JSPM’s

RAJARSHII SHAHU COLLEGE OF ENGINEERING

(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune)

DEPARTMENT
OF

ELECTRONICS AND TELECOMMUNICATION

ENGINEERING
[EC2207T]: Communication Systems

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

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 Course Contents

UNIT-II Digital Transmission of Analog Signal 05 Hours

Sampling theorem,
Proof of Sampling Theorem,
Sampling types,
Aliasing,
Aperture Effect,
Quantization,
Quantization Types,
PCM Generation,
Commanding, Delta Modulation, ADM.
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 Why Sampling is Required ?
• Sampling plays an essential role in digital communication systems because it
turns continuous analog signals into discrete digital data, allowing them to be
 processed,
 transmitted,
 stored, and
 manipulated efficiently
in the digital world.
• Noise reduction, error detection and correction, compression, signal processing,
and interoperability are all enabled by this conversion, which is crucial for
modern communication systems.
• Digital representation provides for long-distance data transmission with reduced
signal deterioration, as well as precise modulation, demodulation, and other
signal processing processes, facilitating dependable communication and
compatibility among diverse devices and platforms.
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 Sampling in Digital Communication

• Sampling in digital communication is converting a continuous-time signal

into a discrete-time signal.

• The sampling process includes the following steps:

1. The continuous signal is taken as an input.

2. Sampling is performed to convert this signal into a digital representation.

3. In addition to sampling, quantization of a signal is performed.

4. After the above step, encoding of the signal is done.

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 sampling

https://www.geeksforgeeks.org/sampling-in-digital-communication/#sampling-process-in-
digital-communication
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 Important Terminologies of Sampling
in Digital Communication

There are few important terminologies of Sampling in Digital

Communication discussed below :
•Sampling
•Sample
•Sampling Rate or Sampling Frequency
•Nyquist Rate
•Nyquist Interval
•Quantization

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 Important Terminologies of Sampling
in Digital Communication
1. Sampling
It is the process by which, we convert CTS (continuous time signal) into
DTS (discrete time signal) by taking the signal values at some distinct
points in time, meaning that this is used to take samples of analog signals
at some points in time (regular or irregular)

2. Sample
It can be defined as the numeric value of an analog signal at a specific
time. It is just the signal's measured amplitude at a particular time and
converting it to a digital representation.
3. Sampling Rate or Sampling Frequency
It refers to the number of samples or data points taken per unit of time from
an analog signal to convert it into a digital format. It is also known as
sampling frequency. It is measured in Hertz (Hz).

The formula for sampling rate or sampling frequency is given by:

where,
Sampling Rate= 1/Ts​​ = fs Ts = sampling time
fs = sampling frequency
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 Important Terminologies of Sampling
in Digital Communication
4. Nyquist Rate

It is the minimum sampling rate required to accurately capture an analog

signal in digital form without information loss.
It is also known as Nyquist Frequency or Nyquist Limit.

It is defined as twice the maximum frequency component present in the

analog signal.
Mathematically it can be represented as:

fs=2fmax
where,
fs = Sampling Rate or Nyquist Rate (Hz)
fmax = Maximum frequency Department
component (Hz)
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 Important Terminologies of Sampling
in Digital Communication

5. Nyquist Interval

The Nyquist interval, also known as the Nyquist period, is the time interval
between consecutive samples in a digital signal or digital sampling system.

It is the reciprocal of the Nyquist rate, which is the smallest sampling rate
required to accurately capture an analog signal in digital form without
information loss.

Mathematically it can be represented as:

T=1/ Nyquist Rate
Where,
T=Nyquist interval (sec)
Nyquist Rate is the sampling rate (Hz)

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 Important Terminologies of Sampling
in Digital Communication

6. Quantization

It is the process to represent a continuous-valued signal with a limited set of

discrete values.
In other words, it involves mapping a continuous signal's infinite range of potential
values to a finite collection of discrete values.

Quantization of a Signal

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 Nyquist - Shannon Sampling Theorem

The theorem states that for reconstructing a sampled signal

accurately from the available samples, the sampling
frequency should be at least twice as much as the highest
frequency component of the signal.
It can be understood by the following expression:
2×fmax⩽fs
Where,
fmax = maximum frequency component of the original signal
fs= sampling frequency

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 Books
"A signal can be exactly reproduced if it is sampled at
the rate fs which is greater than twice the maximum
frequency (f_{max} ).”

