
M
-M
Therefore
-M
H(z) = Z H1(z) = Z
h( n) Z
N 1
samples.
2
n M
H(z) =
b Z
i 0
Properties:
1. N=2M+1, impulse response co-eff, bi = 0 to 2M.
2. h(n) is symmetric about bM
Ex: M=4
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H (e jT ) e jMT H1 (e jT )
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
= 0 for c
h(n) =
s
2
1 c jnT
e
d
s c
2 c
Cos (nT )d
=
s 0
=
2 SincnT
s
nT
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2
=
1
Fs
2Fsn.
SincnT
1
SincnT
n
bi = h(i-M)
2M
bZ
H(z) =
i 0
w= T
s
2Fs 1
T
2
2 Fs
Ex:
Design LPF that approximate following freq response.
H(F) = 1
=0
0 F 1000Hz
When the sampling frequency is 8000 SPS. The impulse response duration is to be
limited to 2.5ms
Ti = 2MT
2.5 *10 3
10
M=
1
2*
800
N=21
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1 c jnT
1.e
d
h(n) =
s c
1 Fc j 2FnT
1.e
2dF
=
2Fs Fc
=
1 Fc j 2FnT
2 Fc
e
dF
Cos(2FnT )dF
=
Fs Fc
Fs 0
1
1
Sin 2FcnT
Sin (0.25n )
n
n
________________________________________________________________
OR
w = T = 2 *1000 *
Hc(w) = 1
=
1
2
else where
1.e
8000 4
jwn
dw = 1 Sin (0.25n )
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(i-10)
20
H(z) =
b Z
i 0
FIR HPF
h(n) =
s / 2
1 c
1.e jnT d e jnT d
c
s s / 2
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1 e jnT
=
s jnT
c
s / 2
e jnT
jnT
s / 2
c
s
s
j nT
j nT
jcnT
2
1 e
e
e 2 e jcnT
= s
jnT
s
s
j nT
j nT
jcnT
jcnT
2
2 1 e
e
e
e 2
= s nT
2j
2j
s nT
Sin c nT Sin 2
2
2FsnT
1
sin c nT Sinn
n
1
sin c nT
n
FIR BPF
2 u
cos nT
h(n) =
s l
d =
1
sin u nT sin l nT
n
Ex:
Desing a BPF for H(f) = 1
=0
160 F 200Hz
else where
Fs = 800SPS
Ti = 20 ms
M=
Ti 20 *10 3
8
1
2T
2*
800
N = 17
Sin
2
F
nT
Sin
2
F
nT
h(n) =
=
u
l
n
n
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h(0) = 0.1
h(4) = 0.07568
h(1) = 0.01558
h(5) = 0.06366
h(2) = -0.09355
h(6) = -0.05046
h(3) = -0.04374
h(7) = -0.07220
16
H(z) =
b Z
h(8) = 0.02338
i 0
bi = h(i-8)
h(-n) = h(n)
WINDOWING
Disadvantage of F.S is abrupt truncation of FS expansion of the freq response. This
truncation result in a poor convergence of the series.
The abrupt truncation of infinite series is equivalent to multiplying it with the rectangular
sequence.
n M
WR(n) = 1
=0
else where
H (e jw ) H (e jw ) *WR (e jw )
1
=
2
H (e
)WR (e j ( w ) )d
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wN
wN
NSa
2
2
1.e jwn dw
jw
WR(e ) =
=
w
w =
N 1
Sa
n
Sin
2
2
2
N 1
2
Sin
4
& 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
4. zero in the range n
N 1
2
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Let us consider the effect of tapering the rectangular window sequence linearly from the
middle to the ends.
