DiscreteTimeSignals (Basics)
DiscreteTimeSignals (Basics)
t = n T , n an integer
x(n) = x(nT )
n
-2 -1 0 1 2 3 11 12 13 14 15 16
2πf
Ω = ωT = (rad/sample)
fs
f
F = fT = (cycles/sample)
fs
x1(n) x 0 (n)
f1 = fs − f0
F1 = 1 − F0
and
∞
X
E = |x(n)|2 < + ∞
n=− ∞
respectively.
A DT signal is causal if
Hence
τ X sin ( nπτ
∞
T ) jnωs t
xS (t) = x(t) nπτ e
T n=− ∞ T
∞
τ X
= sinc(nfs τ )x(t)e jnωs t
T n=− ∞
f
fh 0 f h
f
-2f s -1/ τ -f s 0 fs 1/τ 2fs
(a)
f
-2f s -f s -f s /2 0 fs /2 fs 2fs
f
-2f s -f s -f s /2 0 fs /2 fs 2fs
(b)
where
∞
X
c(t) = δ(t − nT )
n=− ∞
Hence
∞
1 X
xS (t) = x(t)e jnωs t (1)
T n=− ∞
and, on taking the Fourier transform, the spectrum of the sampled
signal is given by
∞
X
XS (f ) = fs X (f − nfs )
n=− ∞
It is clear that
◮ all images of X (f ) are of equal magnitude
◮ if X (f ) is bandlimited such that the highest frequency
fh < f2s then X (f ) can be recovered by using an ideal low
pass filter (LPF) of the form (see also below)
f
Hi (f ) = rect( )
fs
If, however, fh > f2s then the images of X (f ) overlap giving
aliasing distortion.
fs ≥ 2fh
h i(t)
-4 -3 -2 -1 0 1 2 3 4
t/T
The filter output, denoted by xr (t), then is simply the sum of the
weighted and shifted impulse responses, i.e.
∞
X
xr (t) = x(nT )hi (t − nT )
n=− ∞
∞
1 X t − nT
= x(nT )sinc( )
T n=− ∞ T
∞
X
= fs x(nT )sinc(fs t − n)
n=− ∞
xr (t) = fs x(t)
h i(t-nT)
xr(t) = fs.x(t)
non-ideal LPF
X s(f) H(f)
f
0 fs /2 fs 2fs
t − T2
hz (t) = rect( )
T
Hence on applying the Fourier transform, the effective transfer
function of this filter is given by
ıωT
Hz (f ) = T sinc(fT)e− 2
Hz(f)
f
-2f s -f s -f s /2 0 fs /2 fs 2fs
Xr(f)
f
-2f s -f s -f s /2 0 fs /2 fs 2fs