DIGITIZATION OF ONE-DIMENSION

SIGNALS
EEEN 464 – DIGITAL COMMUNICATION
Friday, January 24, 2025
 RECAP: ADVANTAGES OF DIGITAL COMMUNICATION NETWORKS /1
1. Ease of multiplexing 𝑠1
𝑠2 MUX 𝑠𝑚
𝑠𝑛
Tel 1 Tel 11
2. Ease of Signaling
Tel 2 EXCH A1 EXCH B1 Tel 12

Tel n Tel 1n

Signaling Issues
• Tel Exch B, I want Tel 1n
• Is there free link to 1n?
• Notify Tel 1 that Tel 1n is ringing
• Notify Exch. A1 that Tel 1n has disconnected.
3. Uses modern computer
technology
4. Integration of transmission and
switching
5. Signal Regeneration
 RECAP: ADVANTAGES OF DIGITAL COMMUNICATION NETWORKS /2

6. Advanced Performance
Monitoring.
7. Ability to integrate other
services.
8. Ability to operate at low Signal-
to-Noise Ratio (SNR).
9. Ease of encryption.

𝒔𝟏 𝒔𝒆 ≠ 𝒔𝟏 𝒔𝟏

𝒌𝟏 𝒌𝟐
Symmetric encryption
𝒌𝟏 = 𝒌𝟐
 ANALOG Vs DIGITAL MULTIPLEXING

1. A multiplexer combines (mux)

multiple signals into a single
composite signal. The composite
signal is transmitted over a shared
medium, such as a fiber optic cable
(WDM) or radio wave (FDM).
2. A demultiplexer (demux) separates
the composite signal back into the
original signals.
3. The Cost of digital multiplex
systems, e.g. TDM is much lower
that that of analogue multiplex
systems, e.g. FDM.
SIGNALLING IN ANALOG TELECOMMUNICATION SYSTEM - LOCAL
1. Signaling in telecommunications is the process of using
signals to control and manage communication networks.
2. Telecommunication signals include: dialed digits, status of
the call e.g. telephone ringing, set-up and tear-down
information.
Telephone
Subscriber A Exchange Subscriber B

Service request
Dialed digits
0724770687
Ringing tone Ringing signal
. Answer
. .
Clear forward .
Clear forward
SIGNALLING IN ANALOG SYSTEM WITH TRANSIT EXCHANGE
 SIGNALLING IN DIGITAL SYSTEMS
1. Digital Signaling systems allow control information to be
inserted into and extracted from a message stream independent
of the mode of transmission.
2. Signaling equipment is designed separate from transmission
systems allowing control functions and formats to be designed
and modified independently.
 USE OF MODERN TECHNOLOGY-LSI & VLSI
1. Multiplexer and switching matrix for
digital systems are implemented with
the same basic circuits used in
computers.
2. Special LSI Circuits have been
developed specifically for
telecommunication functions e.g Voice
Codecs, Multiplexing, DSPs, etc.
3. Low-cost of digital circuitry allow for
implementations that would be very
expensive if developed on analogue
platforms, e.g large non-blocking
exchanges.
4. Digital technology provides easier and
cheaper interfaces to fibre-optic cable
systems.
 INTEGRATION OF TRANSMISSION AND SWITCHING`

1. Digital communication allows data

to be terminated directly onto
telephone switches unlike analog Frequency division link
systems which require modulation
and demodulation in separate
channel banks.
2. Cable entrance requirements and
distribution of wire pairs is greatly
reduced because all trunks are
implemented as sub-channels in a
TDM signal.
 SIGNAL REGENERATION

Digital Signals can be regenerated at suitable intervals unlike analogue

signals.
 BENEFITS OF FULLY INTEGRATED DIGITAL COMMUNICATION
NETWORKS
1. Long-distance and local voice quality are
identical in terms of noise, signal level, and
distortion.
a) At physical layer, repeaters regenerate clean
pulses from distorted ones.
b) At data link layer, error detection and correction
are used.

