JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY KAKINADA

KAKINADA – 533 001 , ANDHRA PRADESH

GATE Coaching Classes as per the Direction of

Ministry of Education
GOVERNMENT OF ANDHRA PRADESH

Analog Communication
26-05-2020 to 06-07-2020

Prof. Ch. Srinivasa Rao

Dept. of ECE, JNTUK-UCE Vizianagaram
 Analog Communication-Day 9, 03-06-2020
Presentation Outline

❑ Sampling Theorem
❑ Methods of Sampling
❑ Significance of Sampling Rate
❑ Anti-aliasing Filter
❑ Applications of Sampling Theorem – PAM/TDM
❑ problems

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 Learning Outcomes

• At the end of this Session, Student will be able to learn:

• LO 1 : Sampling and different Types of Sampling
• LO 2 : Sampling of Baseband Signals
• LO 3 : Sampling of Bandpass Signals

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 Sampling
Sampling is the processes of converting continuous-time analog signal, xa(t), into a
discrete-time signal by taking the “samples” at discrete-time intervals.
• Sampling analog signals makes them discrete in time but still continuous valued
• If done properly (Nyquist theorem is satisfied), sampling does not introduce
distortion.
Sampled values:
The value of the function at the sampling points.
Sampling interval:
• The time that separates sampling points (interval b/w samples), Ts
• If the signal is slowly varying, then fewer samples per second will be required
than if the wave is rapidly varying
• So, the optimum sampling rate depends on the maximum frequency component
present in the signal
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Analog-to-digital conversion is (basically) a 2 step process:
1.Sampling:
Convert from continuous-time analog signal xa(t) to discrete-time
continuous value signal.
Is obtained by taking the “samples” of xa(t) at discrete-time intervals, Ts
2.Quantization:
Convert from discrete-time continuous valued signal to discrete time
discrete valued signal

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 Applications

There are few applications of sampling theorem are listed below. They are
❑ To maintain sound quality in music recordings.
❑ Sampling process applicable in the conversion of analog to discrete form.
❑ Speech recognition systems and pattern recognition systems.
❑ Modulation and demodulation systems
❑ In sensor data evaluation systems
❑ Radar and radio navigation system sampling is applicable.
❑ Digital watermarking and biometric identification systems, surveillance
systems.

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 Sampling
Sampling Rate (or sampling frequency fs):
The rate at which the signal is sampled, expressed as the number of samples per
second (reciprocal of the sampling interval), 1/Ts = fs
Where Ts is the Sampling Time
Nyquist Sampling Theorem (or Nyquist Criterion):
the sampling is performed at a proper rate, no info is lost about the original signal
and it can be properly reconstructed later on
Statement:
If a signal is sampled at a rate at least, but not exactly equal to twice the max
frequency component of the waveform, then the waveform can be exactly
reconstructed from the samples without any distortion”.

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 Sampling theorem for Band limited Signals
If an analog information signal containing no frequency outside the specified
bandwidth W Hz, it may be reconstructed from its samples at a sequence of
points placed 1/(2W) seconds apart with zero-mean squared error.

The minimum sampling The reciprocal of Nyquist rate, 1/(2W), is called

rate of (2W) samples per the Nyquist interval, that is, Ts= 1/(2W).
second, for an analog The phenomenon of the presenceof
signal bandwidth of W high-frequency component in the
Hz, is called the Nyquist spectrum of the originalanalog signal is called
aliasing or simply foldover.
rate.

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 Sampling Theorem for Baseband Signal
Sampling Theorem for Baseband S ignal - A baseband signal having no frequency
components higher than fm Hz may be completely recovered from the
knowledge of samples taken at a rate of at least 2 fm samples per second, that is,
sampling frequency fs ≥ 2 fm.

A baseband signal having no frequency

The minimum sampling components higher than fm Hz is completely
described by its sample values at uniform
rate fs = 2 fm samples per
intervals less than or equal to 1/(2fm) seconds
second is called the apart, that is, the sampling interval Ts
Nyquist sampling rate. ≤1/(2fm) seconds.