•If fs<2×fmax , then it is the

case of under sampling.

•If fs=2×fmax , then it is the

case of sampling at Nyquist
Rate and it is perfect
sampling.

•If fs>2×fmax , then it is the

case of oversampling.

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 Oversampling & Undersampling

1. Undersampling

The spectra of X(ω) are overlapped in this scenario because the sampling
rate is less than the Nyquist rate, making it impossible to extract the original
signal from the sampled signal.
Because the spectra overlap, some frequency components of the original
signal will acquire a new frequency; this process is known as
frequency Aliasing.
An application of undersampling can be when the high frequency
components are not useful and user needs to reduce the amount of data being
processed.

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 Nyquist - Shannon Sampling Theorem

2. Oversampling

Over-sampling is when more samples are taken that are necessary to

capture the signal's frequency.
It can be done to measure more accurately, enhancing SNR, providing
more detailed information for further processing. It can be seen in the
above figure-'Sampling cases'.

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 Nyquist - Shannon Sampling Theorem

Aliasing
• It is a phenomenon that occurs when a high-frequency signal is represented
at lower frequency.
• Means it occurs when the sampling rate is insufficient and fails to capture the
signal properly.
• When the signals are sampled at lower frequency than the nyquist frequency,
high frequency components fold back (gets aliased) in the low frequency
range. This may lead to distorted signal representation.
• In simple words a high frequency component of a signal taking the identity of
low frequency component of a signal when it is undersampled.

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18
19
 Methods to Avoid Aliasing

It can be avoided by the following two methods :

1.Sampling at Nyquist Rate

2.Using Anti-Aliasing Filter (Low Pass Filter): It helps

removing the component above the nyquist frequency which
may lead to aliasing.

What is Anti-Aliasing Filter?

An anti-aliasing filter, often known as a "anti-alias filter" or simply "AAF,"
is a filter used in signal processing and digital data collection systems to
prevent or eliminate aliasing effects.

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 Anti-Aliasing Filter

Working Anti-Aliasing Filter

1. Filtering High-Frequency Components: The anti-aliasing filter is used to
remove high-frequency components in analog signals that exceed the Nyquist
frequency, which is half the sampling rate of the ADC. If these high-frequency
components are not filtered, they will cause aliasing which results in incorrect
information.
2. Preventing Aliasing: The anti-aliasing filter ensures that only the desired
frequency are represented in the digital signal by attenuating the high-
frequency components.
3. Improved Signal Quality: It improves signal quality and allows for more
accurate data gathering. It helps to retain the original signal's integrity and
decreases the possibility of errors in later digital processing.

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Nyquist - Shannon Sampling Theorem

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Nyquist - Shannon Sampling Theorem

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Nyquist - Shannon Sampling Theorem

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 Nyquist - Shannon Sampling Theorem
There are three types of sampling techniques:

•Impulse sampling

•Natural sampling

• Flat Top sampling.

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 Methods of Sampling

1. Ideal Sampling
Concept: Ideal sampling, also known as impulse or Dirac sampling, is a
theoretical notion in which samples of a continuous signal are taken at specific
time intervals, often at the Dirac delta function impulse points.
Sampling Process: Each sample in perfect sampling is an impulse or delta
function at the sampling instant. The sampled signal can be represented
mathematically as the product of the continuous signal and the Dirac delta
function.

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Methods of Sampling

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 Nyquist - Shannon Sampling Theorem

Reconstruction: The reconstruction of the original signal from ideal

samples, we can use interpolation which uses the functions. Ideal
sampling is a simple approach to express and analyze sampling theory,
however it is not practical due to the requirement for infinite bandwidth.

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 Ideal Sampling

We cannot use this practically because

 pulse width cannot be zero and
 the generation of impulse train is not possible
practically.
 Poor SNR
 BW is large

Disadvantages
 Samples are very small, SNR is low
 No Ideal filter available to cut off the signal

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 Sampling
2. Natural Sampling
Concept: Natural sampling, also known as zero-order hold sampling,
involves taking discrete interval samples of a continuous signal,
similar to uniform sampling. The difference, though, is in how the
samples are gathered.
Sampling Process: Each sample is taken in natural sampling by
retaining the value of the continuous signal constant for the duration of
the sampling period.

