Signal Systems
Signal Systems
CHAPTER – 1
Signal: - A signal is a function representing a physical quantity or variable
Classification of Signals:-
A. Continuous time and Discrete – Time Signals:-
A Signal n(t) is a continuous –time signal if it is a continuous variable if it is a discrete variable,
that is, x(t) is defined at discrete times, then x(t) is a discrete – time signal
B. Analog and Digital Signals:-
Analog:- A Signal can take on infinite number of distinct values in amplitude
Digital:- A Signal can take on finite number of distinct values in amplitude
2 x(n)
x(t)
1 1
1 2 3 -1 0 1 2
Q.4 (1) (a) x(1 – t) (b) x(1 – t) + 2(2 – t) (c) x (3t) (d) x(t/3)
1 2 3 4 5 t 0 t
x(n)
x(n)
4
01 2 3 4 1 2 3
T = where T is min. value of t after which f(t) repeats is called fundament period.
f(t) = f1(t) f2(t) f3(t)
T T1 T2 T3
SIGNALS AND SYSTEMS
T1 T T
rational 2 2
T2 T2 T1
T = LCM [T1 & T2 & T3]
f1(t) = ASMT
2
T= 2
1
Q. if x(t), there check, periodic or a periodic, final fundamental period,
(a) 2 sin t + 4 sin 3t (b) sin 5 t + sin 7 t (c) sin + sint
j10t
(d) 2 + 3 cos πt (e) je (f) e(-1+ j)t
(g) ej7πn (h) 3ej3π(n+1/2)15 (i) 3ej3/5(n+1/2)
j[(π/2)t-1)]
( j) 2cos (10t + 1) – sin (4t – 1) (k) e (l) cos2π/8n
(m) u(n) + u[-n] (n) [ (n 4n) (n 1 4k )]
n
x(n) = x(n N)
N – integral No.
x(n) = e j0 n
x(n + N) = e j0 ( n N )
= e j0 n e j0 N
e j 0 n 1 e j 2
0 N 2
2 2
N= m M smallest integer which connects to integer no.
0 0
x1 (n) x 2 (n) x3 (n)
x(n) =
N1 N2 N3
N = LCM [N1 & N2 & N3]
Note :- Sum & difference of discrete periodic signal is periodic and sum & difference of
continuous periodic signal is periodic under certain condition
G. Energy and power signals : -
x(t )
2 2
E x(t ) dt , E
n
Th N
1 1
T T
2
P Lim x (t ) dt , P Lim [ x(n)]2
N 2 N 1
Th n N
0 0 t0
1
3). (at ) (t ) (-t) = (t) even function
a
4). x(t) (t) = x(0) (t) if x(t) is continuous at t = 0
x(t) (t – t0) = x(t0) (t – t0) if x(t) is continuous at t = t0
Any continuous signal x(t) can be expressed as
x(t) = x(t ) (t c)dc
5). nth derivative g(n)(t)
n
d (t ) g ( n ) (t ) dt ( 1)
d ( n ) (t ) g (t ) dt
du (t )
6). (t) =
dt
SIGNALS AND SYSTEMS
d (t ) (t )dt = - (0)
(a) k (b) n
Properties of δ(t)
1. (t)dt 1
2. x(t) (t)dt x(0)
3. x(t) (t t )dt x(t )
0 0
4. x( ) (t )d x(t)
1
5. (at) (t)
a
6. x(t) δ(t –t0) = x(t0) δ(t-t0)
t2
T
2
T 0
or, equivalently a 0 f (t )dt
T /2
Multiplying both sides of Eq.2.2 by cos m 0 t and integrating, we have f (t ) cos m
T / 2
0 t dt =
T/2 T /2 T /2
1
T T/ 2
a 0 cos m 0 t dt n
T / 2 n 1
a cos n 0 t cos m 0 t dt bn sin n 0 t cos m 0 t dt
T / 2 n 1
T /2
1
2 T/ 2
Here, a0 cos m 0 t dt 0
T /2 T /2 0, for m n