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 Methods of Sampling

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 Ideal Sampling

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 Ideal Sampling ( or Impulse Sampling)
It is accomplished by the multiplication of the signal x(t) by the uniform train
of impulses (comb function)

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 Ideal Sampling ( or Impulse Sampling)
This shows that the Fourier Transform of the sampled signal is the Fourier
Transform of the original signal at rate of 1/Ts

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 Ideal Sampling ( or Impulse Sampling)
As long as fs> 2fm,no overlap of repeated replicas X(f- n/Ts) will occur in Xs(f)
Minimum Sampling Condition:

Sampling Theorem: A finite energy function x(t) can be completely

reconstructed from sampled value x(nTs) with

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 Ideal Sampling ( or Impulse Sampling)
This means that the output is simply the replication of the original signal at
discrete intervals, e.g

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 Ideal Sampling ( or Impulse Sampling)
Ts is called the Nyquist interval: It is the longest time interval that can be used for sampling
a band limited signal and still allow reconstruction of the signal at the receiver without
distortion

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 Natural Sampling

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 Natural Sampling
If we multiply x(t) by a train of rectangular
pulses xp(t), we obtain a gated waveform
that approximates the ideal sampled
waveform, known as natural sampling
gating
x s (t ) = x(t ).x p (t )

= x(t ).  c n e j 2nf s
n = −

X s ( f ) = FT [ x(t ).x p (t )]

= n
c FT
n = −
[ x (t ) e j 2nf s
]

= c
n = −
n X [ f − nf s ]

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 Natural Sampling
• Each pulse in xp(t) has width Ts and amplitude 1/T
• The top of each pulse follows the variation of the signal being
sampled
• Xs (f) is the replication of X(f) periodically every fs
• Xs (f) is weighted by Cn : Fourier Series Co-effiecient
• The problem with a natural sampled waveform is that the tops of the
sample pulses are not flat.
• It is not compatible with a digital system since the amplitude of each
sample has infinite number of possible values
• Another technique known as flat top sampling is used to alleviate this
problem
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 Flat-top Sampling

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 Flat-top Sampling
Here, the pulse is held to a constant height for the whole sample period
Flat top sampling is obtained by the convolution of the signal obtained
after ideal sampling with a unity amplitude rectangular pulse, p(t)
This technique is used to realize Sample-and and-Hold (S/H) operation

S/H, input signal is continuously sampled and then the value is held for
as long as it takes to for the A/D to acquire its value.

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 Flat-top Sampling

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 Flat-top Sampling

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 Flat-top Sampling

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 Recovering the Analog Signal
• One way of recovering the original signal from sampled signal Xs(f) is to
pass it through a Low Pass Filter (LPF) as shown below

If fs > 2B then we
recover x(t) exactly
Else we run into some
problems and signal is
not fully recovered

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 Significance of Sampling Rate

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• Under sampling and Aliasing
If the waveform is under sampled (i.e. fs < 2B) then there will be
spectral overlap in the sampled signal

• The signal at the output of the filter will be

different from the original signal spectrum
This is the outcome of aliasing!
• This implies that whenever the sampling condition is not met, an
irreversible overlap of the spectral replicas is produced
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This could be due to:
1. x(t) containing higher frequency than were expected
2. An error in calculating the sampling rate Under normal conditions, under
sampling of signals causing aliasing is not recommended
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 Solution 1: Anti-Aliasing Analog Filter
• All physically realizable signals are not completely bandlimited
• If there is a significant amount of energy in frequencies above half the
sampling frequency(fs/2), aliasing will occur.
• Aliasing can be prevented by first passing the analog signal through an
anti-aliasing filter (also called a prefilter) before sampling is performed.
• The anti-aliasing filter is simply a LPF with cutoff frequency equal to half
the sample rate.

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 Anti aliasing Filter

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 Solution 2: Over Sampling and Filtering in the Digital Domain
• The signal is passed through a low performance (less costly) analog low
the bandwidth.
• Sample the resulting signal at a high sampling frequency.
• The digital samples are then processed by a high performance digital
filter and down sample the resulting signal.

