Digital Signal Processing

Sampling &
Reconstruction
EEE & ECE Department
BITS-Pilani, Hyderabad campus

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Digital Signal Processing

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Digital Signal Processing

Sampling of Continuous time

signals
Discrete time signals in many
applications are generated by
sampling continuous time signals

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nn

ggpp(t)
(t) ggaa(t)p(t)
(t)p(t)
(nT)(t(t--nT)
nT)
ggaa(nT)
nn--

Since impulse is a periodic signal of period T, it

can be expressed as trigonometric Fourier series.
11
pp((tt))
((ttnT
nT)) 112cos
2cos
oott2cos2
2cos2
oott2cos3
2cos3
oott..........
..............

TT
nn

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11 ggaa(t)
(t)2g
2gaa(t)cos
(t)cos
oott2g
2gaa(t)cos2
(t)cos2
oott
ggpp((tt))ggaa((tt))pp((tt))

TT 2g
2gaa(t)cos3
(t)cos3
oott..........
..............

11 G
Gaa(j(j
))2G
2Gaa(j(j
)cos
)cos
oott2G
2Gaa(j(j
)cos2
)cos2
oott
G
Gpp((jj
))

(j

)cos3

..........
..
TT 2G
2Gaa (j)cos3oot ............

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Both the forms given below

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Bandpass Signal Sampling

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Aliasing
Timelimited signals have infinite bandwidth.
Because of infinite bandwidth the spectrum of the sampled signal consists of overlapping cycles regardless of the sampling rate.

Even if an ideal lowpass filter is used

the recovered signal will be distorted

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Two causes for distortion:

Loss of tail of F() beyond frequency > F s/2.
Due to folding frequency where the frequency
components greater than F s/2 impersonate
as components < Fs/2. e.g. Fs/2 + Fx appears
as Fs/2 Fx. This tail inversion is called
aliasing or spectral folding.

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Solution
Use antialiasing filter before sampling.
The antialiasing filter eliminates
components beyond the folding frequency
Fs/2 Hz. Thus components greater than
Fs/2 Hz will not appear in the spectrum of
the sampled signal as folding frequencies
and will not corrupt the spectrum.
We end up losing these frequency
components.
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Critical Sampling

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Under Sampling

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Over Sampling

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Problem

(1) A continuous time signal xa(t) is composed

of a linear combination of sinusoidal
signals of frequencies 300 Hz, 500 Hz, 1.2
kHz, 2.15 kHz and 3.5 kHz. The signal xa(t)
is sampled at a 2.0 kHz rate and the
sampled sequence is passed through an
ideal low pass filter with a cut-off
frequency of 900 Hz, generating a
continuous time signal of ya(t)
What are the frequency components
present in the output signal ?
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Recovery of The Signal

The discrete time signal must pass through
an analog lowpass filter.

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ADC

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DAC
Recovery of The Signal

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Recovery of The Signal

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Recovery of The Signal

FT

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Recovery of The Signal

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Recovery of The Signal

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Recovery of The Signal

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DAC
Recovery of The Signal
The zero-order hold is the hypothetical filter or LTI system
that converts the sequence of modulated Dirac impulses
xs(t)to the piecewise-constant signal

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DAC
Recovery of The Signal

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DAC
Recovery of The Signal
For the first order hold, the signal is reconstructed as a
piecewise linear approximation to the original signal that was
sampled.

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DAC
Recovery of The Signal

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Recovery of The Signal

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Recovery of The Signal

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Problem

The continuous-time signal

xa (t) = 4 sin (20t) 5 cos (24t) + 3 sin (120 t)
+ 2 cos (176 t)
is sampled at a 50 Hz rate, generating the
sequence x[n].
Determine the exact expression of x[n].

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