2. Since digital baseband circuits are inherently four-

wire, network-generated echoes are eliminated,
and true full-duplex, four-wire digital circuits are
available.
 RECAP: OPEN SYSTEMS INTERCONNECT MODEL

1. Open Systems Interconnection

(OSI) model is a conceptual
framework that describes how
data travels across a network.
2. OSI was created by the
International Standards (ISO) and
published in 1984.
ENCRIPTION: SUBSTITUTION & TRANSPOSITION CIPHERS
Substitution ciphers replace each group of letters in the message with another
group of letters to disguise it

Simple single-letter substitution cipher

Transposition ciphers reorder letters to disguise them

Key gives column order

Column 5 6 7 8

Simple column transposition cipher

DIGITIZATION OF SPEECH

EEEN 464 – DIGITAL COMMUNICATION

Friday, January 24, 2025
 HUMAN HEARING & VOICING FREQUENCY RANGE

1. Human ear can detect sounds with frequencies

between 20 Hz and 20,000 Hz.
2. Sound with frequencies below the human
hearing range is referred to as infrasound.
3. Sound with frequencies above is called
ultrasound.
4. Humans produce two types of sounds:
a) Voiced sounds are made when the vocal folds
vibrate.
b) Voiceless sounds are made when the vocal folds
do not vibrate.
 SOUND FREQUENCY SPECTRUM
Infrasound: Voice (Acoustic) Ultrasound:
Audible to: Audible to: Humans Applications: Medical Imaging, Non-
Elephants, Destructive Examination(NDE)
Whales, Hippos Frequency range: 1 – 20 MHz (medical
Ultrasound Imaging)

Infrasound Acoustic Ultrasound

0 Hz 20 Hz 20 kHz 200 kHz 2 MHz 200 MHz

1. Fundamental voicing frequency in humans varies from 85 Hz to 1,100 Hz.

Women's voices are generally higher in pitch than men's voices.
2. Harmonic series of a voice is the series of frequency components above the
fundamental frequency.
 VOICED Vs UNVOICED SOUNDS
1. Voiced sounds produced with the
vocal cords vibrating.
a) Vowels: Sounds ( a, e, i, o, u)
Frequency: 250 – 2,000 Hz
a) Consonants e.g. b, d, g, m, n,
l, r, v, z
Frequency: 250 – 4,000 Hz
2. Voiceless sounds
a) Consonants, e.g. p, t, k, f, s, sh,
ch
Frequency: 2,000 – 8,000 Hz
HEARING & VOICING AMONGST DIFFERENT ANIMALS
 TELEPHONE FREQUENCY RANGE
1. The frequency range for a
standard telephone is 300–3,400
hertz (Hz). This range is known as
the voiceband in telephony.
2. The voiceband was chosen
because:
1. Acceptable level of
intelligibility is obtained by
transmitting voice in range 0.3
-3.4 KHz
2. Most of the voice energy is
concentrated in this band.
3. Wideband audio (HD Voice)
has frequency range to 50 Hz
to 7,000 Hz.
 SAMPLING THEORY

Sampling theorem can be stated by any of the

following:
1. A signal that is band-limited of finite energy at
a frequency fm can be completely described by
samples taken at a uniform time intervals of no
1
less than apart.
2𝑓𝑚
2. A band-limited signal of finite energy with no
frequency above 𝒇𝒎 may be adequately
recovered from samples taken at the rate
2𝑓𝑚 samples per second.
 MATCHING THEORY TO SAMPLING REQUIREMENTS

REQUIREMENT SAMPLING THEORY

1. There should be sufficient number of A signal that is band-limited of finite energy at a
samples so that the original signal is frequency fm can be completely described by
adequately represented samples taken at a uniform time intervals of no less
1
than apart.
2𝑓𝑚
2. It should be possible to reconstruct the A band-limited signal of finite energy with no
original signal from its samples. frequency above 𝑓𝑚 may be adequately recovered
from samples taken at the rate 2𝑓𝑚 samples per
second.
 PULSE-AMPLITUDE MODULATION
Pulse-amplitude modulation (PAM) is a form of signal modulation where
the message information is encoded in the amplitude of a series of signal
samples.