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 Natural Sampling

Natural sampling is similar to impulse sampling, except the impulse train is replaced by pulse
train of period T. i.e. you multiply input signal x(t) to pulse train as shown below

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 Natural Sampling

The exponential Fourier series representation of p(t) can be given as

Substitute Fn value in equation 2

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 Natural Sampling

To get the spectrum of sampled signal, consider the

Fourier transform on both sides.

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Natural Sampling

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 Natural Sampling

Reconstruction: The reconstruction of the original signal from

natural samples, it usually involves connecting the samples with
flat line segments. This method simplifies the reconstruction
process compared to ideal sampling.

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 Natural Sampling

Advantages
 Generation is easy
 Practical LPF can be used

Disadvantages
 Amplitude of sampled pulse is varying
 For large value of Tao crosstalk (External EMI interference)

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 Flat-Top Sampling
3. Flat-Top Sampling
Concept: Flat-top sampling is a type of natural sampling in which each
sample is obtained by maintaining the value of the continuous signal
constant for a set period of time, resulting in a flat-top waveform.
Sampling Process: Instead of retaining the value for the whole sample
interval, flat-top sampling holds it only for a portion of the interval while
allowing it to change at the beginning and end.

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 Flat-Top Sampling
 During transmission, noise is introduced at top of the transmission pulse
which can be easily removed if the pulse is in the form of flat top.
 Here, the top of the samples are flat i.e. they have constant amplitude.
Hence, it is called as flat top sampling or practical sampling.
 Flat top sampling makes use of sample and hold circuit.

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 Flat-Top Sampling

Theoretically, the sampled signal can be obtained by

convolution of rectangular pulse p(t) with ideally sampled signal
say yδ(t) as shown in the diagram:

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 Flat-Top Sampling

Reconstruction: The reconstruction of the original signal from flat-top

samples, we can use interpolation techniques. Flat-top sampling is used in
applications where it is desirable to minimize the effects of finite
bandwidth and aliasing.

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 Natural Sampling

Advantages
 Generation is easy
 Better SNR
 Practical filter can be used

Disadvantages
 Aperture effect

APERTURE EFFECT -- THE SAMPLED SIGNAL IN THE FLAT TOP

SAMPLING CONSISTS OF ATTENUATED HIGH FREQUENCY
COMPONENTS AND THIS EFFECT

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Flat-Top Sampling- Aperture effect

The distortion caused by the use of pulse-

amplitude modulation to transmit an analog
information-bearing signal

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 Flat-Top Sampling - Equalizer

• Decreasing the in-band loss of the

reconstruction filter as the frequency
increases
• The amplitude response of the
equalizer is

1 1 f
• Heq (f) =
 
H ( f ) T sin c( fT ) sin(fT )

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 Equalizer

• Decreasing the in-band loss of the

reconstruction filter as the frequency
increases
• The amplitude response of the
equalizer is

• Heq (f) = 1 1 f
 
H ( f ) T sin c( fT ) sin(fT )
Comparision of various Samplingtechnique

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Comparision of various Samplingtechnique

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 Advantages and Disadvantages of Sampling

Advantages of Sampling Disadvantages of Sampling

•It is an important step in converting •When sampling rate is not proper, it may

analog signal to digital signal which lead to the problem of aliasing, resulting

allows efficient digital storage and in distorted signals

signal processing •In the process of Analog to Digital

•By taking samples at some specific conversion, the next step after sampling

interval time, it helps compressing is quantization which may result in: loss

the original signal, which in turn of information

helps in efficient transmission •While conversion (Analog to Digital)

•Digital signals can be easily sampling may introduce errors due to

processed using various algorithms factors such as quantization noise,

temperature variations, etc.

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 Solved Examples on Sampling

Question 1: Given the signal, what will be the sampling frequency for
which the signal can be reconstructed

x(t)=cos(2π10t)

Solution:
Comparing the given signal with cos(2πft)

fmax​=10Hz

then according to the sampling theorem, the sampling frequency will be

2×fmax⩽2×10⩽20Hz.

