Wallaga University

College of Engineering and Technology

School of Electrical and Computer Engineering

Course Title: Introduction to Communication Systems

 Chapter-Four
Baseband Pulse Signaling
Outline
Sampling theorem
Quantizing
Encoding
Digital modulation
ASK
FSK
PSK
QAM
Sampling Theorem
Converting a continuous-time signal into a discrete-time signal.
Process of measuring the discrete instantaneous values of a continuous-time
signal.
The sampling process includes the following steps:
The continuous signal is taken as an input.
Sampling is performed to convert this signal into a digital representation.
In addition to sampling, quantization of a signal is performed.
After the above step, encoding of the signal is done.
Types of Signals in Sampling

There are several types of signals in sampling that are discussed below.
Continuous Time Signals

Discrete Time Signals

Digital Signals
1. Continuous Time Signals
CTS are those signals that are continuous in both time and amplitude.
They are represented by functions that stay continuous over a range of
time and values of amplitude.
2. Discrete Time Signals
DTS are those that are continuous in amplitude but discrete in time meaning that
they are signals with values at some specific instants of time. It is to be taken
care of that these signals are discrete only in time whereas they can be either
continuous or discrete in amplitude.
3. Digital Signals
After sampling and quantization, the resulting signals are in digital
format are hence called Digital Signals. These signals have both their
time and amplitude in discrete format.
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:
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:
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:
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.
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.
Oversampling & Under sampling
1. Under sampling
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 under sampling can be when the high frequency
components are not useful and user needs to reduce the amount of data
being processed.
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.
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 under sampled.
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:
o transmitted,
o stored, and manipulated efficiently in the digital world.
o Noise reduction,
o error detection and correction,
o compression, signal processing, and interoperability are all enabled by
this conversion, which is crucial for modern communication systems.
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.
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.
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.
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.
1. 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.
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.
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.
Scope of Fourier Transform
It is well noticed that we seek the assistance of Fourier series and
Fourier transforms in analyzing signals and proving theorems. This is
because:

The Fourier Transform is used for a non-periodic signal which is helpful

in analysis of the signal.
The Fourier transform is helps to observe signals in several domains
and readily analyze them making it a strong mathematical tool.
Using this Fourier transform, any signal may be decomposed into the
sum of sines and cosines.
Advantages and Disadvantages of Sampling
Advantages of Sampling
It is an important step in converting analog signal to digital signal which allows
efficient digital storage and signal processing
By taking samples at some specific interval time, it helps
• Compressing the original signal, which in turn helps in efficient transmission
• Digital signals can be easily processed using various algorithms
Disadvantages of Sampling
• When sampling rate is not proper, it may
• Lead to the problem of aliasing, resulting in distorted signals
• In the process of Analog to Digital conversion, the next step after sampling is
quantization which may result in: loss of information
• While conversion (Analog to Digital) sampling may introduce errors due to factors
such as quantization noise, temperature variations, etc.
Digital Modulation
provides more information capacity
 High data security
 Quicker system availability with great quality communication
 Hence, digital modulation techniques have a greater demand, for their
capacity to convey larger amounts of data than analog modulation
techniques.
Types of digital modulation techniques
• Why Do We Need Digital-to-Analog Conversion?
The medium/channel is band pass, and/or
Multiple users need to share the medium.
ASK
• ASK – strength of carrier signal is varied to represent binary 1 or 0
• Both frequency & phase remain constant while amplitude changes
commonly, one of the amplitudes is zero.