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 Summery of Sampling
Ideal Sampling
(or Impulse Sampling)

Natural Sampling 
(or Gating) = x(t ).  c n e j 2nfs
n = −

Flat-Top Sampling

For all sampling techniques

If fs > 2B then we can recover x(t) exactly
Wednesday, June 3, 2020
If fs < 2B) spectral overlapping known as aliasing will occur
Prof.Ch.Srinivasa Rao, JNTUK-UCEV 33
 Applications of Sampling theorem-
PAM/TDM
• Design of PAM/TDM System

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 Key Points
∞

1. 𝑋𝑠 𝑓 = 𝑓𝑠 ෍ 𝑋(𝑓 − 𝑛 𝑓𝑠 )
𝑛=−∞
2.fs >2𝑓𝑚 this condition is called over sampling and Guard band will be present
in it
Guard band=𝑓𝑠 − 2𝑓𝑚
Reconstruction filter Used: practical Low pass Filter
3. 𝑓𝑠 = 2𝑓𝑚 (Nyquist rate)
Reconstruction filter Used: Ideal Low pass Filter
4. 𝑓𝑠 < 2𝑓𝑚
Aliasing effect or fold over error
5. X(t)=𝑥1 𝑡 + 𝑥2 𝑡 + ⋯ … … … … … . 𝑥𝑁 𝑡
Where N indicates No of signals
Then sampling frequency 𝐹𝑠 = 2 ∗ max(𝑓1 , 𝑓2 , 𝑓3 , … … … … 𝑓𝑁 )

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6. X(t)=𝑥1 𝑡 × 𝑥2 𝑡 × ⋯ … … … … … .× 𝑥𝑁 𝑡
Where N indicates No of signals
Then sampling frequency 𝐹𝑠 = 2 ∗ sum 𝑓1 , 𝑓2 , 𝑓3 , … … … … 𝑓𝑁
7. X(t)=𝑥1 𝑡 ∗ 𝑥2 𝑡 ∗ ⋯ … … … … … .∗ 𝑥𝑁 𝑡 (* denotes convolution)
Where N indicates No of signals
Then sampling frequency 𝐹𝑠 = 2 ∗ max(𝑓1 , 𝑓2 , 𝑓3 , … … … … 𝑓𝑁 )
8.Frequency components present in sampled version of single tone
signal is
k𝑓𝑠 ± 𝑓𝑚 where k=0,1,2…………….N
Where N indicates No of signals
i.e. 𝑓𝑚 , 𝑓𝑠 ± 𝑓𝑚 , 2𝑓𝑠 ± 𝑓𝑚 , … … … … … … . .

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 Problems on Sampling
1.

[GATE 2017: 2Marks]

[GATE 2017: 2Marks]

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❑ Solution

fc+ fm
-7 2 11
fc- fm 7 16 25
fm 2fc- fm 2fc+ fm
-fm

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2.A band limited low-pass signal x(t) of bandwidth 5 kHz is sampled
at a sampling rate fs .The signal x(t) is reconstructed using the
reconstruction filter H(f) whose magnitude response is shown below:

The minimum sampling rate (in kHz) for perfect reconstruction of is

_________. (a) 1.5
(b) 2.5
(c) 13
(d) Null
[GATE 2018: 2Marks]
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❑ Solution

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3.A continuous-time function x(t) is periodic with period T. The
function is sampled uniformly with a sampling period Ts. In which
one of the following cases is the sampled signal periodic?
(a) T=√2 Ts
(b) T=1.2 Ts
(c) Always
(d) Naver
[GATE 2016: 1Mark]

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Solution

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4.Consider two real valued signals, x(t) band-limited to [−500 Hz,
500Hz] and y(t) band-limited to [−1kHz, 1kHz]. For z (t) = x(t). y(t),
the Nyquist sampling frequency (in kHz) is __________
(a) 2
(b) 3
(c) 4
(d) 6
[GATE 2014: 1Mark]

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Solution

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5.A band-limited signal with a maximum frequency of 5 kHz is to be
sampled. According to the sampling theorem, the sampling frequency
which is not valid is
(a) 5KHZ
(b) 12KHZ
(c) 15KHZ
(d) 20KHZ
[GATE 2014: 1Mark]

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Solution

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6.Benefit(s) of Bandpass sampling over low pass sampling