Triangular Window:
WT (n) 1
2n
N 1
=0
N 1
2
else where
In this side lobe level is smaller that that of rectangular window, being reduced from -13
to -25dB to the maximum. However, the main lobe width is now
8
. There is trade off
N
2n
W(n) = (1 )Cos
N 1
=0
for n
N 1
2
else where
Hamming Window
4n
2n
Blackman Window
0.08Cos
N 1
N 1
2n
Io 1
N
Wk (n)
Io( )
=0
for n
N 1
2
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
1 x k
Io(x) = 1+ k! 2
k 1
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Window
Peak amplitude
of side lobe dB
Transition width
of main lobe
Minimum stop
band deviation dB
Rectangular
-13
4
N
k=1
Triangular
-25
8
N
k=2
Hanning
-31
8
N
k=2
Hamming
-41
8
N
k=2
BlackMan
-57
12
k=3
N
-74
Kaiser
variable
variable
-21
-25
-44
-53
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
of window used. If Wt is the transition width, we must have Wt = W2-W1 k .
2
N
N K
2
w2 w1
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N 1
for n
N 1
2
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
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H(z) =
JNTU World
2M
b' Z
i 0
bi' = h(i-M)
h(-n) = h(n)
0 i 20
Ex:
Find a suitable window and calculate the required order the filter to design a LP digital
filter to be used A/D-H(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
Digital .. w1 wc cT 30
w2 2T 45
1
0.3 k1 3dB
100
1
0.45
100
k 2 50dB
2
2.2
=26.65
w2 w1 0.45 0.3
N = 27 is selected
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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 :
s 10
0.05 As
Ap = 20 log 10
100.05Ap =
10 0.05 Ap 1
, p 0.05 Ap
10
1
1 p
1 p
As = 20 log 10
=> As =
1 p
s
20 log s
1
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
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Step 3:
Calculate the parameter as follows for
=0
for Aso 21 dB
Step 4:
Calculate D as follows
for Aso 21 dB
D = 0.9222
=
As 7.95
14.36
Step 5:
Select the lowest odd value of N satisfying the inequality
N
samD
1
t
for LPF
= p- s
for HPF
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1
p s
2
c1 = p1
t
t
; c2 p2
2
2
c1 = p1
for BPF
t
t
; c2 p2
2
2
for BSF
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:
The impulse response samples can be calculated using h(n) =
Where c =
1
sin c nT
n
1
(1.5 2.5) =2rad/s
2
And the no. of points required in this sequence can be found as follows
Step1:
5.7564*10 3
Step 2:
3
0.4
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D = 2.566
Step 5:
N
10(2.566)
1 26.66
1
=> N=27
2n
Io 1
N 1
Wk (n)
Io( )
Wk (0)
Io( )
1
Io( )
Io 3.9524 1
26
Io(3.94)
10.269
0.9899
Wk (n)
=
Io(3.9524) 10.3729
Io(3.9524)
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OBJECTIVE PAPER-1
1)What is the parsvals theorem expression in DTFT :
2
n=-|x(n)|2=(1/2)
dw
Match the following:
2) E = , P = 0
3) E , P = 0
4) E = , P 0, P
a) power
b) Neither energy nor power
c) Energy
a) power
b) Neither energy nor power
c) Energy
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
17) DFT [ x* (-n) ] = ?