Nyquist Criterion/Theorem
• fs > 2fmax where fmax is the highest frequency in the analog input
signal
 WHY ARE WE INTERESTED IN FOURIER TRANSFORM OF PAM SIGNALS?

1. Fourier transform of a
signal represents the
signal in the frequency
domain.
2. The Fourier transform of
the PAM signals assists
us to design the
parameters of the low-
pass reconstruction filter.
 RECAP: FOURIER TRANSFORM OF A PULSE

1. The spectrum of a single pulse has a sin x/x shape and covers a
band of frequencies.
2. The width of the central lobe of the spectrum of a single pulse is
inversely proportional to the pulse width.
RECAP: FOURIER TRANSFORM OF A PULSE TRAIN

Duty cycle of 0.27: The

spectral content at closest
to 3f is quite small.
• At a duty-cycle of exactly
one-third, the spectral
content at 3f would be
zero.

Duty cycle is 0.5: The

spectral content at 2f (and
4f and 6f etc..) is always
zero.
 PULSE AMPLITUDE MODULATION (PAM) SPECTRUM
1. Low-pass filter is used to
demodulate PAM signals by
passing the baseband and
removing higher
frequencies.
2. Cutoff frequency of the
low-pass filter should be
large enough to
accommodate the highest
frequency component of
the message signal.
3. Reconstructed signal may
have some amplitude
distortion due to the
aperture effect.
 ALIASING/FOLD-OVER DISTORTION

1. Aliasing/fold-over occurs when the sampling frequency 𝑓𝑠 is less than

the Nyquist frequency, 2fmax resulting in an overlap of the spectrum.

2. Aliasing/fold-over distortion is avoided in traditional telephony by:

a) Band-limiting the signal to the range 0.3-3.4 KHz when the
sampling frequency is 4 KHz.
b) over-sampling at 8KHz instead of 6.8 KHz which is the Nyquist rate.
 EXAMPLE OF ALIASING DISTORTION

As shown above, a signal of 5.5 KHz sampled at 8 KHz appears to be a

2.5KHz signal (dotted).
 END-TO-END PAM COMMUNICATION SYSTEM

Transmitter Receiver

The response of the reconstructive filter is usually modified to account

for the spectrum of the wider staircase samples.
 BAND-LIMITING FILTER DESIGNED TO MEET INTERNATIONAL TELECOMMUNICATION
UNION (ITU) RECOMMENDATION FOR PCM VOICE CODERS

1. ITU Standard for

telephone speech
sampling requires14 dB
attenuation is provided at 4
KHz.
2. This can be restated as
‘the signal strength in
watts at 4 KHz is 4% that
at 3.4 KHz.
 SAMPLING TECHNIQUES

There are three sampling techniques, i.e Ideal, Natural and Flat-top as
shown below.
 IDEAL SAMPLING

• Ideal Sampling (Instantaneous sampling or Impulse Sampling)

uses a train of impulses and the principle used is known as
multiplication principle.
 NATURAL SAMPLING

1. Natural sampling multiplies

the original signal by a train of
rectangular sampling pulses
with unit amplitude
2. The output signal follows the
original waveform's contours.
3. Natural sampling is usually
realized using a MOSFET
switch as shown.
 FLAT-TOP SAMPLING

• 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.
PULSE CODE MODULATION
(PCM)
EEEN 464 – DIGITAL COMMUNICATION
Friday, January 24, 2025
 PULSE CODE MODULATION (1)

1. Pulse-code modulation (PCM) is a method used to digitally

represent analog signals.
2. PCM is used in digital audio in computers, compact discs, digital
telephony and other digital audio applications.
 BLOCK DIAGRAM OF PCM COMMUNICATION SYSTEM

Low-pas Filter(LPF) Encoder converts quantized

Anti-aliasing filter which values into binary codes
removing the high-frequency
components.