an
an cos n 0 t cos m 0 dt 2 [cos( m n ) 0 t cos ( m n ) 0 t ]dt T a , for m n
T / 2 T / 2 2 n
T /2 T /2
bn
bn sin n 0 t dt
T / 2
2 T/ 2
[sin(m n) 0 t sin (m n) 0 t ]dt 0
T /2
Ta n
Therefore, f (t ) cos n t dt =
T / 2
0
2
, for m n
SIGNALS AND SYSTEMS
T /2
2
T T/ 2
Hence, a n f (t ) cos n 0t dt (2.4)
T
2
T 0
or, equivalently a n f (t ) cos n 0t dt
Similarly, multiplying both sides of Eq. 2.2 by sin m 0t and integrating, we get
T /2 T /2
1
f (t ) sin m 0t dt =
T / 2
2 T/ 2
a0 sin m 0t dt
T /2 T/2
+ an cos n 0 t sin m 0 t dt b n sin n 0 t sin m 0 t dt
T / 2 n 1 T / 2 n 1
T /2
1
2 T/ 2
Here, a 0 sin m 0t dt = 0
T /2
a
T / 2
n cos n 0 t sin m 0 t dt 0
T /2 0, for m n
bn sin n 0 t sin m 0 t dt 1 bn for m n
T / 2 2
T /2
T
Therefore, f (t ) sin m 0 t dt bn , for m n
T / 2
2
T /2
2
T T/ 2
Hence, bn f (t ) sin n 0 t dt (2.5)
T
2
T 0
or, equivalently, bn = f (t ) sin n 0 t dt
a
f (t ) dt 2 f (t ) dt
a
f (t ) dt = 0. The sum of two or more odd functions is an odd function and the product of two
a
odd functions is an even function.
SIGNALS AND SYSTEMS
(iii) If f(t + T/2) = f(t), only even harmonics are present.
(iv) If f(t + T/2) = - f(t), only harmonics are present and hence the waveform has half – wave
symmetry.
A, from T / 2 t T / 4
where f(t) = A, from T / 4 t T / 4
A, from T / 4 t T / 2
Therefore,
2A
T / 4 T/4 T /2
an ( cos n 0 t ) dt cos n 0 t dt ( cos n 0 t ) dt
T T / 2 T / 4 T /4
2 A sin n 0 t sin n 0 t
T / 4 T /4 T /2
sin n 0 t
T n 0 n T / 4 n T / 4
0 0
2A n 0T n 0T n T
sin sin sin 0
n 0T 4 2 4
n 0T n T n T
sin sin 0 sin 0
4 2 4
8A n 0 T 4A n T
sin sin 0
n 0 T 4 n 0 T 2
When 0 T = 2 , the second term is zero for all integer values of n.
Hence,
8A n 4 A n
an sin sin
2 n 2 n 2
a0 = 0 (d.c.term)
4A 4A
a1 sin
2 n
4A
a2 sin 0
4 A 3 4A
a3 sin
3 2 3
………………………………………………………
………………………………………………………
T /4
2 2A T
=
T T / 4
A dt A
T 2
T/2
2
T T/ 2
an = f (t ) cos n 0 t dt
T /4
2
T T/ 4
= A cos n 0 t dt
T /4
2 A sin n 0 t
=
T n 0 T / 4
4A
= sin ( n 0T / 4)
n 0 T
When 0T = 2 , we have
2A n
an sin
n 2
= 0, for n = 2, 4, 6,…….
2A
, for n 1, 5, 9,13, ........
n
2A
, for n 3, 7,11,15, ........
n
Substituting the values of the coefficients in Eq. 2.2, we obtain
A 2A 1 1
f(t) = cos 0 t cos 3 0 t cos 5 0 t ...........
2 3 5
Example: Obtain the trigonometric Fourier series for the half-wave rectified sine wave shown in
Solution As the waveform shows no symmetry, the series may contain both sine and cosine
terms. Here, f(t) = A sin 0 t
To evaluate a0:
T
2
a 0 A sin 0 t dt
T 0
SIGNALS AND SYSTEMS
T /2
2
T A sin t dt
0
0
2A 2A
cos 0 t T0 / 2
cos( 0T / 2) 1
0T 0T
2A
Substituting 0 T = 2 , we have a0 = .