a) X * (K) b) X * (-K) c) X * (N-K) d) none
1
2
3
4
5
6
7
8
9
10 11
b
c
a
c
a
b
a
b
d
a
12
c
13
c
14
b
15
a
16
a
17
a
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OBJECTIVE PAPER-2
1) The region of convergence of the Z-transform of a unit step function is
a)|Z| >1 b) |Z|<1
c) (real part of Z ) >0
d) (real part of Z ) <0
2) The Z T of the function f(nT) = anT is
a) Z/(Z-aT)
b) Z/(Z+aT) c) Z/(Z-a-T)
d) Z/(Z+a-T)
b) Z/(Z-1)
(n k ) is
k 0
d) (Z-1)2/Z
c) Z/( Z-1)
x(n)
y(n)
+
+
Z-1
-b
-1
-1
a) (1+aZ )/ ( 1+bZ )
c) (1+aZ-1)/ ( 1-bZ-1)
a
b) (1+bZ )/ ( 1+aZ )
d) (1-bZ-1)/ ( 1+aZ-1)
-1
-1
6) A linear discrete time system has the characteristic equation Z3-0.8 Z=0, the system
a) is stable
b) is marginally stable
c) is un stable
d) stability cannot be assessed from the given information
7) The advantage of Canonic form realization is
a) smaller no of delay elements
b) larger no of delay elements
c) hard ware flexibility
d) none
3
8) y(n) =
akx(n k )
k 2
k 1
b k(yn k )
d) 11
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1
a
2
a
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3
c
4
c
5
a
6
a
7
a
8
c
10
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)
13) To realize FIR filter
a) no feedback paths and forward path
c) feedback paths and no forward path
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
F
12
T
13
a
14
12
15
24
16
240
Z
Z
x(n)
y(n)
Z-1
b1
b2
anp
bm
a1
Z-1
b3
Z-1
bnp
Z-1
a2
a3
Z-1
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OBJECTIVE PAPER-3
State TRUE or FALSE
1) u(n) =
(n k )
K 0
5)
h(k )
2 linear
7) y(n) = |x(n)|
3 Stable
8) y(n) = x(n2)
4 Time invariant
7
4
a) y (n) =
b) y (n) =
h( k ) x ( n k )
d) y (n) =
9
B
10
11
B
12
A
h( k ) x ( n k )
K 0
1
c) y (n) =
8
2
h( k ) x ( n k )
K 0
h( k ) x ( n k )
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c) x(k)
d) none
a) 1, 0, 2, 0, 3
b) 1, 1, 2, 2, 3, 3
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
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OBJECTIVE PAPER-4
1) If x(n) = {-1, 0, 1, 2, 1, 0, 1, 2, 1, 0, -1} What is X(0)
a) 6
b) 10
c) 0
2) If x(n) = 1, |n|2
0, other wise
Find DTFT
a) sin(5w)/sinw
b) sin(4w)/sinw
d) none
c) sin(2.5w)/sin(0.5w)
c) r(n-1)
d) none
d) None
d)
(T / F)
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13) Reversing the N point sequence in time is equivalent to reversing the DFT values (T /
F)
14) FT of non periodic discrete time sequence is non periodic
(T / F)
Match the following: For a real valued sequence, the DTFT follow the properties as
15) Re [H (jw) ]
a) Real valued function of w
16) Im[ H(jw) ]
b) even function of w
17) F.T [even symmetric sequence]
c) Imaginary valued function of w
18) F.T [odd symmetric sequence]
d) odd function of w
n=N-1
19) Write DFF & IDFT formulas.
X(k)=x(n)Wnnk
n=0 N-1
x(n)=(1/N)X(k)Wnnk
K=0
20) Total no of real multiplications in DFT is:
1
2
3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20
A C A A C B A A A A T F T F B D A C
4n2
OBJECTIVE PAPER-5
Choose the Correct Answers
1. The Fourier transform of a finite energy discrete time signal, x(n) is defined as [
x(n) e
a) X( )=
jn
n=-
b) X( )=
n
x(n) e
n=-
c) X( )=
x(n) e
-jn
n=-
-jn
x(n) e
d) X() =
n=0
1
N
1
N
N 1
n 0
n 0
j 2kn
X(k) e N
j 2kn
X(k) e k
b) a) x(n) =
1
N
d) a) x(n) =
1
N
N 1
n 0
n 0
[ ]
j 2kn
X(k) e N
j 2kn
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) e-j2nk
b) 1
c) e-j2nok/N
d) e-j2nk /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+M-1 c) L if L>M d) 2L if L=M
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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
[
]
Fill in the blanks
9. The 4-point DFT of a sequence x(n) is ________
