Bahir Dar University

Institute of Technology
Faculty of Electrical and Computer
Engineering

INTRODUCTION TO
COMMUNICATION SYSTEM

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 Chapter Three
Pulse and Digital Modulation
Techniques

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 Lecture outlines

Sampling
pulse Analog Modulation Systems:
▪ PAM
▪ PPM
▪ PDM /PWM
Introduction to Digital communication systems
Pulse code modulation techniques
Line Coding
Introduction to digital modulation techniques
▪ ASK
▪ PSK
▪ FSK

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 Sampling and sampling
• Theorem
Sampling :The process of measuring the instantaneous values
of continuous-time signal in a discrete form.
• Sampling is common in all pulse modulation techniques.
• To discretize the signals, the gap between the samples should
be fixed. That gap can be termed as a sampling period Ts.
• 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

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 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,

This rate of sampling is called as Nyquist rate.

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 Sampling Theorem
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.
consider a band-limited signal, i.e., a signal whose value is
non-zero between some –W and W Hertz.

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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.

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Introduction to communication
▪

if the sampling rate is equal to twice the highest frequency(2W).

The result will be as shown in the above figure.

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 If

The resultant pattern will look like

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.

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 Sampling methods

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 Practical Sampling Rates
■ Speech
- Telephone quality speech has a bandwidth of 4
kHz (actually 300 to 3300Hz)
- Most digital telephone systems are sampled at
8000 samples/sec
■ Audio:
- The highest frequency the human ear can hear is
approximately 15kHz
- CD quality audio are sampled at rate of 44,000
samples/sec
■ Video
- The human eye requires samples at a rate of at
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least 20 frames/sec to achieve smooth motion
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 .
1. Pulse Amplitude Modulation Systems (PAM)

A modulation system in which amplitude of discrete carrier signal

changes in accordance with the instantaneous amplitude of
modulating signal(message signal) keeping width and position of
carrier constant is called as PAM.

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Advantage of PAM
▪ Generation and detection is easy.
Disadvantage
▪ Added noise cannot be removed easily as it has impact on
amplitude which carries information.
▪Transmission bandwidth is too large.

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 2. Pulse Width Modulation
• The modulation system in which width of the discrete carrier signal
changes in accordance with the instantaneous amplitude of
modulating signal(message signal) keeping amplitude and position
of carrier constant is called as PWM.

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 Pulse Width Modulation

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 • Advantage
1. Very good noise immunity.
2. Its possible to separate out signal from noise.
▪ Disadvantage
✔ Bandwidth requirement is large as compared to PAM.

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 3. Pulse Position Modulation
The modulation system in which position of the
discrete carrier signal changes in accordance with the
instantaneous amplitude of modulating signal(message
signal) keeping amplitude and Width of carrier constant
is called as PPM.

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 Difference Between PAM, PWM, and PPM
The below table gives the detailed difference between PWM, PAM, and PPM

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 Introduction to Digital communication systems
The Necessity of Digitization
The conventional methods of communication used analog
signals for long distance communications, which suffer from
many losses such as distortion, interference, and other losses
including security breach.
In order to overcome these problems, the signals are digitized
using different techniques. The digitized signals allow the
communication to be more clear and accurate without losses.
The digital signals consist of 1s and 0s which indicate High and
Low values respectively.
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 Advantages of Digital Communication
As the signals are digitized, there are many advantages of digital
communication over analog communication, such as −
The effect of distortion, noise, and interference is much less in
digital signals as they are less affected.
Digital circuits are more reliable.
Digital circuits are easy to design and cheaper than analog circuits.
The hardware implementation in digital circuits, is more flexible
than analog.

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Signal processing functions such as encryption and compression are
employed in digital circuits to maintain the secrecy of the
information.
The probability of error occurrence is reduced by employing error
detecting and error correcting codes.
Spread spectrum technique is used to avoid signal jamming.
Combining digital signals using Time Division Multiplexing (TDM) is
easier than combining analog signals using Frequency Division
Multiplexing (FDM).
The configuring process of digital signals is easier than analog signals.
Digital signals can be saved and retrieved more conveniently than
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analog signals.
 Elements of Digital Communication
The elements which form a digital communication system is
represented by the following block diagram for the ease of
understanding.

