Unit 3PCS
Unit 3PCS
Sampling Theorem:
Sampling is defined as, “The process of measuring the instantaneous values of continuous-
time signal in a discrete form.”
Sample is a piece of data taken from the whole data which is continuous in the time domain.
When a source generates an analog signal and if that has to be digitized, having 1s and 0s i.e.,
High or Low, the signal has to be discretized in time. This discretization of analog signal is
called as Sampling.
The following figure indicates a continuous-time signal x ( t) and a sampled signal xs (t).
When x (t) is multiplied by a periodic impulse train, the sampled signal xs (t) is obtained.
Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be termed
as a sampling period Ts.
SamplingFrequency=1/Ts=fs
Where,
Sampling frequency is the reciprocal of the sampling period. This sampling frequency, can be
simply called as Sampling rate. The sampling rate denotes the number of samples taken per
second, or for a finite set of values.
For an analog signal to be reconstructed from the digitized signal, the sampling rate should be
highly considered. The rate of sampling should be such that the data in the message signal
should neither be lost nor it should get over-lapped. Hence, a rate was fixed for this, called as
Nyquist rate.
Nyquist Rate
Suppose that a signal is band-limited with no frequency components higher than W Hertz. That
means, W is the highest frequency. For such a signal, for effective reproduction of the original
signal, the sampling rate should be twice the highest frequency.
Which means,
fS=2W
Where,
A theorem called, Sampling Theorem, was stated on the theory of this Nyquist rate.
Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of
sufficient sample rate in terms of bandwidth for the class of functions that are band limited.
The sampling theorem states that, “a signal can be exactly reproduced if it is sampled at the
rate fs which is greater than twice the maximum frequency W.”
To understand this sampling theorem, let us consider a band-limited signal, i.e., a signal whose
value is non-zero between some –W and W Hertz.
If the signal x (t) is sampled above the Nyquist rate, the original signal can be recovered, and
if it is sampled below the Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the frequency
domain.
The above figure shows the Fourier transform of a signal xs(t). Here, the information is
reproduced without any loss. There is no mixing up and hence recovery is possible.
Let us see what happens if the sampling rate is equal to twice the highest frequency (2W)
That means,
fs=2W
Where,
We can observe from the above pattern that the over-lapping of information is done, which
leads to mixing up and loss of information. This unwanted phenomenon of over-lapping is
called as Aliasing.
Aliasing
• In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before
the sampler, to eliminate the high frequency components, which are unwanted.
• The signal which is sampled after filtering, is sampled at a rate slightly higher than the
Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate, also helps in the easier design
of the reconstruction filter at the receiver.
It is generally observed that, we seek the help of Fourier series and Fourier transforms in
analyzing the signals and also in proving theorems. It is because −
• The Fourier Transform is the extension of Fourier series for non-periodic signals.
• Fourier transform is a powerful mathematical tool which helps to view the signals in
different domains and helps to analyze the signals easily.
• Any signal can be decomposed in terms of sum of sines and cosines using this Fourier
transform.
Low pass and Band pass Signals
The signals which we design on our computers lie in a low-frequency band or aka Lowpass
signals, and the signal which we actually transmit over the air are known as Bandpass signals,
aka High-Frequency signals.
Why transmit on Bandpass and not on Lowpass? A straightforward question one can ask is
why there is a need to transmit the signals on Bandpass frequencies ( high frequencies) why
don’t we transmit in Lowpass?
Well, there are multiple reasons for that. One of the reasons is to avoid crosstalk and
interference. Let’s take a very simple example: Suppose you are sitting in a classroom, or any
public gathering and all of the people there start speaking at the same time now this will create
noise, and it will present a scenario of a fish market. Another reason is the availability of high
bandwidth. At high frequencies, there is a lot of bandwidth available, which means a high data
rate. Very important reasoning in this context is the suitable characteristics of the channel.
Bandpass signals support different types of communication channels like fiber optics, copper
cables, coaxial cable, twisted pair, wireless channel, etc.
Designing signals in Baseband only: Baseband is another term used for Lowpass signals.
Earlier I said that we design signals at low frequencies and then transmit over the high
frequencies using SDR. Why is that? Why don’t we design the signals directly at high
frequencies?
Answer: Well! The designing of electronic circuits at high frequencies is cumbersome and
expensive. One needs to design everything from all the way to
modulator/differentiator/integrator/encoder/decoder etc. at high frequencies, and this requires
a lot of effort, and thus, designing the circuits at high frequency is expensive. So, the easy
solution is to design everything at low frequencies and use a frequency upconverter or SDR
to shift the signal at high frequency.
This is one of the major reasons why we first study the difference of Bandpass and Lowpass
signals and how they can be represented in terms of one another. Some facts about Bandpass
and Lowpass signals:
• All Bandpass signals are real signals (i.e., whatever you transmit in the air is a real
signal)
• All Lowpass signals are complex in nature. They can be real, but it depends on the
way they are designed.
• Bandpass signals lies on high frequencies but lowpass signals are always designed
at fc=0
Pulse modulation::
PAM: In this scheme high frequency carrier (pulse) is varied in accordance with sampled value
of message signal.
PWM: In this width of carrier pulses are varied in accordance with sampled values of message
signal. Example: Speed control of DC Motors.
PPM: In this scheme position of high frequency carrier pulse is changed in accordance with
the sampled values of message signal.
2. Digital Pulse Modulation
In systems utilizing digital pulse modulation, the transmitted samples take on only discrete
values. Two important types of digital pulse modulation are:
1. Delta Modulation (DM)
2. Pulse Code Modulation (PCM)
There are two types of sampling techniques for transmitting messages using pulse amplitude
modulation, they are
FLAT TOP PAM: The amplitude of each pulse is directly proportional to instantaneous
modulating signal amplitude at the time of pulse occurrence and then keeps the amplitude
of the pulse for the rest of the half cycle.