To evaluate an:
T
2
T 0
an f (t ) cos n 0 t dt
T /2
2 A n sin 0 t sin n 0 t cos n dt 0 t
0T n2 1
0
Substituting 0 T = 2 , we have
A
an [cos 1]
(1 n 2 )
2A
Hence, an , for n even
(1 n 2 )
= 0, for n odd
For n = 1, this expression is infinite and hence we have to integrate separately to evaluate a 1.
T /2
2
T 0
Therefore, a1 A sin 0 t cos 0 t dt
T /2
A
=
T sin 2 t dt
0
0
A
[ cos 2 0 t ]T0 / 2
2 0T
When 0 T= 2 , we have a1 = 0.
To find bn :
T
2
bn f (t ) sin n 0 t dt
T 0
T /2
2
T A sin t sin n t dt
0
0 0
T /2
A n sin 0 t cos n 0 t sin n 0 t cos 0 t
0T n2 1
0
When 0 T = 2 , we have bn = 0.
For n = 1, the expression is infinite and hence b1 has to be calculated separately.
T /2
2A
b1
T 0 sin 2 0 t dt
T /2
2A 0 t sin 2 0 t
2
0T 4 0
SIGNALS AND SYSTEMS
A
When 0 T = 2 , we have b1 =
2
Substituting the values of the coefficients in Eq. 2.2, we get
A 2 2
f(t) = 1 sin 0 t cos 2 0 t cos 4 0 t .........
2 3 15
COMPLEX OR EXPONENTIAL FORM OF FOURIER SERIES
From Eq. 2.2, the trigonometric from of the Fourier series is
1
f(t) = a 0 (a n cos n 0 t bn sin n 0 t )
2 n 1
An alternative but convenient way of writing the periodic function f(t) is in exponential form
with complex quantities. Since
e j n 0t e j n0t
cos n 0 t =
2
j n 0 t
e e j n0t
sin n 0 t =
2j
Substituting these quantities in the expression for the Fourier series gives
1
e j n0t e j n0t e j n0t e j n0t
f(t) = a 0 a n bn
2 n 1 2 n 1 2j
1 (a n jbn )e j n0t (a n jbn )e j n0t
= a0
2 n 1 2 jb
1
Here, taking cn = (an – jb n)
2
1
c-n = (an – jbn)
2
c0 = a0
Where c-n is the complex conjugate of cn. Substituting expressions for the coefficients an and bn
from Eqs 2.4 and 2.5 gives
T /2
1
T 0
cn = f (t )[cos n 0 t j sin n 0 t ] dt
T /2
1
f (t ) e
jn 0t
= dt
T 0
T /2
1
T T/ 2
and c-n = f (t )[cos n 0 t j sin n 0 t ] dt
T /2
1
f (t ) e
jn 0t
= dt
T 0
1
with f(t) = c0 + c n e j n 0 t
n 1
c e
n
n
j n 0 t
where the values of n are negative in the last term and are included under the sign. Also, c0
may be included under the sign by using the value of n = 0. Therefore,
f(t) = c e
n
n
j n 0 t
SIGNALS AND SYSTEMS
PARSEVAL’S IDENTITY FOR FOURIER SERIES
A periodic function f(t) with a period T is expressed by the Fourier series as
1
f (t ) a 0 ( a n cos n 0 t bn sin n 0 t )
2 n 1
1
Now, [ f (t )] 2 a 0 f (t ) [ a n f (t ) cos n 0 t bn f (t ) sin n 0 t ]
2 n 1
T/2 T /2
1 (a / 2)
Therefore, [ f (t )]2 dt 0
T T / 2 T T/ 2
[ f (t )]dt
1
T /2 T/2
n
a
T n 1 T / 2
f (t ) cos n 0 t dt b n f (t ) sin n 0 t dt
T / 2
From Eqns 2.2, 2.3 and 2.4, we have
T /2
2
T T/ 2
a0 f (t ) dt
T /2
2
T T/ 2
an f (t ) cos n 0 t dt
T /2
2
T T/ 2
bn f (t ) sin n 0 t dt
T /2
1
= cn f (t ) e
jn 0t
dt
n T T / 2
= c c
n
n n
c
2
= n , watts
n
SIGNALS AND SYSTEMS
From Eq. 2.12, the above equation becomes
2
a0 1
(a n2 bn2 ) c n
2
2 2 n 1 n 0
a0
Here, c0 and c n a n2 bn2 , (n 1)
2
Thus the power in f(t) is
P = …..+|c-n|2 + ……. + |c-1|2 + |c 0|2 + ……+ |cn|2 + ……….