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 Source
▪ The source can be an analog signal. Example: A Sound signal
Input Transducer
▪ This is a transducer which takes a physical input and converts it
to an electrical signal (Example: microphone). This block also
consists of an analog to digital converter where a digital signal is
needed for further processes. A digital signal is generally
represented by a binary sequence.
Source Encoder
▪ The source encoder compresses the data into minimum number
of bits. This process helps in effective utilization of the
bandwidth. It removes the redundant bits (unnecessary excess
bits, i.e., zeroes).

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 Channel Encoder
▪ The channel encoder, does the coding for error correction. During
the transmission of the signal, due to the noise in the channel, the
signal may get altered and hence to avoid this, the channel encoder
adds some redundant bits to the transmitted data. These are the
error correcting bits.
Digital Modulator
▪ The signal to be transmitted is modulated here by a carrier. The
signal is also converted to analog from the digital sequence, in
order to make it travel through the channel or medium.
Channel
▪ The channel or a medium, allows the analog signal to transmit from
the transmitter end to the receiver end.

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 Digital Demodulator
▪ This is the first step at the receiver end. The received signal is
demodulated as well as converted again from analog to digital.
The signal gets reconstructed here.
Channel Decoder
▪ The channel decoder, after detecting the sequence, does some
error corrections. The distortions which might occur during the
transmission, are corrected by adding some redundant bits. This
addition of bits helps in the complete recovery of the original
signal.
Source Decoder
▪ The resultant signal is once again digitized by sampling and
quantizing so that the pure digital output is obtained without the
loss of information. The source decoder recreates the source
output.
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 Output Transducer
▪ This is the last block which converts the signal into the original
physical form, which was at the input of the transmitter. It converts
the electrical signal into physical output (Example: loud speaker).
Output Signal
▪ This is the output which is produced after the whole process.
Example − The sound signal received.

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Pulse code modulation techniques
DEFINITION: Pulse code modulation (PCM) is essentially
analog-to-digital conversion of a special type where the
information contained in the instantaneous samples of an analog
signal is represented by digital words in a serial bit stream.

The advantages of PCM are:

• Relatively inexpensive digital circuitry may be used
extensively.

• PCM signals derived from all types of analog sources may be

merged with data signals and transmitted over a common
high-speed digital communication system.
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•In long-distance digital telephone systems requiring repeaters, a
clean PCM waveform can be regenerated at the output of each
repeater, where the input consists of a noisy PCM waveform.
•The noise performance of a digital system can be superior to that
of an analog system.
•The probability of error for the system output can be reduced
even further by the use of appropriate coding techniques.

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 Basic Elements of PCM

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 The PCM signal is generated by carrying out three basic
operations:
1. Sampling
2. Quantizing
3. Encoding
1. Sampling operation generates a flat-top PAM signal.
2. Quantizing operation approximates the analog values by using a
finite number of levels. This operation is considered in 3 steps

a) Uniform Quantizer
b) Quantization Error
c) Quantized PAM signal output
3.Encoding PCM signal is obtained from the quantized PAM
signal by encoding each quantized sample value into a digital word.

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 V
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V
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 Quantization
• Sampling results in a series of pulses of varying amplitude values
ranging between two limits: a min and a max.

• The amplitude values are infinite between the two limits.

• We need to map the infinite amplitude values onto a finite set of

known values.

• This is achieved by dividing the distance between min and max into
L zones, each of height Δ.

Δ = (max - min)/L

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 Quantization Levels
• The midpoint of each zone is assigned a value from 0 to
L-1 (resulting in L values)
• Each sample falling in a zone is then approximated to
the value of the midpoint.

