PULSE AND DIGITAL COMMUNICATION

MODULATION: Modulation is the process of transmitting a message

signal (Baseband signal with low frequency) from transmitter to
receiver without changing its characteristics (like amplitude,
frequency, phase)

Sampling theorem
The sampling theorem specifies the minimum-sampling rate at
which a continuous-time signal needs to be uniformly sampled so
that the original signal can be completely recovered or
reconstructed by these samples alone. This is usually referred to as
Shannon's sampling theorem in the literature.
Sampling theorem:
If a continuous time signal contains no frequency components
higher than W Hz, then it can be completely determined by uniform
samples taken at a rate fs samples per second where

fs≥2W
or, in term of the sampling period

T≤12W
A signal with no frequency component above a certain maximum
frequency is known as a band limited signal. Figure 2.4 shows two
typical band limited signal spectra: one low-pass and one band-
pass.
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Figure 2.4. Two bandlimited spectra

The minimum sampling rate allowed by the sampling theorem (fs =

2W) is called the Nyquist rate.
It is interesting to note that even though this theorem is usually
called Shannon's sampling theorem, it was originated by both E.T.
and J.M. Whittaker and Ferrar, all British mathematicians. In
Russian literature, this theorem was introduced to communications
theory by Kotel'nikov and took its name from him. C.E. Shannon
used it to study what is now known as information theory in the
1940s. Therefore in mathematics and engineering literature
sometimes it is also called WKS sampling theorem after Whittaker,
Kotel'nikov and Shannon.
Pulse Amplitude Modulation
By varying the amplitude of the pulses (the carrier signal) in proportion to
the instantaneous values of the analog signal (the message signal).

Pulse Amplitude
Modulation (PAM) Signals
The above figure illustrates the time-domain representation of the PAM
technique which mentions the analog message and PAM modulated signal
as an output.

Pulse amplitude modulation is used in the popular Ethernet communication

standard. The PAM modulator and demodulator circuits simple compared
to other kinds of modulation and demodulation techniques.

There are two categories of PAM techniques, one is the pulses have the
same polarity and the other in which the pulses can have both positive and
negative polarities according to the amplitude of the modulating signal.
Pulse Width Modulation
The Pulse width Modulation– By varying the width of the pulses (the carrier
signal) in proportion to the instantaneous values of the analog signal (the
message signal).

The width of the pulse varies, but the amplitude of the pulse remains
constant. Amplitude limiters are used to make the amplitude constant.
These circuits clip-off the amplitude, to a preferred level and hence the
noise is limited.

There are three types of PWM. They are

 The leading edge of the pulse being constant, the trailing edge varies
according to the message signal.
 The trailing edge of the pulse being constant, the leading edge varies
according to the message signal.
 The center of the pulse being constant, the leading edge and the trailing
edge varies according to the message signal.
Pulse Position Modulation

By varying the position of the pulses (the carrier signal) in proportion to the
instantaneous values of the analog signal (the message signal).

Pulse position modulation is done in accordance with the pulse width

modulated signal. Each trailing of the pulse width modulated signal
becomes the starting point for pulses in the PPM signal.

Hence, the position of these pulses is proportional to the width of the PWM
pulses. But the main disadvantage of the PPM modulation technique is,
The synchronization between transmitter and receiver must be needed.

Difference
Between PAM, PWM, and PPM

In all the above cases, we detect the message of the pulse modulated
signal and reconstruct the original analog signal.
Amplitude Shift Keying
The amplitude of the resultant output depends upon the input data whether it should
be a zero level or a variation of positive and negative, depending upon the carrier
frequency.
Amplitude Shift Keying (ASK) is a type of Amplitude Modulation which represents
the binary data in the form of variations in the amplitude of a signal.
Following is the diagram for ASK modulated waveform along with its input.

Any modulated signal has a high frequency carrier. The binary signal when ASK is
modulated, gives a zero value for LOW input and gives the carrier output for HIGH
input.

Frequency Shift Keying

The frequency of the output signal will be either high or low, depending upon the
input data applied.
Frequency Shift Keying (FSK) is the digital modulation technique in which the
frequency of the carrier signal varies according to the discrete digital changes. FSK
is a scheme of frequency modulation.
Following is the diagram for FSK modulated waveform along with its input.
The output of a FSK modulated wave is high in frequency for a binary HIGH input
and is low in frequency for a binary LOW input. The binary 1s and 0s are
called Mark and Space frequencies.