P = |c 0|2 + 2|c1|2 + 2|c2|2 + ……. + |cn|2 + ………
Hence, for a periodic function, the power in a time waveform f(t) can be evaluated by adding
together the powers contained in each harmonic, i.e., frequency component of the signal f(t).
The power for the nth harmonic component at n 0 radians per sec is|cn|2 and that of –n
0 is|c-n|2 . For the single real harmonic, we have to consider both the frequency components
n 0 .
Here, cn = c-n and hence |cn|2 = |c-n|2. The power for the nth real harmonic f(t) is
Pn = |cn|2 + |c-n|2 = 2|cn|2
The effective or RMS value of f(t)
Using Eqns 2.12, 2.13 and 2.14, the RMS value of the function f(t) expressed by Eq. 2.1 is
2
a 1 1 1 1
Frms 0 a12 a 22 ...... b12 b22 ..............
2 2 2 2 2
1 1
c02 c12 c 22 ............
2 2
f (t )e
jt
dt
SIGNALS AND SYSTEMS
Inverse transform 1
F ( j )
2
jt
F ( j ) e d
( jt )
f (t )
f ( j `)d `
Time convolution f 1 (t ) * f 2 (t ) F1 ( j ) F2 ( j )
f ( ) f
1 2 (t )d
F(j )
2. f(t) e j 0 t 2 ( 0 ) 2
t 0 0
0
F(j )
3. f(t) cos 0 t [ ( 0 ) ( 0 )]
( )
1
0 t
-1 - 0 0 0
Eternal cosine
4. f(t)
sin 0 t F(j )
[ ( 0 ) ( 0 )] /j
j
0 t
- 0 0 0
Eternal sine
- /j
1 2
0
t
0
F(j )
6. f(t)
t 2
1 f (t ) sgn(t )
|t | j
t
0
-1 0
f(t) F(j )
7. u(t) 1
( )
j
1
0
0 t
f(t) F(j )
8. e at u (t ) 1
1/a
1 a j
0
0 t
f(t) F(j )
9. e a|t | 2a
1 a 2
2 2/a
1/a
0 1/a t -a 0 a
-1/a
Double exponential
SIGNALS AND SYSTEMS
a j
F(j )
10. f(t) e at
cos 0 t u (t )
( a j ) 2 02
0 t 0
- d d
f(t) |F(j )|
11. T T T
Au t u t AT Sinc
A 1
2 2 2
T
sin
T 2 -2 /T 0 2 /T
T
-T/2 0 T/2 t
2
f(t)
f (t ) 1, for | t | T sin(T )
F(j )
12.
2T 2T sin c (T )
1 0, otherwise T 2T
f (t ) u (t T ) u (t T )
-2 /T - /T
/T 2 /T
-T 0 T t
f(t) |F(j )|
13. |t | T
2
f (t ) A1 , | t | T
T sin 2 2
AT
A AT AT sin c
), elsewhere T
2 -4 /T -2 /T 0 2 /T 4 /T
-T 0 T t
f(t) F(j )
14. 0 t u ( 0 ) u ( 0 )
sin c 0 1
/
0
- 0 0 0
0 t
- / 0 / 0
Since pulse
f(t) F(j )
15.
