MIRPUR UNIVERSITY OF SCIENCE AND TECHNOLOGY

(MUST), MIRPUR

MIRPUR INSTITUTE OF TECHNOLOGY

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Communications Technology
ET-353

Lecture No.25: Sampling

Engr. Faisal Iqbal

Lecturer
Date: March 24, 2022

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Lecture Outline
Sampling
Sampling Theorem
Ideal Sampling
Sampling Rate
Nyquist Sampling Theorem
Nyquist Frequency and Aliasing

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 ADC
 Generally signals are analog in nature (eg:speech,weather signals).
 To process the analog signal by digital means, it is essential to convert them to discrete-
time signal , and then convert them to a sequence of numbers.
 The process of converting an analog to digital signal is ‘Analog-to-Digital Conversion’.
 The ADC involves three steps which are:
 1)Sampling
 2)Quantization 3)coding

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 TYPES OF SIGNALS
 Analog signals: continuous in time and amplitude
– Example: voltage, current, temperature,…
 Digital signals: discrete both in time and amplitude
– Example: attendance of this class, digitizes analog signals,…
 Discrete-time signal: discrete in time, continuous in
amplitude
– Example: hourly change of temperature in Austin

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 During sampling process, a continuous-time signal is converted into
discrete -time signals by taking samples of continuous-time signal at
discrete time intervals.

x(nTs)  x(t)
T=Sampling Interval
x (t)=Analog input signal

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 Sampling theorem gives the criteria for minimum number
of samples that should be taken.
 Sampling criteria:-”Sampling frequency must be twice of the
highest frequency”
fs=2W
fs=sampling frequency w=higher frequency content 2w also known as Nyquist rate

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 Nyquist rate is defined as the minimum sampling rate for the
perfect reconstruction of the continuous time signals from
samples.
 Nyquist rate=2*highest frequency component
 =2*W
 So sampling rate must be greater than or equal to nyquist rate

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 Sampling

Sampling theorem, Ideal sampling, Flat top sampling,

Natural sampling, reconstruction of signals from samples,
aliasing effect, up sampling and down sampling, discrete
time processing of continuous time signals

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 Sampling Theorem
A continuous time band limited signal can be represented by its samples, and can
be recovered from its samples,
provided that Sampling frequency s≥2m, (mmaximum frequency of signal)
The condition is referred as Nyquist criterion

t 0 Ts t
Sampling
Continuous time Discrete time
signal g(t) signal g(t)
Sampling frequency S

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 Ideal Sampling

Let continuous time band limited signal be

g(t)  G( ) s.t. G( )    0 t Ideal t
m 0 Ts
Continuous time signal g(t) Sampling Discrete time signal g(t
Let periodic impulse train be
 
 p (t)    (t  T )     (  n ) ; where   2
s s s s Ts Sampling p(t)
k  n
1
frequency S
Then Sampled signal g (t)  g(t)  p (t) 0 Ts
t

G()*[ s   (  ns )]
1
G ()  F[g  (t)]  Using multiplication property of FT
2 n

1 
or G () 
Ts n
G(  ns ) Using linearity property of FT and convolution property of impulse


Thus g  (t) 
1
G(  ns)
Ts n
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 Ideal Sampling

In Time domain: g(t) Signal G()

A

Sampling results in conversion of

t  -m m 
continuous time signal into discrete time p(t) Impulse train p()
1 s
signal
0 Ts 2Ts 3Ts
t  -2s -s 0 s 2s

g(t) Sampled signal
In Frequency domain: s> 2 m F[g(t)]
A/Ts

Sampling results in multiple translation of 0 Ts 2Ts 3Ts

t
 -2s -s 0 s 2s 

signal spectrum (linear combination of

shifted signal spectrum at integer
multiples of sampling frequency.

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 Ideal Sampling:
Reconstruction

g(t) Sampled signal Reconstruction Filter

g(t)
Low Pass filter 1
Cut-off m g(t)
0 Ts t Ts Amplifier
  m t
1 with gain Ts
1  H L( )  
g(t)  G(  ns )
Ts n
0 otherwise
1
G()
G()
Ts

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 Sampling Example: Musical CD
Audio frequency range is 20Hz-20KHz.
Musical CD consists of two channels of music (for stereo sound) sampled at 44.1 KHz
(oversampling satisfying Nyquist criterion) and quantized to 16 bit. Compute the data size of CD
for 70 minutes music.