10.DFT of a sequence x(n) = (n-n0) is __________
11.An N point sequence is called ________________ if it is antisymmetric about point
zero on the circle
12.The two methods
_______________
of
13.DFT of multiplication
_____________________
sectioned
of
two
convolution
sequences
are
DFT
________________
{x1
(n)
x2(n)
&
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 __________________
16. Match the following
1 DFT [ x1(n) x2(n) ]
a) X (N-K)
1
[ X1(k) X2(k)]
N
2. DFT [ x*(n) ]
b)
3. DFT [ x((-n))N ]
c) X*(N-K)
4. X1(k) X2*(k)
d) x1(n) x2(n)
e) x1(n) x2*(-n)
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1=
JNTU World
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(2-n)
(2M)
(2M)
OBJECTIVE PAPER-6
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 tan-1(Ts/2)
b) w = tan-1(Ts/2)
c) w = 2 tan-1(Ts)
d) w = 2 tan-1(/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 Z-plane as
a) Z=e-SPTs
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
B
2
A
3
C
4
A
5
B
6
B
7
F
8
T
9
F
10
T
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k/(s2-10552.7s+(2076)2
(2M)
(2M)
OBJECTIVE PAPER-7
Choose the correct Answer
1. In impulse invariant transformation the mapping of analog frequency to digital
frequency is
[
]
a) one to one
b) many to one
c) one to many none
2. The digital frequency in Bi linear transformation is
[
]
-1
-1
a) = 2 tan ( T /2)
b) = tan ( T /2)
-1
c) = 2 tan ( T )
d) = 2 tan-1( /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
a) S = 2
1 z 1
b) S =
1 z 1
1 z 1
c) S = 2
1 z 1
d) S=
Z 1
Z 1
1
1
b)
c)
1
1
d) 1 2
1
1 C N ( / c )
1
2
c) H a () =
2 2
1 C N ( / c )
2
a) H a () =
1
1 C ( / c )
1
d) H a () =
2
1 C N ( / c )
2
b) H a () =
2
N
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a)
Sinwn / 2
Sinw / 2
b)
Sinwn / 2
Sinw
c)
Sinwn / 2
Sinwn
d)
Sinwn / 2
Sinwn / 2
15.For same specifications, the order N of chebyshev filter is less compared to Butter
worth filter.
[
]
16.FIR filter have non-linear phase characteristics.
20.In FIR filter with constant phase delay the impulse response is symmetric
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OBJECTIVE PAPER-8
CHOOSE THE CORRECT ANSWER
1. The DTFT of a sequence x(n) is
a)
x(n)e
jwn
b)
x(n)e
jwn
c)
x(n)e
jwn
d)
dw
c) x[ e
j ( wwo )
x(n)e
jwn
a) X1[w] X2[w]
dw
d) x[ e
j ( w wo )
1
d) X1[w] * X2[w]
N
c) X1[w] * X2[w]
4. The smallest value of N for which x(n +N) = x(n) holds is called
[
]
a)
Fundamental period b) Fundamental frequency c) fundamental signal d) None
5. DFS of real part of periodic signal is
a) Xe(K)
b) Xo (K) c) XR(K)
a)
x(n)W
n 0
N 1
Kn
N
b)
x(n)W
n 0
Kn
N
N 1
C)
x(n)W
K 0
N 1
Kn
N
d)
x(n)W
n 0
1
1
X1[K] * X2[K] b)
X1[K] + X2[K]
N
N
d) XIm(K)
Kn
N
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)
9. Zero padding means
a) increasing length by adding zeros at the end of sequence
b) Decreasing length by removing zeros at the end
c) Inserting zeros in between the samples
d) None of the above
II STATE TRUE OR FALSE
10. The F.T of discrete signal is a discrete function of
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
[
]
]
]
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OBJECTIVE PAPER-9
CHOOSE THE CORRECT ANSWER
1. Power signal is
a) Periodic
b) aperiodic
c) Continuous
d) none
2. WN nK is
a) e
j 2 K
N
b) e
j 2 nK
c) e
j 2 Kn
N
d) e
2 Kn
N
3. When the sequence is circularly shifted in time domain by m samples i.e. x((n-m))N
then on applying DFT, it is equivalent multiply sequence in frequency domain by
a) e
j 2 Km
N
b) e
j 2 Km
N
c) e
j 2 Km
d) e
2 Km
N
DFT
d) x1(n) x2(n)
X1(K)X2(K)
K 0
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15.How many multiplications and additions are required to compute N-point DFT using