Phase Shift Keying

The phase of the output signal gets shifted depending upon the input. These are
mainly of two types, namely BPSK and QPSK, according to the number of phase
shifts. The other one is DPSK which changes the phase according to the previous
value.
Phase Shift Keying (PSK) is the digital modulation technique in which the phase of
the carrier signal is changed by varying the sine and cosine inputs at a particular
time. PSK technique is widely used for wireless LANs, bio-metric, contactless
operations, along with RFID and Bluetooth communications.
PSK is of two types, depending upon the phases the signal gets shifted. They are −

Binary Phase Shift Keying (BPSK)

This is also called as 2-phase PSK (or) Phase Reversal Keying. In this technique,
the sine wave carrier takes two phase reversals such as 0° and 180°.
BPSK is basically a DSB-SC (Double Sideband Suppressed Carrier) modulation
scheme, for message being the digital information.
Following is the image of BPSK Modulated output wave along with its input.
Quadrature Phase Shift Keying (QPSK)
This is the phase shift keying technique, in which the sine wave carrier takes four
phase reversals such as 0°, 90°, 180°, and 270°.
If this kind of techniques are further extended, PSK can be done by eight or sixteen
values also, depending upon the requirement. The following figure represents the
QPSK waveform for two bits input, which shows the modulated result for different
instances of binary inputs.

QPSK is a variation of BPSK, and it is also a DSB-SC (Double Sideband

Suppressed Carrier) modulation scheme, which send two bits of digital information
at a time, called as bigits.
Instead of the conversion of digital bits into a series of digital stream, it converts
them into bit-pairs. This decreases the data bit rate to half, which allows space for
the other users.

Differential Phase Shift Keying (DPSK)

In DPSK (Differential Phase Shift Keying) the phase of the modulated signal is
shifted relative to the previous signal element. No reference signal is considered
here. The signal phase follows the high or low state of the previous element. This
DPSK technique doesn’t need a reference oscillator.
The following figure represents the model waveform of DPSK.

It is seen from the above figure that, if the data bit is LOW i.e., 0, then the phase of
the signal is not reversed, but is continued as it was. If the data is HIGH i.e., 1, then
the phase of the signal is reversed, as with NRZI, invert on 1 (a form of differential
encoding).
If we observe the above waveform, we can say that the HIGH state represents
an M in the modulating signal and the LOW state represents a W in the modulating
signal.

Application of Digital Transmission

 Digital transmission is employed in order to achieve high reliability and
because the cost of digital switching systems is much lower than the cost of
analog systems.
 With fiber-optic cabling or wireless communication, variation in intensity or
some other physical quantity is used.

 Digital signals use discrete values for the transmission of binary

information over a communication medium such as a network cable or a
telecommunications link.
Advantages of digital transmission:

1. Immunity to transmission noise and interference.

2. Regenerative of the coded signal along the transmission path is possible.
3. Communication can be kept “private” and “secured” through the use of
encryption.
4. It is possible to use a uniform format for different kinds of baseband signals.
5. It is possible to store the signal and process it further.
6. Digital signals are better suited for processing and multiplexing
7. Digital transmission systems are more immune to noise
8. Measurement and evaluation of digital signals is simpler.
9. It is possible to evaluate error performance of digital systems.

Disadvantages:

1. The required bandwidth is increased due to digital technology.

2. System Complexity is increased
3. In order to convert the analog signal to digital prior to transmission and then
from digital to analog receiver, we need to use the additional encoder and
decoder circuit.
4. Synchronization is necessary for digital systems (between transmission and
receiver clocks)
5. Digital transmission systems are not compatible to older analog transmission
systems.

Characteristics of Data Transmission Circuits:

Bandwidth Requirements – Characteristics of Data Transmission Circuits is the most
instances consists of pulse-type energy. The data stream is similar to a square-wave
signal with rapid transitions from one voltage level to another, with the repetition
rate depending on the binary representation of the data word. For instance, if an 8-
bit word has the value 01010101, the resulting voltage graph would appear as a
series of four square waves with each negative half-cycle equal to each positive half-
cycle. If, however, the data word has the form 00001111, the voltage graph would
appear as a single square wave with negative and positive half-cycles equal but
longer than the first example. Figure 14-16 shows the voltage graphs for these and
other binary words. It can be seen that data circuits must provide a bandwidth for the
data transmissions they carry. This will be governed by the pulse rate variations just
explained, and by the fact, indicated in earlier, that even a single square wave
occupies a frequency range because of the harmonics present.
Since many data transmissions utilize telephone channels, the bandwidth of the
telephone is an appropriate consideration. The internationally accepted standard
telephone channel occupies the frequency range of 300 to 3400 Hz, this referred to
within the industry as a 4-kHz channel. In certain difficult or expensive applications,
such as HF radio or some submarine cables, 3-kHz circuits, in which the frequency
range is 300 to 2800 Hz, are used. Neither channel will encompass all the audible
spectrum, but each will cover the range into which speech falls and convey enough
of the components of speech to ensure intelligibility and voice recognition. The
signals which fall outside the, channel bandwidth are attenuated by filters so that
they will not interfere with other signals.

When data is sent over telephone channels, the speed must be limited to ensure that
the bandwidth required by the Characteristics of Data Transmission Circuits will not
exceed the telephone channel bandwidth. The faster the data is transmitted, the
greater the bandwidth will need to be to accommodate it.

Data Transmission Speeds – The rate of data transfer depends on several aspects of
the transmission channel, of which signaling speed is very important. Transmission
engineers often refer to the transmission speed of a communications channel as the
channel’s baud rate. The baud is an important unit of signaling speed. In a system in
which all pulses have equal duration, the speed in bauds is equal to the maximum
rate at which signal pulses are transmitted. This should be recognized as different
from information bit rate. In a system which uses only one information bit per
signaling pulse, i.e., a binary system, the baud rate and the bit rate happen to be the
same. In systems which encode the data in such a way that more than one
information bit can be placed on each signaling pulse, the information bit rate will
exceed the baud rate.

To relate baud rate to bandwidth, the observations of the twentieth-century electrical

engineer Nyquist are used. Nyquist determined that one cycle of a Characteristics of
Data Transmission Circuits can contain a maximum of two bands. The result is that
the maximum signaling speed in bauds is equal to twice the bandwidth of the
channel. This is theoretical and could be achieved only in an ideal channel which had
no noise or distortion.

As indicated above, the baud is a unit of signaling speed, but information transfer
can occur at a rate equal to or different from the baud rate. Multilevel and encoded
data elements can be used to provide information transfer rates at speeds greater
than the baud rate. In the Bell system 201A and 201B data sets, for example, data
streams are converted to 2-bit pairs. Each 2-bit pair can have only one of four values,
00, 01, 10 or 11. Each of the 2-bit pairs is converted to a phase value in the data set,
00 being represented by 90 degrees, 01 by 180 degrees, 10 by 270 degrees, and 11
by 0 degrees. Each of the 2-bit elements is called a dibit. This is, therefore, a four-
level code. Dibit-encoded data can be transmitted by using half the number of bands
required for the nonencoded data.

Noise – The Shannon-Hartley law is related to random noise, but impulse noise can
also be harmful to signals. The sampling theorem shows that all values of a signal
can be determined by sampling the signal at a rate equal to at least twice the
bandwidth. Noise affects this sampling process because the noise pulse will be
interpreted as a data bit ( see Figure 14-17), if the noise impulse occurs at the time a
sample is taken, and has an amplitude equal to or exceeding the minimum level
recognized by the system as a mark. The potential for impulse noise to become a
source of errors increases with the number of levels of each code element. To
achieve the 30,880-bps rate mentioned in the above example, it may be shown that
five levels would be required for each code element. A noise-free channel would be
necessary to preclude noise-induced data errors, but noise-free channels do not exist
in practice. It is noise, among other impairments, which tends to limit the actual 4-
kHz channel data speeds to 10,800 bps or less.

The effect of noise on the data channel can be reduced by increasing the signal-to-
noise ratio. For an ideal 3-kHz channel, the Nyquist rate (twice the bandwidth, as
discussed) would be 6000 bps. A binary system using this channel would require a
minimum signal-to-noise ratio of 3:1, or 4.8 dB. This is calculated by using Equation
(14-1), as follow’s:

where
S/N = Signal-to-noise ratio

NR = Nyquist rate

δf = Channel bandwidth

For the ideal 3-kHz channel:

To obtain the decibel value:

It can be shown that a system using a three-level code must have a signal-to-noise
ratio of 8.5 dB, or 3.7 dB greater, for equal performance in the same channel. A four-
level code requires a signal-to-noise performance of 11.7 dB. Improvement in the
signal-to-noise ratio makes use of multilevel encoding feasible.