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Question 2: Sample the given signal at 60Hz

x(t)=cos(2π50t)

Solution:
We know that x[n] = cos(2π fmax / fs n)
Here, x[n] = (2π50/60n)
Finally, we get the discrete time signal as:
x[n]=cos(2π5/6n)

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Pulse Code Modulation

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Basic elements of a PCM system

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The advantages offered by digital pulse
modulation
• Performance
• Digital pulse modulation permits the use of regenerative repeaters, when
placed along the transmission path at short enough distances, can
practically eliminate the degrading effects of channel noise and signal
distortion.
• Ruggedness
• A digital communication system can be designed to withstand the effects of
channel noise and signal distortion
• Reliability
• Can be made highly reliable by exploiting powerful error-control coding
techniques.
• Security
• Can be made highly secure by exploiting powerful encryption algorithms
• Efficiency
• Inherently more efficient than analog communication system in the tradeoff
between transmission bandwidth and signal-to-noise ratio
• System integration
• To integrate digitized analog signals with digital computer data

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A PCM Generator/Transmitter

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PCM Transmission Path

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PCM Receiver

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INFLUNCE OF NOISE ON THE
PCM SYSTEM

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 Advantages of Digital
Representation of Signals
1. Immunity to transmission noise and interference
2. It is possible to regenerate the coded signal along
the transmission path
3. Communication can be kept private and secured
by the use of encryption technique.
4. It possible to store the signal and process it
whenever required.

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 Disadvantages

1. Increased transmission bandwidth

2. Increased system complexity

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 Performance Evaluation of PCM

• Channel Noise

• Quantization Noise

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Probability of Error in PCM system

Pe Pe decreases very rapidly as the value of

10-2
Emax/No is increased

10-4
A very small increase in signal power

10-6 makes the system error free

10-8

10-10

10-12

10 15 20 SNR(dB)

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 Error Thrshold

• Emax/No --- above this Pe <<

• Channel Noise --- repeaters

PCM IS NOISE RESISTANT or RUGGED SYSTEM

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 SNR at Destination

• (S/N)D = 3 q2 Sx

• q=2v (PCM) so (S/N)D = 3 22v Sx

• q=Mv (M-Array) Mb b=BW/fm so (S/N)D= 3 Mb Sx

• Sx Signal Power at Destination

• (SNR)D PCM is greater than (SNR)D FM

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Performance comparison of PCM and analog modulation

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Quantizer

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 Quantizer
Step -1- The voltage range is divided into L equal levels
Step-2-2 The step between two interval is called step size(Delta)
= (Vmax-Vmin) /L
L=2^n n= bit depth
Step 3—Draw mid line at Delta/2, representing quantization level
Step 4—Assign binary code to each quantization level
Step 5—Calculate Quantization error
Qe = X(nTs)-Xq(nTs)
Qemax=Delta/2

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Quantizer

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Quantizer

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Quantizer

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Quantizer

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Quantizer

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 Quantizer
• Classification of Quantization Process

Quantization

Uniform quantization Nonuniform quantization

Midtread type Midrise type

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 Quantizer
• Classification of Quantization Process

Quantization

Uniform quantization Nonuniform quantization

Midtread type Midrise type

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 Working Principal of Quantizer
• Midtread type

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Midrise type

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Quantization error
• ε = xq(nTs) - x(nTs)

Δ/2
-2 Δ - Δ 0 Δ 2Δ 3Δ

- Δ/2

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 Midrise type
Quantization error ε = xq(nTs) - x(nTs)

• When x(nTs) =0 , xq(nTs) = +/- Δ/2

• If we assign Δ/2 then,

• ε = xq(nTs) - x(nTs)
• = Δ/2 - 0
• = Δ/2
• Max Quantization error ε = | Δ/2 |

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Uniform quantization with Incorrect
Quantization Characteristics

-2 Δ - Δ 0 Δ 2Δ 3Δ

Max Quantization error ε = | Δ |

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NECESSITY OF NON UNIFORM
QUANTATISATION

• It is natural to expect that the

amount of distortion introduced
depends on the quantizer, and so
one can try to minimize the
distortion by choosing a “good”
quantizer.