radix 2 FFT
16.How many multiplications and additions are required to compute N-point DFT
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OBJECTIVE PAPER-10
1. How we can calculate IDFT using FFT algorithm.
(2M)
5. Find the Z-Transform and ROC for the signal x(n) = an u(n).
6. Find the Z-Transform and ROC for the signal x(n) = - an u(-n-1).
8. Z{(n)} = ..
9. Find inverse Z-Transform for X(z) =
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 PAPER-11
ANSWER THE FOLLOWING
1. What are the advantages of digital filter over analog filter.
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.
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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.
[
]
19.Butterworth filter exhibit monotonic behavior both in passband and stopband.[
20.For the given specifications order of the Chebyshev filter is more as compared to
Butterworth filter.
[
]
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OBJECTIVE PAPER-12
1.
a.
b.
c.
Dynamic system
d.
Recursive System
e.
2.
3.
4.
5.
6.
------ 2 Marks
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7.
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8.
Find the Convolution Sum Graphically with all the steps-------3 Marks
x(n)=
2
1
h(n)=
1 1
-1
0 1
9.
--------2 Marks
10.
11.
12.
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OBJECTIVE PAPER-13
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.
[
]
3. The inverse Z transform of z/z-a is an u(n)
5. As the order of Butter worth filter increases than the response is closer to ideal filter
response.
[
]
Answer the following
6. Find the transfer function H(z) of the given difference equation
Y(n) = 0.7 y(n-1) 0.12y(n-2) + x(n-1) + x(n-2)
7. Indicate the poles and zeros of the given system and also check the stability of the
system
H(z) =
z ( z 1)
( z 0.2)( z 0.4)( z 0.5)
(2M)
3 3.6 z 1 0.6 z 2
1 0.1z 1 0.2 z 2
(2M)
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(2M)
1
(1 0.5 z )(1 0.5 z 1 )
1
z
using partial fraction
( z 2)( z 3)
method.
(2M)
z
for ROC :z >2
( z 1)( z 2)
(2M)
14.The order of the Butter Worth filter is obtained by using the formula N
______________________
15.The cut- off frequency c is obtained by using the formula
___________________________
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OBJECTIVE PAPER-14
Fill in the Blanks
1. The expansion of FFT is _______________
2. The main advantage of FFT is _____________
3. The number of multiplications needed in the calculation of DFT using FFT with 32point sequence = ________________
4. __________________ number of additions are required to compute N pt DFT using
radix 2 FFT.
5. What is decimation in time algorithm.
State TRUE or FALSE
6. For DIT FFT algorithm the input is bit reversed and the output is in natural order
[ ]
7. By using radix 2 DIT FFT algorithm it is possible to calculate 6-point DFT.[
8. WNNK 1
9. WNNK/ 2 1
10.In DIT FFT, the input sequence is divided into smaller subsequences
(2M)
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14. Write the steps for the calculation of IDFT using DIT FFT
(2M)
d) none
[ T/F ]
[ T/F ]
[ T/F ]
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ii)
(1M)
(2M)
(1M)
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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 UNIT-5
1 (a) Draw the frequency response of N-point rectangular window.
(b) Design a fifth order band pass linear phase filter for the following specifications.
i. Lower cut-off frequency = 0.4 rad/sec
ii. Upper cut-off frequency = 0.6 rad/sec
iii. Window type = Hamming
Draw the filter structure. [4+12]
2) Design a band pass filter to pass frequencies in the range 1-2 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!) = _ ej3!, 3 _ _ 3/4
0 , 3/4 _ || _
Determine H(ej!) for M=7 using a Hamming window. [6+10]
4) (a) Design a linear phase low pass filter with a cut-off frequency of /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 cut-off
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(n-1)+
0.25x(n-2). Calculate the phase delay and group delay. [8+8]
7) (a) Define Infinite Impulse Response & Finite Impulse Response filters and com-pare.
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(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 UNIT-7
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.
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