Digital Signal Processing: b, where (a, b) can be (t t x π
Digital Signal Processing: b, where (a, b) can be (t t x π
Lecture 1 - Chapter 1
Discrete-time signals are defined at discrete-time instants and between the two discrete time instants are undefined but are not zero. They can be obtained either by sampling analog signals or they can be discrete in nature like discrete measurement signals.
A discrete-time signal having a set of discrete values is called a digital signal. Note that sampling an analog signal produces a discrete-time signal. Then quantization of its values produces a digital signal.
= A cos(2Ft + ) = 2F
<t <
2 Also, for complex exponential signals, x a (t ) = Ae j ( t+ ) . A sinusoidal signal then can also be expressed as
xa (t ) = A cos( t + ) =
1 1 Ae j( t + ) + Ae j ( t + ) 2 2
x (n ) = A cos( 0 n + ) = 2 f
To determine the period N of a periodic discrete time sinusoid, we express f as two relatively prime numbers. Observe that a small change in frequency can result in a 1 large change in period. For example, x1 (n ) = A cos 2 n + = A cos( n + ) its 2 period is N1 = K 30 =2= . f1 60
1 1 f or equivalently to or in 2 2 1 or = + . To see 2
what happens for 2 consider 1 = 0 and 2 = 2 - 0 . When 1 varies between to 2, then 2 varies between and 0. Now x1 (n ) = A cos1 n = A cos 0 n
3 x2 (n) = A cos( 2 )n = A cos(2 0 )n =* A cos( o n ) = A cos on = x1 (n ) * this is only true because n is an integer value, i.e., x(n) is a discrete signal. Hence, x2 (n ) = x1 (n) is an alias because < F < + maps only to by sampling.
1 1 f 2 2
Sampling:
t = nTs =
1 1 f 2 2
Nyquist rate
1 F 1 f = 2 Fs 2
Example 1:
xa1 (t ) = cos 20 t
xa 2 (t ) = cos 100 t F2 = 50 Hz
F1 = 10 Hz
20 n x1 (n ) = xa1 (nTs ) = xa1 = cos n = cos n 40 2 40 100 5 n x2 (n ) = xa 2 = cos n = cos n = cos 2 + n 40 2 2 40 = cos n = x1 (n )!! 2
Therefore, by this sampling rate x2(n) has become same as x1(n), which is an aliasing error. Equivalently, in this case, the frequency of 50 Hz is an alias of 10 Hz by sampling
4 rate of 40 Hz. Furthermore, all frequencies (F1 + 40K) are aliases of F1. Hence, do not use the Nyquist rate blindly. Example 2: xa (t ) = 3 cos 50 t + 10 sin 300 t cos 100 t
14 4 2 3
x1
14243 1 24 4 4 4 3
x2 x3
Nyquist rate? f1 = 25, f2 = 150, f3 = 50 Hz Fmax = 150 Hz Fs = 300 Hz?! Problem: with Fx = 300 Hz, x2 (n ) = xa 2 = 10 sin n = 0 all the time! F s If it had a phase shift o < < n , then it would have been fine, but it is best to choose a higher sampling rate.
Sampling
xa (t )
xa(t)
x
P(t)
x(n)
Ts
x(n)
x p (t ) = x a (t ) p (t )
p(t ) = (t nTs )
x (n ) = x p (t ) | t = nT = x (nTs ) (t nTs )
x
1 [ X a ( )* P ( )] 2 2 + P( ) = ( k s ) Ts Ks X p ( ) = X p ( ) = 1 Ts
X ( k )
a s
5 X ( ) Obviously, if s 2 M there is no
Xp()
In practice however, generating very narrow impulse is very difficult. Therefore, the practical way for sampling is zero-order hold. Such a system samples xa(t) at a given sampling instant and holds that value until the succeeding sampling instant. Reconstruction of xr(t) from the output of this system requires a cascade of low-pass filters or a non-constant gain of LPF.
P(t)
H0(t)
It is as if x a (t )
|Hr()|
xo (t )
xp(t)
-s /2
s /2 <Hr() xr(t)
hr (t )
-s /2
H r ( )
2
s /2