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• In Uniform Quantization the
quantizer has linear
characteristics.
• The step size also remains same
throughout range of quantizer.
• Therefore maximum quantization
error also remains same.
• εmax = l Δ / 2 l

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 Δ = 2Xmax / q

If x(t) is normalized, its maximum value is

Xmax = 1
Δ=2/q

εmax = l 1 / q l

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 NECESSITY OF NON UNIFORM
QUANTIZATION FOR SPEECH SIGNAL

• Speech and music signal have large crest factor

• Crest Factor = Peak value/ RMS value
= Xmax / [X2(t)]1/2
= 1 / [P]1/2 ……[P=X2(t)/R, R
=1]

To get large Crest Factor P << 1

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• SNR = 3 P 2v -----------(1)
•P=1
• So actual SNR is << -----------(1)
• At low signal levels (P<<) SNR ↓
• signal P ↓
• Quantization Noise ↑
• This can be improved by ----------

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• Step size should be varied according to signal level to
keep SNR at required level

This is nothing but

Non uniform quantization

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 Non uniform quantization

• If the quantizer Characteristic is

nonlinear and step size is not
constant instead if it is variable,
dependent on amplitude of input
signal then the quantization is
known as
NONUNIFORM QUANTIZATION

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 Nonuniform Quantization

Many signals such as speech have a nonuniform distribution.

• The amplitude is more likely to be close to zero than to be at higher levels.
Nonuniform quantizers have unequally spaced levels
• The spacing can be chosen to optimize the SNR for a particular type of signal.
Output sample
XQ 6

2 Example: Nonuniform 3 bit quantizer

-8 -6 -4 -2 2 4 6 8

-2
Input sample
X
-4

-6

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• For Uniform quantization
• Nq = Δ2 / 12 ------------- constant
• SNR α signal power

• Practical Difficulty – Prediction of Input

Non-uniform quantizers are difficult to make and expensive.

This is practically achieved through process called

COMPANDING
• Companding is non uniform quantization.

• It is used to improve Signal to Noise quantization ratio of weak

signals
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 Companding
• An alternative is to first pass the speech signal
through a nonlinearity before quantizing with a
uniform quantizer.
• The nonlinearity causes the signal amplitude to be
Compressed.
• The input to the quantizer will have a more uniform
distribution.
• At the receiver, the signal is Expanded by an
inverse to the nonlinearity.
• The process of compressing and expanding is called
Companding.

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• Companding = Compressing + Expanding

• Compressor – weak signals –amplified

strong signals –attenuated
• Expander – weak signals –attenuated
strong signals - amplified

Uniform Expander
Compressor
Quantizer

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Compressor and Expander
Characteristics
6 7
5 6

4 5
4
3
3
2
2
1
1
0 0
0 5 10 0 2 4 6

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 Vo Linear Characteristics

Expansion

Vi

Compression

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 A-law and  -law Companding
• These two are standard companding methods.
•  -Law is used in North America and Japan
• A-Law is used elsewhere to compress digital telephone signals

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• A Law --- Midrise Type
•  -law --- Midtread Type

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 -Law Companding
1

• Telephones in the U.S., Canada and

Japan use -law companding:
Output |x(t)|

ln(1   | x(t )|)

| y (t ) |
ln(1   )
• Where  = 255 and |x(t)| < 1

0 1
Input |x(t)|

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 Non Uniform quantizing
• Voice signals are more likely to have amplitudes near zero than at extreme peaks.
• For such signals with non-uniform amplitude distribution quantizing noise will be
higher for amplitude values near zero.
• A technique to increase amplitudes near zero is called Companding.

Effect of non linear quantizing can be

can be obtained by first passing the
analog signal through a compressor
and then through a uniform quantizer.

x x’ x’ y
Q(.)
C(.)
Compressor Uniform Quantizer

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 Example: -law Companding
1

0.5

x[n]=speech /song/ 0

-0.5

-1
0 1000 2000 3000 4000 5000 6000 7000 8000 9000 10000

0.5

y[n]=C(x[n]) 0

-0.5
Companded Signal -1
0 1000 2000 3000 4000 5000 6000 7000 8000 9000 10000

0.5
Close View of the Signal
Segment of x[n] 0

-0.5

-1
2200 2300 2400 2500 2600 2700 2800 2900 3000

0.5
Segment of y[n] 0

Companded Signal -0.5

-1
2200 2300 2400 2500 2600 2700 2800 2900 3000

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 SNR of Compander

SNRQ  1.8  6n for sinusoidal signal

SNRQ  10 log 10 c  2nlog 10 2    6n dB for other signal

 3m~ 2 (t )
 for non - compressed signal
 mp 2
c
 3
for compressed signal
 ln(1   )2