2
2n
(t nT )
n T
n T
2_
T
-2T -T 0 T 2T t
Impulse train -4_ -2_ 0 2_ 4_
T T T T
f(t) F(j )
16. 1
f 2
(a/ ) 1/2
a 2
exp at
2
exp
a
1
0
0 t
Gaussian Pulse
SIGNALS AND SYSTEMS
17. f(t) t exp( at )u (t ) 1 |F(j )|
( a j ) 2 1/a
2
0
0 t
18. t n1 1
exp(at )u (t )
(n 1)! ( a j ) n
19. f(t) A exp( at ) sin ( 0 t ) u (t ) A 0 A 1 2 0- a
2 a + a2+ (2 0- a) 2
(a j ) 2 02 A 0
2 2
a+ 0
+a
0
0 t 0
A 1 2 0+a
2 a + a2+ (2 0- a) 2
(a j ) 2 02
0 t
0 - 0 0
Aa
2 2
a+ 0
j
or DTFT x(n) = X(e )
And Inverse Discrete – Time Fourier Transform (IDTFT) is expressed as
1
x(n) = X (e j )e j n d
2
or IDTFT X(e j ) = x(n)
(ii) From equation (4.15) and (4.17), it is clear that x(n) and X (e j ) are a Symbolically, this
may be expressed as
x(n) X (e j )
DTFT
2 X
1 ( ) X 2 (e j )d
n n
-3 -2 -1 1 2 3 -
x(n)= (n) j
X(e ) =1
j
2 A X(e )
1
1
0
n -
-L 0 L
j
3 x(n) X(e )
1
n - c c
- 0 c
c
4 1
a n
x(n) =
for n 0 X(ej ) =
for n 0 1 ae j
0
SIGNALS AND SYSTEMS
CHAPTER-3
S.No. f(t) F(s)
1. (t) 1
2. (t-a) e-as
3. u(t) 1
s
4. u(t – a) e as
s
5. tn 1
u(t), n positive integer (-1)n
n! s n 1
6. e-atu(t) 1
sa
7. t n e at 1
u(t)
n! s a n 1
8. sin ( 0 t) u(t) 0
s 02
2
9. cos ( 0 t) u(t) s
s 02
2
15. cos h 0 t s
s 02
2
16. e-atsin h 0 t 0
( s a) 2 02
17. e-atcos h 0 t sa
( s a ) 2 02
18. sin ( 0 t + ) s sin 0 cos
s 2 02
19. cos( 0 t + ) s cos 0 sin
s 2 02
t F (s) ds
s
f (t ) e
st
where F1(s) = dt
0
SIGNALS AND SYSTEMS
CHAPTER- 4
The analysis of any sampled signal or sampled data system in the frequency domain in extremely
difficult using s – plane representation because the signal or system equations will contain infinite long
polynomials due to the characteristic infinite number of poles and zeros. Fortunately, this problem may
be overcome by using the z-transform, which reduces the poles and the zeros to a finite number in the z
–plane.
The purpose of the z – transform is to map (transform) any point s = j in the s-plane to a
log z
corresponding point z (r ) in the z-plane by the relationship s , Where T is the sampling
T
period (seconds)
Under this mapping, the imaginary axis, = 0 maps on to the unit circle |z| = 1 in the z-plane. Also,
the left hand half –plane < 0 corresponds to the interior of the unit circle |z| = 1 in the z – plane.
This correspondence is shown in Fig.
Im(S) Im(Z)
+j
e Z(s)=esT
2 /T
|Z|=1
d
2 /T Re(S) c`
ba c + 1 a` Re(Z)
=0, s
b`=d`=e`/1 s .....
-j
j T
Mapping of s-plane for z=plane for z =e
Considering that the real part of x is zero, i.e. = 0, we have z = e jT = 1 jT , which gives the
values of z.
We know that the Laplace transform gives
x * (t ) X (s) = x(nT )e nsT
n0
But we have z = e for the z – transform and hence z-n = e-nsT, which corresponds to a delay of n
sT
In order to simplify the notation, the nth sample may be written as x(n) and hence the above equation
becomes
X(z) = x ( n)z n
n 0
Therefore, if r = 1, the z –transform evaluated on the unit circle gives the Fourier transform of
the sequence x(n).
SIGNALS AND SYSTEMS
The z – transform method is used to analyse discrete –time systems for finding the transfer
function, stability and digital network realizations of the system.
Example 4.1 Determine the z – transform for the analog input signal x(t) = e -at applied to a
digital filter.
Solution We know that
X * (t) = x(t ) (t nT )
n
For t = nT, the sampled signal sequence is
X * (nT) = [e0, e-aT,e-2aT, e-3aT, …….]
By applying Eq. 4.1 of X(z) = x(n) z-n, we get
n 0
X(z) = 1 + e-aTz-1 + e-2aT z-2 + e-3aTz-3 + …….