(a) It reduces speed requirement of A/D convertor

(b) Increase the amount of digital memory necessary to capture a
given interval of signal.
(c) Both (A) and (B) are correct
(d) Both (A) and (B) are incorrect
[ISRO 2018: 1Mark]

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Solution

Bandpass sampling require less bandwidth to reconstruct the signal

compare to low pass sampling. Bandpass sampling time is more i.e.
speed requirement is less because it will store less sample compared
to low pass sampling . it will decrease the memory requirement ,
because it store less sample when compared to low pass sampling
Practice

**The Correct Answer Among All the Options is A**

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 Additional Problems on Sampling

1.𝑥1 (𝑡) is bandlimited to 2 KHz and 𝑥2 (𝑡) is bandlimited to 3 Khz

Find sampling frequency of
a) 𝑥1 (𝑡)
b)𝑥2 (𝑡)
c) 𝑥1 (2𝑡)
d)𝑥2 𝑡 − 3
e) 𝑥1 (𝑡)+𝑥2 (𝑡)
f) 𝑥1 (𝑡) 𝑥2 (𝑡)

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 Solution
a) 𝑥1 (𝑡) F.T 𝑥1 (𝑓)
𝑠𝑖𝑛𝑐𝑒 𝑥1 (𝑡) is bandlimited to 2 KHz
𝑓𝑚 = 2Khz
Therefore 𝑓𝑠 = 2*2 KHz =4 Khz

𝑏) 𝑥2 (𝑡) F.T 𝑥2 (𝑓)

𝑠𝑖𝑛𝑐𝑒 𝑥2 (𝑡) is bandlimited to 3 KHz
𝑓𝑚 = 3 Khz
Therefore 𝑓𝑠 = 2* 3KHz =6 Khz

1 𝑓
𝑐) 𝑥1 (2𝑡) F.T 𝑥( )
2 2

Compressed by Expanded by a
a factor of 2 in factor of 2 in
time domain frequency domain

Since the signal is expanded by a factor of 2 , 𝑓𝑚 = 4 Khz

Therefore 𝑓𝑠 = 2* 4 KHz =8 Khz

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 𝑑) 𝑥2 𝑡 − 3 F.T 𝑒 −𝑗∗2∗𝑝𝑖∗3 𝑥2 (𝑓)
Phase does not effect the sampling frequency and 𝑥2 (𝑡) is bandlimited to 3 KHz
𝑓𝑚 = 3 Khz
Therefore 𝑓𝑠 = 2* 3KHz =6 Khz
𝑥1 (𝑡)+𝑥2 (𝑡) F.T 𝑥1 (𝑓)+𝑥2 (𝑓)
Concept to be remembered
X(t)=𝑥1 𝑡 + 𝑥2 𝑡 + ⋯ … … … … … . 𝑥𝑁 𝑡
Where N indicates No of signals
Then sampling frequency 𝐹𝑠 = 2 ∗ max(𝑓1 , 𝑓2 , 𝑓3 , … … … … 𝑓𝑁 )
𝐹𝑠 = 2 ∗ max(2 𝐾𝐻𝑧, 3 𝐾𝐻𝑧)=6 Khz
𝑒) 𝑥1 (𝑡) 𝑥2 𝑡 F.T 𝑥1 𝑓 ∗ 𝑥2 (𝑓)
where * denotes convolution

𝐹𝑀 𝑖𝑠 𝑏𝑎𝑛𝑑𝑙𝑖𝑚𝑖𝑡𝑒𝑑 𝑡𝑜 5 𝐾𝐻𝑧

𝐹𝑠 =2*5KHz=10 KHz
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 Problem 3
Find Nyquist sampling interval for the signal sinc(700t)+sinc(500t) is
Concept to be remembered
X(t)=𝑥1 𝑡 + 𝑥2 𝑡 + ⋯ … … … … … . 𝑥𝑁 𝑡
Where N indicates No of signals
Then sampling frequency 𝑓𝑠 = 2 ∗ max(𝑓1 , 𝑓2 , 𝑓3 , … … … … 𝑓𝑁 )