In single polarity pulse amplitude modulation, there is fixed level of DC bias added to the
message signal or modulating signal, so the output of modulating signal is always positive. In
the double polarity pulse amplitude modulation, the output of modulating signal will have both
positive and negative ends.
Advantages of Pulse Amplitude Modulation (PAM):
• It is the base for all digital modulation techniques and it is simple process for both
modulation and demodulation technique.
• No complex circuitry is required for both transmission and reception. Transmitter and
receiver circuitry is simple and easy to construct.
• PAM can generate other pulse modulation signals and can carry the message or information
at same time.
• Bandwidth should be large for transmitting the pulse amplitude modulation signal. Due to
Nyquist criteria also high bandwidth is required.
• The frequency varies according to the modulating signal or message signal. Due to these
variations in the signal frequency, interferences will be there. So noise will be great. For
PAM, noise immunity is less when compared to other modulation techniques. It is almost
equal to amplitude modulation.
• Pulse amplitude signal varies, so power required for transmission will be more, peak power
is also, even at receiving more power is required to receive the pulse amplitude signal.
Definition: A multiplexing technique by which multiple data signals can be transmitted over a
common communication channel in different time slots is known as Time Division
Multiplexing (TDM).
It allows the division of the overall time domain into various fixed length time slots. A single
frame is said to be transmitted when it’s all signal components gets transmitted over the
channel.
Theory of TDM
As we know, multiplexing allows the transmission of several signals over a common channel.
However, one may need to differentiate between the various signal for proper data
transmission. So, in time division multiplexing, the complete signal gets transmitted by
occupying different time slots.
The name itself is indicating here that basically time division is performed in order to multiplex
multiple data signals.
Let us have a look at the figure below in order to have a better understanding of the TDM
process.
As we can see that source A, B and C wants to transmit data through a common medium. Thus,
the signal from the 3 sources, is divided into multiple frames each having their fixed time slot.
Here, 3 units from each source are taken into consideration, that jointly form the actual signal.
A frame is transmitted at a time that is composed of one unit of each source. As these units are
entirely different from each other thus the chances of unnecessary signal mixing can be
eliminated.
Here, we have taken the example of 3 different sources, but one can perform multiplexing of n
source signals. It is noteworthy here that units of a single source must be equivalent to the total
number of source signals to be transmitted.
Synchronous TDM is known as synchronous and is essential because, each time slot is pre-
assigned to a constant source. The time slots are sent irrespective of whether the sources have
a few records to share or not.
TDM devices can manage the source of various data rates. This is completed by authorising
fewer slots per cycle to the passive input devices than the rapid device.
Both multiplexing and demultiplexing operations for synchronous TDM, are demonstrated in
the figure given below.
Statistical TDM
One disadvantage of the TDM method is that some of the time slots in the frame are wasted. A
specific terminal has no information to send to a particular instant of time and will share an
unfilled time slot. It is also called asynchronous TDM or intelligent TDM.
In TDM, the data flow of each input stream is divided into units. One unit may be 1 bit, 1 byte,
or a block of few bytes. Each input unit is allotted an input time slot. One input unit corresponds
to one output unit and is allotted an output time slot. During transmission, one unit of each of
the input streams is allotted one-time slot, periodically, in a sequence, on a rotational basis.
This system is popularly called round-robin system.
Example
Consider a system having four input streams, A, B, C and D. Each of the data streams is divided
into units which are allocated time slots in the round – robin manner. Hence, the time slot 1 is
allotted to A, slot 2 is allotted to B, slot 3 is allotted to C, slot 4 is allotted to D, slot 5 is
allocated to A again, and this goes on till the data in all the streams are transmitted.
Definition: A modulation technique where the width of the pulses of the pulsed carrier wave
is changed according to the modulating signal is known as Pulse Width Modulation (PWM).
It is also known as Pulse duration modulation (PDM).
It is a type of Pulse Time Modulation (PTM) technique where the timing of the carrier pulse
is varied according to the modulating signal
In pulse duration modulation (PDM), the amplitude of the pulse is kept constant and only the
variation in width is noticed. As the information component is present in width of the pulses.
Thus, during signal transmission, the signal undergoes pulse width modulation. Due to constant
amplitude property, it gets less affected by noise. However, during transmission channel noise
introduces some variation in amplitude as it is additive in nature. But that is totally easy
removable at the receiver by making use of limiter circuit.
As the width of the pulses contains information. Thus the noise factor does not cause much
signal distortion. Hence the immunity to the noise of a PWM system is better than
the PAM system
The figure below shows the process of pulse width modulation. It is commonly known as an
indirect method of PWM generation.
The message signal and the carrier waveform is fed to a modulator which generates PAM
signal. This pulse amplitude modulated signal is fed to the non-inverting terminal of the
comparator.
A ramp signal generated by the sawtooth generator is fed to the inverting terminal of the
comparator.
These two signals are added and compared with the reference voltage of the comparator circuit.
The level of the comparator is so adjusted to have the intersection of the reference with the
slope of the waveform.
The PWM pulse begins with the leading edge of the ramp signal and the width of the pulse is
determined by the comparator circuit.
The width of the PWM signal is proportional to the omitted portion of the ramp signal by the
comparator level.
The figure below will help you to understand in a better way how PWM signal is generated by
the comparator:
Here, the first image i.e., (a) shows the waveform of the sinusoidal modulating signal and the
second one (b) shows the pulsed carrier. After modulation, a PAM signal is generated that is
shown in (c). This PAM signal, when added with ramp signal shown in (d), is compared with
the reference voltage of the comparator shown in figure (e). Lastly, figure (f) shows the PWM
signal.