= e
n 0
anT
z-n = (e
n0
anT
z 1 ) n
1
Using infinite summation formula of a
k 0
k
1 a
, a 1, we get
1
X(z) =
a e aT z 1
which is the Fourier transform of the modified sequence [x(n)r -n]. If r = 1, i.e. |z| = 1, X(z)
reduces to its Fourier transform. The series of the above equation converges if x(n)r -n is
absolutely summable, i.e.
x ( n) r
n
n
If the output signal magnitude of the digital signal system, x(n), is to be finite, then the
magnitude of its z – transform, X(z), must be finite. The set of z values in the z-plane, for
which the magnitude of X(z) is finite, is called the Region of Convergence (ROC). That is,
convergence of x( n) r
n0
n
guarantees convergence of Eq. 4.1, where X(z) is a function of z -n.
Therefore, the condition for X(z) to be finite is |z| > 1. In other words, the ROC for X(z) is the
area outside the unit circle in the z-plane.
The ROC of a rational z-transform is bounded by the location of its poles. For example, the z –
z
transform of the unit step response u(n) is X(z) = which has a zero at z = 0 and a pole at z
z 1
= 1 and the ROC is |z| > 1 and extending all the way to , as shown in Fig.
SIGNALS AND SYSTEMS
Im(Z)
Re Z
Pole at z=1
Zero at z=0 ROC
x ( n) r
n
n
x( n) r n
n
x ( n) z n x ( n) z n
n n0
= x(n) z
n 1
n
x (n) z n
n0
-2 fm 0 2 fm
0 t -fm 0 fm f
(a) (b)
T8 (t)
x(t) g(t)
Multuplier
0 (t)
T8
t
(c) (d)
g(t) G(j )
A/Ts
0 2 fm 2
2 8 8
0 t
8 8
(e) 0
-2fs fm -fs2fs -fm
(f)
Fig. (a) Any continuous-time signal, (b) Spectrum of continuous-time signal,
(c) Impulse train as sampling function, (d) Multiplier, (e) Sampled signal
(f) Spectrum of sampled signal.
Nyquist Rate and Nyquist Interval
When the sampling rate becomes exactly equal to 2fm samples are second, then it is called
Nyquist rate. Nyquist rate is also called the minimum sampling rate. It is given by
fs = 2fm
Similarly, maximum sampling interval is called Nyquist interval. It is given by
1
Nyquist Interval Tz = seconds
2 fm
Therefore, the original spectrum X(j ) can be recovered from the sampled spectrum by using
a low pass filter with a cut – off frequency m.
G(j )
A/Ts
-3 8 -2 8 - 8
- m m 8 2 8 3 8
(a)
-fm 0 fm f
X(f)
(c)
-fm fm fs-fm
Fig: (a) Spectrum of original signal
(b) Spectrum of sample signal
(c) Amplitude response of practical low-pass filter.
The expression for sampled signal is written as
g(t) = x(t). Ts (t )
1
or g(t) = [x(t) + 2x(t) cos s t + 2x(t) cos 2 s t + ………]
Ts
SIGNALS AND SYSTEMS
From above equation, it may be observed that the sampled signal contains a component
1
g (t ) .
Ts
To recover x(t) or X(j ), the sampled signal must be passed through an ideal low-pass filter of
bandwidth of fm Hz and gain Ts.
Therefore, the reconstruction or interpolating filter transfer function may be expressed as
H(j ) = Ts rect
4f m
The impulse response h(t) of this filter is the inverse Fourier transform of H (j ).
h(t) = F-1[H(j )]
h(t) = F-1 Ts rect 4f
m
0 t
-2 fm 0 2 fm
(a) (b)
Sampled Reconstructed
Signal g(t) Signal x(t)
t
0
(c)
Energy Signal
an energy signal is one which has finite energy and zero average power. Hence, x(t) is an
energy signal if
0 < E < and P = 0
Parseval’s Theorem for Energy Signals
The Parseval’s theorem states that the energy of a signal may be obtained with the help of its
Fourier Transform.
1
X ( j ) d X ( j ) df X (t ) dt
2 2 2
E
2
Above equation is known as Parseval’s theorem for energy signals.
Energy Spectral Density
Let us consider a signal x(t) which is applied to an ideal low pass filter as shown in figure 3.88.