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 SNR of Compander
• The output SNR is a function of input signal level for uniform quantizing.
• But it is relatively insensitive for input level for a compander

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 V.90 56-Kbps PCM Computer modem
• The V.90 PC Modem transmits data at 56kb/s from a PC
via an analog signal on a dial-up telephone line.
• A μ law compander is used in quantization with a value
for μ of 255.
• The modem clock is synchronized to the 8-ksample/ sec
clock of the telephone company.
• 7 bits of the 8 bit PCM are used to get a data rate of
56kb/s ( Frequencies below 300Hz are omitted to get rid
of the power line noise in harmonics of 60Hz).
• SNR of the line should be at least 52dB to operate on
56kbps.
• If SNR is below 52dB the modem will fallback to lower
speeds ( 33.3 kbps, 28.8kbps or 24kbps).

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 DELTA
MODULATION

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What is Delta Modulation?
• It is a modulation technique in which
one bit is transmitted per sample.

• The value transmitted is either 1 or 0

dependinding on the difference
between between present and last
sampled value.

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Working Principle
• The current sampled value is compared with
the previous sampled value and if the
current sampled value is greater than
previous sampled value than 1 is transmitted
otherwise 0 is transmitted.
• The step size is fixed if the difference is
positive the approximated signal is increased
by one step size(Δ) and if the difference is
negative the approximated signal is reduced
by one step size(Δ) .

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Waveform of delta Modulation

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Transmitter Block

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Receiver Block

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ADVANTAGES

• The transmission bandwidth of delta

modulation is quite small because it
transmits only 1 bit per sample.
• The transmitter and receiver implementation
is very simple for delta modulation.

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DISADVANTAGES
Slope Overload Distortion
If the rate of rise of input signal is so high that the
staircase signal cannot approximate it ,then there is large
error between staircase approximated signal and original
signal x(t).This distortion is known as slope overload
distortion.
Condition for avoiding slope over distortion
(dx(t)/dt)max<=(s/Ts)

Remedy
To reduce this error the step size must be increased
when slope of signal x(t) is high.

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Contd
Granular or Idle Noise
This occurs when step size is too large as compared
to the small variations in the input signal. Thus for
small variations in input signal there is large variation
in the staircase signal due to large step size.

Remedy
This noise can be avoided by decreasing the step
size.

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Signal to Noise Ratio of Delta
Modulation

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Adaptive delta
modulation

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What is Adaptive Delta
Modulation
• The performance of a delta modulator can be
improved significantly by making the step size of
the modulator assume a time-varying form. In
particular, during a steep segment of the input
signal the step size is increased. Conversely, when
the input signal is varying
slowly, the step size is reduced. In this way, the size
is adapted to the level of the input
signal. The resulting method is called adaptive delta
modulation (ADM)

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Block Diagram Of Adaptive Delta
Modulation
• 1) Transmitter

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• Transmitter consists of logic for step size
control, one bit quantizer n delay blocks.
• The step size increases or decreases
according to a specified rule depending
upon one bit quantizer output.
• For Example if one bit quantizer output
is high (i.e. 1) then step size maybe
doubled for next sample. If the quantizer
output is low then the step size is
reduced by one bit .

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• Waveform

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• 2) Receiver

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• The receiver of adaptive delta
modulation consists of two portions.
• The first portion produces the step size
from each incoming bit. The previous
input and present input decides the step
size.
• It is then applied to accumulator which
builds up staircase form.
• Low Pass Filter smoothens out the
staircase waveform to produce original
signal.

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Advantages
• The Signal to Noise ratio becomes better than
ordinary delta modulation because of the reduction
in slope overload distortion.
• Because of variable step size, the dynamic range of
ADM is wider than simpler DM
• Utilization of Bandwidth is better than Delta
Modulation

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Department of Electronics & Telecommunication
Department of Electronics & Telecommunication
Department of Electronics & Telecommunication
Department of Electronics & Telecommunication
Nyquist - Shannon Sampling Theorem

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Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

Nyquist - Shannon Sampling Theorem

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication

 JSPM’s
RAJARSHII SHAHU COLLEGE OF ENGINEERING
(An Autonomous Institute Affiliated to Savitribai Phule Pune University, Pune )

Department of Electronics & Telecommunication