Signal is given as
sinc(700t)+sinc(500t)

sin(700∗𝑝𝑖∗𝑡) sin(500∗𝑝𝑖∗𝑡)
= +
700∗𝑝𝑖∗𝑡 2∗𝑝𝑖∗𝑡

2*pi*𝑓𝑚1 = 700 ∗ 𝑝𝑖
𝑓𝑚1 =350
2*pi*𝑓𝑚2 = 500 ∗ 𝑝𝑖
𝑓𝑚2 =250
Then sampling frequency 𝐹𝑠 = 2 ∗ max(350,250)
=2*350=700Hz

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Problem 4
The Nyquist sampling frequency (in Hz)of a signal 6 x 104 𝑠𝑖𝑛𝑐 2 400𝑡 ∗ 106 ∗ 𝑠𝑖𝑛𝑐 3 100𝑡 𝑖𝑠

Concept to be remembered
1.Sampling frequency of
𝑓𝑠 𝑥 𝑛 𝑡 = 𝑛 ∗ 𝑓𝑠 (𝑥 𝑡 )
2.X(t)=𝑥1 𝑡 ∗ 𝑥2 𝑡 ∗ ⋯ … … … … … .∗ 𝑥𝑁 𝑡
Where N indicates No of signals
Then sampling frequency 𝐹𝑠 = 2 ∗
sum(𝑓1 , 𝑓2 , 𝑓3 , … … … … 𝑓𝑁 )
2
2
sin(400 ∗ 𝑝𝑖 ∗ 𝑡)
𝑠𝑖𝑛𝑐 400𝑡 =
400 ∗ 𝑝𝑖 ∗ 𝑡
sin(400∗𝑝𝑖∗𝑡)
= 𝑓𝑠 ( )=> 2*pi*𝑓𝑚1 = 400 ∗ 𝑝𝑖
400∗𝑝𝑖∗𝑡
𝑓𝑚 =200
𝑓𝑠 𝑥 𝑛 𝑡 = 𝑛 ∗ 𝑓𝑠 (𝑥 𝑡 )
𝑓𝑚1 =2* 𝑓𝑚
𝑓𝑚1 =2*200 Hz=400Hz
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 3 sin(100∗𝑝𝑖∗𝑡) 3
𝑠𝑖𝑛𝑐 100𝑡 =
100∗𝑝𝑖∗𝑡
sin(100∗𝑝𝑖∗𝑡)
𝑓𝑠 ( )=2*pi*𝑓𝑚2 = 100 ∗ 𝑝𝑖
100∗𝑝𝑖∗𝑡
𝑓𝑚 =50 Hz
𝑓𝑚2 =3* 𝑓𝑚 = 3 ∗ 50 = 150 𝐻𝑧

Then sampling frequency 𝐹𝑠 = 2 ∗

sum 𝑓1 , 𝑓2 , 𝑓3 , … … … … 𝑓𝑁 = 2 ∗ (400 +
150) = 2 ∗ 550 = 1100 ℎ𝑧

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 Problem 5

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 Problem 6

Sol:
This is the case of band pass sampling
𝐹𝐻 = 1800 𝐻𝑧
𝐹𝐿 = 300 𝐻𝑧
Band width = 𝐹𝐻 - 𝐹𝐿 =1800-300=1500 Hz
Now according to Bandpass sampling theorem
𝑓𝐻 1800
K=I𝑛𝑡𝑒𝑔𝑒𝑟 𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ(𝐵) =1500 = 1.2 = 1
2𝑓 2x1800
minimum sampling rate= 𝑓𝑠 = 𝐾𝐻= 1 =3600 samples/sec

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References
❑Communication Systems by Simon Haykin, Wiley, 2nd Edition.
❑Principle of Communication System by Taub ,Schilling & Saha,
TMH.
❑Modern digital and Analog Communications system by BP Lathi,
Ding and Gupta, Oxford.
❑Electronic Communication Systems by Kennedy and Davis, TMH.
❑Communication Systems Analog and Digital by Singh and Sapre,
TMH

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Wednesday, June 3, 2020 Prof.Ch.Srinivasa Rao, JNTUK-UCEV
JNTUK UCEV 61