Crosstalk – Any transmission system which conveys more than one signal
simultaneously can experience crosstalk, which is interference due to the reception of
portions of a signal from one channel in another channel. This is common in
multiplexed systems in which inadequate procedures are employed to ensure that
over-modulation of the various carriers of the multiplexed groups is prevented. In
modem transmission systems which convey many channels of voice and data
simultaneously, the systems will become “loaded,” or heavily utilized, so that the
control of levels of the individual channels and the group levels becomes very
important in order to preclude crosstalk. Data transmission engineers have
developed specific level-setting parameters to ensure that as the circuit loading
increases, crosstalk will not become a problem.

Crosstalk interference can also occur through electromagnetic interaction between

adjacent wires. If the wires of two signal-carrying circuits run parallel with each other,
it is possible for the signal from one circuit to be induced by electromagnetic
radiation into the second circuit. This phenomenon becomes more pronounced when
the length of parallel circuits is extensive. This type of crosstalk is reduced by using
twisted pair cables and balanced circuits along with shielding.
In a balanced circuit, a transformer is placed at each end of the circuit. The
transformers are carefully constructed to provide a center tap which is at the exact
electrical center of the winding which connects to the transmission circuit. The center
taps at each end are grounded. As shown in Figure 14-18, if twisted pair cables are
used for the transmission circuit, noise or signals from other circuits will be induced
into both wires at equal levels. When the crosstalk or noise reaches the transformer,
it enters as out-of-phase signals from the two wires and cancels out in the
transformer windings. The circuit signal, however, enters the transformer in phase.
Each side of the transformer forms a circuit with ground and the signal transfers
through the transformer intact. The crosstalk and noise are reduced, but the signal is
unaffected.

Another way to reduce crosstalk is to use shielded cables. If the twisted pairs are
placed inside a braided or metal foil shield, the induction between pairs take place as
easily. The shields are grounded to drain off the induced signals and noise.

Echo suppressors – As discussed already, echo suppressors or echo cancellers are

used on long-distance circuits, in an effort to overcome echoes caused by circuit
imbalances. This is of significance to Characteristics of Data Transmission Circuits
because a lot of it occurs over the public switched telephone network, nationally and
internationally.

Although the use of echo suppressors improves voice communications, it is

incompatible with Characteristics of Data Transmission Circuits. Because a lot of data
transmissions are bothway, or quickly alternating from one direction to the other,
they require the capability of bidirectional transmission at standard levels, or at least
rapid response and interrupt capability. For this type of operation to be
accomplished, it is necessary to disable the echo suppressor. In fact, so-called “tone-
disablable” echo suppressors have been designed to accommodate the needs of
data users. If a 2025-Hz tone is applied to the line for approximately 300 ms prior to
the start of transmission, such an echo suppressor will be disabled and bidirectional
communication can proceed. If a gap in the transmission greater than 100 ms occurs,
the echo suppressor will be reactivated.
Distortion – Communication channels tend to react to signals of different speeds
within their bandpass in different ways. Specifically, signals of different frequencies
can be passed by the channel with different values of amplitude attenuation and at
different propagation speeds. The result is distortion.

Of great importance to systems using phase modulation is phase delay (or envelope
delay) distortion. Phase delay distortion occurs in a channel when signals of one
frequency are passed through the circuit at a different speed than other signals. The
resulting distortion can take the form of intersymbol interference. Since characters
which have lower-frequency components pass at a different speed than data
characters with high-frequency components, it is possible in higher-speed circuits for
portions of one character to enter or remain in the time slot allocated to other
characters.

Equalizers – Phase delay distortion can be reduced to acceptable levels by using

equalization on the channel. As shown in Figure 14-19, it is possible to plot the delay
characteristics of the channel and insert an equalizer which can be adjusted to
compensate for the delay abnormalities. The result is a channel relatively free of
phase delay.

Equalizers can be obtained which are automatic in nature. These equalizers precede
data transmission with a short “training period” during which test pulses are used to
determine the delay characteristics of the channel. The equalizer automatically varies
its delay characteristics while sampling the return signal to determine when the
channel delay plus equalizer delay reach proper tolerances. At that time, data
transmission commences. The Characteristics of Data Transmission Circuits is
thereafter sampled during transmission to ensure that equalization settings are
appropriate, with modifications made as required. This type of equalization is
called Adaptive Equalization.