CD data= 2 x (44.1 x 103) x 16 x 60 x70 bits = 740.88 MB

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 Sampling and Recording
When perform a measurement … a transducer converts the
measure and into an electrical signal … and this signal is
“sampled” using a digital computer
We normally record a continuous signal y(t) by a set of
samples ys(t) at discrete intervals of time t.

y(t) yS(t)

t t

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 Sampling Frequency

t
yS(t)

The number of samples recorded each second is

defined as the sampling frequency, fS.

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 Resemble Sampling Data
Original signal Sampled signal

If a signal is sampled and recorded relative rapidly, the

sampled data will closely resemble the original signal.

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 Under Sampling of Test Data
Original signal Sampled data

If we sampled too slowly, a recorded data will present a distortion from the
original signal. Such distortion will introduce some measurement errors.

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 Sampling and Hold

Almost any analog to digital converter will have some form of voltage “hold”
before sampling. A sampling and hold unit is used to hold each sample value
until the next pulse occurs. The sampling and hold unit is necessary because
the A/D converter requires a finite amount of time.

y(t) yS(t)

t t

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A 10 Hz Sine Wave Signal

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Sampling Rate ~ 5 Hz

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Sampling Rate ~ 11 Hz

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 Sampling Rate ~ 18 Hz

At a sampling frequency of 18 Hz, the reconstructed sine wave appears to be of 8 Hz sine wave. That is, the
frequency of the sine wave reconstructed from the sampled data is still different from that of the original
signal.
These incorrect frequencies that appear in the output data range are known as alias frequencies or aliases.
The alias frequencies are false frequencies that appear in the output data, that are simply artifacts of the
sampling process, and that do not (in any matter) occur in the original data.

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 Sampling Rate ~ 20.1 Hz

• Now, consider the reconstruction of the original signal based on the sampling rate of 20.1 Hz.
• The above figure shows the signal reconstructed from the data sampled at a frequency of 20.1 Hz. The
reconstructed wave has a frequency of 10 Hz, which is the same frequency as the original signal. However, the
amplitude of the reconstructed wave is lower than the original one.

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 Nyquist Sampling Theorem
A continuous signal can be represented by, and reconstituted from,
a set of sample values providing that the number of samples per
second is at least twice the highest frequency presented in the
signal.

f max is the signal frequency (or the maximum signal

frequency if there is more than one frequency in the signal)
f S is the sampling rate

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Nyquist Sampling Theorem

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Nyquist Sampling Theorem

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 Discussion Question

Nyquist sampling theorem tells us that the sampling frequency should be at

least twice the highest frequency presented in the signal to be able to resemble
the pattern of the original signal.
Use your common sense knowledge, to determine what is the proper sampling
frequency if you are requested to design a vibration monitoring system for a
passenger car. Assume that the natural frequency of the car system is 10 Hz,
the driving stimulated vibration on driver’s seat is 120 Hz, and the measurement
system natural frequency is 10 kHz.
What is the sampling frequency you will select?

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 Nyquist Frequency and Aliasing (1)

High frequency signal to be sampled by a low sampling rate may cause to “fold” the sampled data
into a false lower frequency signal. This phenomena is known as aliasing.

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Aliasing Example

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Higher Frequency Aliases

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 Nyquist Frequency and Aliasing (2)
Definitions:
Sampling Time, T: Total measuring time of a signal T = N t = N/fs [sec]

Sampling Interval t: Time between two samples

Sample Rate : 1/t, the number of samples per second

Nyquist Frequency, Fnyq: Maximum frequency that fs 1

fNyq   fs 
can be captured by a sample interval, t 2 t
Resolution Bandwidth: Minimum frequency that can be represented by a sample

1 f
f   s
Nt N

• The faster you sample, the higher frequency you can represent
• The longer you sample, the smaller the frequency represented
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 Frequency Resolution
• Duration of the sample T = N t = N/fs [sec]

• Minimum frequency that can be resolved is function of

sample length …. “bandwidth resolution”
1 fs
• Must capture a full period of the frequency f  
Nt N

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 Alias Frequency
A simple method to estimate alias frequencies is by using the folding diagram
shown below.

fm/ fN and

where, fN is the Nyquist (folding) frequency equal to half the sampling frequency fS.

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 Aliasing Formulas

fa  Alias frequency

fN  Folding frequency (Nyquist frequency)

fN  fs / 2
fa  ( fa / f N ) f N

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 Question 2
Given that sampling frequency equals 250Hz and the
ratio of folding frequency to alias frequency equals
0.5, find the alias frequency.

Solution:
f s  250
f a / f N  0.5
f N  250 / 2  125
f a  ( f a / f N ) f N  0.5 125  62.5Hz
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Any Question?

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