Y ( j ) = K ( j ) H ( j )
Here X ( j ) = Fourier transform of x(t)
and Y ( j ) = Fourier transform of y(t)
Also, the energy E0 of the output signal y(t) may be expressed as (using Parseval’s theorem)
1
2
E0 = Y ( j ) d
2
1
2
or, E0 = H ( j ). X (( j ) d
2
it may be observed that H ( j ) = 0 for all the frequencies except for the narrow band - m to m
for which it is unity.
Therefore,
1 m 1 m
2 2
E0 1. X ( j ) d X ( j ) d
2 m 2 m
Further, we may assume that Fourier transform X ( j ) is constant with frequency for a narrow
band - m to m . Therefore, the energy of the signal over this narrow band 2 m will be
1 m
. X ( j ) 1..d
2
E0 =
2 m
1 2
E0 = . X ( j ) ( 2 m )
2
1 2 2
E0 = . X ( j ) ( ) X ( j ) ( f )
2
SIGNALS AND SYSTEMS
Therefore, energy contribution per unit band-width will be
E0 2
X ( j )
f
2
Hence, X ( j ) represents energy per unit band-width and is known as Energy spectral
Density or Energy Density spectrum.
It is generally denoted by ( ) .
2
Hence ( ) = X ( j )
1
2
E= ( )d
Now, we can find the relationship between energy-densities of input and output (response) as
under:
Y(j ) = H(j ).X(j )
therefore, |y(j )|2 = |H(j )X(j )|2 = |H(j )|2 |X(j )|2
Now, let y ( ) be the energy spectral density of output y(t) and x ( ) be the energy spectral
density of input x(t), then
y ( ) =| Y(j )|2
and y ( ) =| X(j )|2
This is the relationship between input and output energy spectral densities.
Obtain the energy spectrum density of a Gate function of width and amplitude A.
A t 2
x(t) =
0 other wise
The Fourier transform is expressed as
X(j ) = x (t ) e
j t
dt
For a gate function, we have
j j
A 2
/2
Ae
j t
X ( j ) e e 2
/ 2
j
sin
2
or X(j ) = A A Sa
2
2
2
( ) X ( ) A 2 2 Sa 2
2
Power Signals
A power signal is one which has finite average power and infinite energy.
Hence, x(t) is a power signal if 0 < P < and E = where, P is the average power and E is
the energy of the signal.
SIGNALS AND SYSTEMS
1
n n n Cn
2
P C .(TC *
) C n .
C *
n
T n n
the average power P of the function x(t).
Therefore,
2
X t ( j )
P Lim df
2
X t ( j )
In the limit , the ratio may be approach a finite value.
Let this finite value be S ( )
2
X t ( j )
So that S ( ) Lim
1
P s( )df
2
s ( ) d
This equation relates probability distribution function and probability density function.
Integrating equation (i) in the limit form x1 to x2 we get
x2
x1
f
x2
x ( x)dx Fx ( x)
x1
P(x1 ≤ X ≤ x2) = f
x1
x ( x )dx ………….(vi)
SIGNALS AND SYSTEMS
From equations (ii) and (vi), the pdf completely characterizes the random variable. If the
probability density function of the random variable is known, then the probability of the event
s(x1 ≤ X ≤ x2) is the area under the pdf curve as shown in the fig.
Fx(x)
P{x 1<X<x 2}
x1 x2 x
Fig. Probability of an event from its
probability density function (pdf), Fx(x)
f
x ( x ) dx =1
(3) For a continuous random variable, the probability of the event P(X = x) is zero.
P(X = x = 0, for any real number x
Some Common pdfs
If the probability density function of a random variable defined over a given random
experiment is known, then the probability of an event associated with the experiment can be
determined.
(ii) The joint probability density function of a random vector consisting of N random variables
is given by
SIGNALS AND SYSTEMS
Fx1 , x2 xn ( x1 , x2 ,......xn )
fx1, x2, x3,……, xn =
x1 x2 ,..... xn
(iii) The joint probability density function can be used to determine the probabilities in the
following way.
x y
P[X ≤ x, Y ≤ x] = f
XY ( x, y ) dx dx
x2 y 2
f
XY ( x, y ) dx dy 1