Two types of quantization: (a) midtread and (b) midrise

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 Quantization Zones

• Assume we have a voltage signal with amplitutes Vmin=-20V

and Vmax=+20V.
• We want to use L=8 quantization levels.
• Zone width Δ = (20 - -20)/8 = 5
• The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to 0, 0 to
+5, +5 to +10, +10 to +15, +15 to +20
• The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5, 17.5

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 Assigning Codes to Zones
• Each zone is then assigned a binary code.
• The number of bits required to encode the zones, or the number
of bits per sample as it is commonly referred to, is obtained as
follows: nb = log2 L
• Given our example, nb = 3
• The 8 zone (or level) codes are therefore: 000, 001, 010, 011,
100, 101, 110, and 111
• Assigning codes to zones:
– 000 will refer to zone -20 to -15
– 001 to zone -15 to -10, etc
• The output of the quantizer is one of M possible signal levels.
– If we want to use a binary transmission system, then we
need to map each quantized sample into an n bit binary
word.
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Encoding is the process of representing each quantized sample by
an ν bit code word.
The mapping is one-to-one so there is no distortion introduced by
encoding.
Some mappings are better than others.
A Gray code gives the best end-to-end performance.
The weakness of Gray codes is poor performance when the sign bit
(MSB) is received in error

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 Quantization and Encoding of a sampled signal

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 Quantization Error

• When a signal is quantized, we introduce an error - the coded

signal is an approximation of the actual amplitude value.
• The difference between actual and coded value (midpoint) is
referred to as the quantization error.
• The more zones, the smaller Δ which results in smaller errors.
• BUT, the more zones the more bits required to encode the
samples -> higher bit rate

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Quantization Noise in PCM System
• Errors are introduced in the signal because of the quantization
process. This error is called "quantization error". We define the
quantization error as:

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 Signal to Quantization Noise ratio in PCM

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 Bandwidth of PCM Signals
• The spectrum of the PCM signal is not directly related to the

spectrum of the input signal.

• The bandwidth of (serial) binary PCM waveforms depends on the

bit rate R and the waveform pulse shape used to represent the data.

• The Bit Rate R is

R=nfs
Where n is the number of bits in the PCM word (M=2n) and fs is
the sampling rate.

• For no aliasing case (fs≥ 2B), the Minimum Bandwidth of PCM

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B is:
 The Minimum Bandwidth of nfs//2 is obtained only when

sin(x)/x pulse is used to generate the PCM waveform.

For PCM waveform generated by rectangular pulses,

the First-null Bandwidth is:

Bpcm = R = nfs
Example
We want to digitize the human voice. What is the bit rate, assuming 8
bits per sample?

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 Delta Modulation
⚫ In Delta Modulation, only one bit is transmitted per
sample
⚫ That bit is a one if the current sample is more positive
than the previous sample, and a zero if it is more
negative
⚫ Since so little information is transmitted, delta
modulation requires higher sampling rates than PCM
for equal quality of reproduction

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⚫ This scheme sends only the difference between pulses, if the
pulse at time tn+1 is higher in amplitude value than the pulse
at time tn, then a single bit, say a “1”, is used to indicate the
positive value.
⚫ If the pulse is lower in value, resulting in a negative value, a
“0” is used.
⚫ This scheme works well for small changes in signal values
between samples.
⚫ If changes in amplitude are large, this will result in large
errors.
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 Distortions in DM system

1. If the slope of analog signal is much higher

than that of approximated digital signal over
long duration, than this difference is called
Slope overload distortion.

2. The difference between quantized signal and

original signal is called as Granular noise. It
is similar to quantization noise.