Regenerative
repeater is used to
compensate for the
signal loss and also
resynchronize the
Reconstruction filter is a low-pass filter
signal.
helps to obtain the original signal.
 WORKED EXAMPLE - 01

Calculate the Nyquist rate for a signal given by:

x(t ) = 3 cos(50t ) + 10 sin(300t ) − cos(100t )

Solution:
x(t ) = 3 cos(2f1t ) + 10 sin(2f 2t ) − cos(2f 3t )

x(t ) = 3 cos(2 25t ) + 10 sin(2 150t ) − cos(2 50t )

Maximum frequency is therefore 150Hz and the maximum

Nyquist rate is fs = 2x150 = 300 Samples/Sec
 WORKED EXAMPLE - 02

Find the Nyquist frequency/rate and the Nyquist interval

for the following signal:
1
x(t ) = cos(4000t ) cos(1000t )
2
 WORKED EXAMPLE - 02 (SOLUTION)

2 cos(A) cos(B) = cos(A + B) + cos(A − B)

1
x(t ) = cos(4000t ) cos(1000t )
2

x(t ) =
1
(cos(4000t + 1000t ) + cos(4000t − 1000t ) )
4
or
1
x(t ) = (cos(5000t ) + cos(3000t ))
4

The maximum frequency is therefore 2,500Hz which implies that the

Nyquist rate is 5,000 samples/sec and Nyquist interval is 1/5000 = 0.2msec
 WORKED EXAMPLE /03

A PCM telephone system has a band-limiting filter of 4 KHz. If

each sample is sampled, quantized at 256 levels, calculate the
bit-rate of the PCM system.

ANS:
1. According to the Nyquist criterion, we must sample at
twice the maximum baseband frequency = 2 x 4 KHz or
8,000 samples/sec.
2. To represent 256 quantized level, the system requires n
bits to represent each sample, where 2𝑛 = 256 or n = 8.
3. The PCM bit rate is therefore 8 x 8,000 = 64 Kbits/s
LINEAR PULSE CODE
MODULATION (LPCM)
EEN 464 – DIGITAL COMMUNICATION
Friday, January 24, 2025
 WHAT IS LPCM?
1. Linear Pulse Code Modulation (LPCM) is a digital audio format that
records sound without compression.
2. Linear Pulse Code Modulation (LPCM) that uses linear
quantization.
3. LPCM Works as follows:
a) Analog signal is sampled at regular intervals
b) Amplitude is quantized at uniform levels and encoded into a
series of digital symbols, usually binary
c) The resulting digital representation is stored as a series of
numbers
4. LPCM is used in audio CDs and WAV files
 LPCM AUDIO FORMATS /01
LPCM encoding is used in the following audio formats.
1. WAV (Waveform Audio File) developed by IBM and Microsoft in
1991 enables storing audio bitstreams. WAV organizes data in
chunks and is the primary format for LPCM audio on Microsoft
Windows platforms.
2. AC3 known commercially as Dolby Digital supports up to 5
channels, including left, right, center, left surround, right
surround, and a low-frequency effects channel.
3. AES3 exchanges digital audio signals between audio devices.
AES3 can transmit two channels of PCM signals or audio over
diverse media, such as unbalanced lines, optical fiber, and
more. AES3 the Audio Engineering Society (AES) and the
European Broadcasting Union (EBU).
 LPCM AUDIO FORMATS /02

4. MPEG audio (Moving Picture Experts Group), is an organization

that creates standards for encoding digital audio and video. It
fully supports LPCM audio signals and is one of the most
commonly used audio formats today.
5. Audio Interchange File Format (AIFF) was developed by Apple
in 1988, is a versatile audio file format that stores sound data for
electronic audio devices and personal computers.
 CLASS EXERCISE

Download the same music file from https://pixabay.com/ in WAV

and MP3 and compare the two formats in terms of file size and
quality of the music.