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 Line Coding
❖ Binary data can be transmitted using a number of different types of
pulses. The choice of a particular pair of pulses to represent the
symbols 1 and 0 is called Line Coding and the choice is generally
made on the grounds of one or more of the following
considerations:
Transmission Bandwidth: as small as possible
Power Efficiency: As small as possible for given BW and
probability of error
Error Detection and Correction capability.
Favorable power spectral density: dc=0
Adequate timing content: Extract timing from pulses
Transparency: Prevent long strings of 0s or 1s

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 Types of Line Coding

Uni polar Signalling

Polar Signalling
BiPolar Signalling

Manchester Signalling

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 Unipolar signalling
❖ Unipolar signalling (also called on-off keying, OOK) is the type
of line coding in which one binary symbol representing a 0 is
represented by the absence of a pulse and the other binary
symbol denoting a 1 is represented by the presence of a pulse .
❖ There are two common variations of unipolar signalling:
Non-Return to Zero (NRZ) and
Return to Zero (RZ).
UNIPOLAR NON-RETURN TO ZERO (NRZ):
❖ In unipolar NRZ the duration of the MARK pulse (Ƭ ) is equal to the duration
(To) of the symbol slot.

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 Advantage

Simplicity in implementation
Does not require a lot of bandwidth for transmission

Disadvantage

Presence of Dc Level
Does not have any error correction capability
Does not posses any clocking component for easy of
synchronization
It is not Transparent

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 Unipolar Return to Zero (RZ):

❖ In unipolar RZ the duration of the MARK pulse (Ƭ ) is

less than the duration (To) of the symbol slot. Typically
RZ pulses fill only the first half of the time slot, returning
to zero for the second half.

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Advantage

Simplicity in implementation

Disadvantage

Presence of Dc Level
Does not have any error correction capability
Occupies twice as much bandwidth as unipolar NRZ
It is not Transparent

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 Polar Signalling
❖ In polar signalling a binary 1 is represented by a pulse g1(t) and a
binary 0 by the opposite (or antipodal) pulse g0(t) = -g1(t). Polar
signalling also has NRZ and RZ forms.

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Advantage

Simplicity in implementation
No Dc Components

Disadvantage

Does not have any error correction capability

Does not posses any clocking component for easy of
synchronization
It is not Transparent

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 Positive voltage indicates a ‘1’, negative voltage indicates a ‘0’.
The voltage goes down to zero in the middle of every tick.

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Advantage

Simplicity in implementation
No Dc Components

Disadvantage

Does not have any error correction capability

Does not posses any clocking component for easy of
synchronization
It is not Transparent
Occupies twice as much bandwidth as polar NZR

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 Introduction to digital modulation techniques
Better performance and more cost effective than analog
modulation methods (AM, FM, etc.)

Used in modern cellular systems

Advancements in VLSI, DSP, etc. have made digital solutions
practical and affordable

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 Performance advantages
1) Resistant to noise, fading, & interference
2) Can combine multiple information types (voice,
data, & video) in a single transmission channel
3) Improved security (e.g., encryption) → deters phone
cloning + eavesdropping
4) Error coding is used to detect/correct transmission
errors
5) Can implement mod/dem functions using DSP
software (instead of hardware circuits).

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 Choice of digital modulation scheme
• Many types of digital modulation methods → subtle
differences
Performance factors to consider
▪ low Bit Error Rate (BER) at low S/N
▪ resistance to interference (ACI & CCI) & multipath fading
▪ occupying a minimum amount of BW
▪ easy and cheap to implement in mobile unit
▪ efficient use of battery power in mobile unit

No existing modulation scheme simultaneously

satisfies all of these requirements well.
Each one is better in some areas with tradeoffs of
being worse in others.
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• Power Efficiency : ability of a modulation
technique to preserve the quality of digital
messages at low power levels (low SNR)
Specified as Eb / No @ some BER (e.g. 10-5) where Eb
: energy/bit and No : noise power/bit
Tradeoff between fidelity and signal power →
BER ↑ as Eb / No ↓

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 digital modulation techniques

1. Amplitude shift keying(ASK)

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In the above Equation, the modulating signal [vm(t)] is a
normalized binary waveform, where + 1 V = logic 1 and -1 V = logic
0. Therefore, for a logic 1 in put,vm (t) = + 1 V,

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 2. Frequency Shift Keying

Frequency Shift Keying (FSK) is the digital modulation technique

in which the frequency of the carrier signal varies according to
the digital signal changes. FSK is a scheme of frequency
modulation.
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.

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 3. Phase Shift Keying

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 End of Slide

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