Programme : RTU- B.Tech.

in EE
Course Name : Principle of Electronic Communication
(Open-Elective), Credit: 3, 150 Marks
Course Code : 7EC6.60.1
Semester : VII
Session : 2020-2021

Faculty Name: : J.P. Vijay (Associate Professor-Department of

Electronics & Communication Engineering

Swami Keshvanand Institute of Technology, Management &

Gramothan
Ramnagaria, Jagatpura, Jaipur-302017, INDIA
Approved by AICTE, Ministry of HRD, Government of India,
Recognized by UGC under Section 2(f) of the UGC Act, 1956
E-mail: info@skit.ac.in Web: www.skit.ac.in
Lecture- 1:
Introduction: Objective, scope and outcome of the course.

Lecture 2-10:
Introduction: Need for Modulation, Frequency translation,
Electromagnetic spectrum,

Gain, Attenuation and decibels.

Simple description on Modulation: Analog Modulation-AM,

Frequency modulation-FM,
Pulse Modulation- PAM, PWM,

PCM,

Digital Modulation Techniques- ASK, FSK, PSK, QPSK

modulation and demodulation schemes
2
 Lecture-2

Principle of Electronic Communication

3
 BASIC INTRODUCTION TO SUBJECT

PRE-REQUISITES:
Knowledge of basic electonics components and electronic devices,
Signals Systems, Mathematical Analysis of Signals.

INTRODUCTION

Communications is the transfer of information or data (a digital bit

stream or analog signal) over a point-to-point or point-to-multipoint
communication channel .

Most common examples of such channels are optical fibers, wireless

communication channels .

The data are represented as an electromagnetic signal such as an

electrical voltage, radio wave, optical microwave, or infrared signal.
4
 COMMUNICATION BLOCK DIAGRAM

Analog Communication Digital Communication

5
Analog and Digital Communication:

The difference between Analog and Digital Communication is

that Analog communication uses analog signals for transmission and
reception of data while digital communication uses digital signals for
transmitting and receiving data.

TV and Radio signals are examples of analog communication.

Digital communication is a communication format in which information

is encoded as a series of discrete values. optical fibres and computer
buses make use of digital communication.

6
 MODULATION and DEMODULATION
Modulation is the process where some characteristics of periodic
waveform, called the carrier signal, is varied according with a
modulating signal that typically contains information to be transmitted.
Modulator is a device that performs modulation

Carrier signal C(t) = A cos ( ωct+ θ )

A= Amplitude of carrier signal (volts)

ωc = Frequency of carrier signal (radian /sec) = 2 π f
θ = Phase of the carrier signal (degree)

Demodulation (the inverse of modulation) is extracting the original

information-bearing signal from the RF carrier wave or modulated
signal. Demodulator (sometimes detector or demod) is a device that
performs demodulation.
7
MODULATION and DEMODULATION

8
MODULATION & DEMODULATION

9
 Lecture-3

Principle of Electronic Communication

-Why we need modulation ?

-Frequency Translation

-Electromagnetic Spectrum

10
 Why we need MODULATION ?

Following are some reasons to use modulation in the

communication systems.

1. To reduce the practical Height of the antenna

The height of the antenna is L= λ /4, λ= wavelength of transmitted

signal

The relation between λ (wavelength ) and f (frequency) is

C= λ .f so λ= c/f

So the height of the anteena L is =

f=20 KHz , L = 15 KM

f=1000 kHz, L = 300 m 11

 Why we need MODULATION ?

2. Avoid Interference- Audio frequencies are within the range of 20

Hz to 20 kHz. Without modulation all signals at same frequencies from
different transmitters would be mixed up. There by giving impossible
situation to tune to any one of them. In order to separate the various
signals, radio stations must broadcast at different frequencies. FM
Broadcast (88 MHz to 108 MHz)

3. Communication range increases.

4. Multiplexing is possible.

5. Improves quality of reception.

12
 Frequency Translation

Frequency transaction also known as frequency mixing /

frequency conversion / heterodyning

The process of transferring a signal form one band to another

band is called Frequency translation.

It is desirable to shift the modulated signal to a new center

frequency (before demodulation), in the processing of signal in
the communication system.

Now the received signal, translated to a fixed intermediate

frequency (IF), can easily be Amplified, Filtered and
Demodulated.

13
 Frequency Translation

For ex. – In commercial AM radio receiver, the received radio

frequency (RF) signal is 535-1605 kHz
.
But this is shifted to an intermediate frequency (IF), Which is 455 KHz
band for the purpose of processing.

This is done because the processing of the signal is easier on single

centered frequency.

The device which is used to perform this operation of frequency

translation of the modulated wave is known as frequency mixer.

For FM broadcast
RF range is 88 to 108 MHz, and IF is 10.7 MHz

14
 Frequency Translation

Superheterodyne Receiver

15
 Electromagnatic Spectrum

Electromagnetic waves or EM waves are waves that are created as a

result of vibrations between an electric field and a magnetic field. In
other words, EM waves are composed of oscillating magnetic and
electric fields

The electromagnetic (EM) spectrum is the range of all types of

EM radiation.

The velocity of em wave is equal to light velocity ( 3 x 108 m/s )

16
 Electromagnatic Spectrum
The other types of EM radiation that make up the electromagnetic
spectrum are microwaves, infrared light, ultraviolet light, X-rays and
gamma-rays.

Voice Spectrum or Voice frequency range: 300 Hz- 3400 Hz

Audio Band: 20 Hz- 20 KHz

Radio Wave: Above 20 KHz

AM spectrum: 535-1605 kHz

FM Spectrum: 88 MHz- 108 MHz

Optical Frequency Range: 300 GHz to 3000 THz

17
Electromagnatic Spectrum

18
 Electromagnatic Spectrum- ISM Bands

The ISM radio bands are portions of the radio spectrum reserved
internationally for industrial, scientific and medical (ISM) purposes
other than telecommunications

The ISM bands are defined by the ITU Radio Regulations

The ISM bands are Licence free band.

ISM Bands gives three microwave spectrum

902 MHz- 928 MHz (915 MHz Band)

2.4 GHz – 2.4835 GHz (2.4 GHz Band)

5.725 GHz- 5.860 GHz (5.8 GHz Band)

19
 Lecture-4
Principle of Electronic Communication

Topics discussed in the Topics to be discussed in

previous Lecture today’s Lecture

-Block diagram of Electromagnatic Spectrum

communication system
ISM Bands
-Modulation and
Demodulation

-Why we need Modulation ?

-Frequency Translation
20
 Electromagnatic Spectrum

✔Electromagnetic waves or EM waves are waves that are created as a

result of vibrations between an electric field and a magnetic field.

✔In other words, EM waves are composed of oscillating magnetic and

electric fields

✔The electromagnetic (EM) spectrum is the range of all types of

EM radiation.

✔It includes radio waves, microwaves, infrared, light, ultraviolet,

X-rays, and gamma rays.

✔The velocity of em wave is equal to light velocity ( 3 x 108 m/s )

21
 Electromagnatic Spectrum

Voice Spectrum or Voice frequency range: 300 Hz- 3400 Hz

Audio Band: 20 Hz- 20 KHz

Radio Wave: Above 20 KHz (20 KHz to 300 GHz)

AM spectrum: 535-1605 kHz

FM Spectrum: 88 MHz- 108 MHz

Microwaves 300 MHz to 300 GHz

Infrared Range: 300 GHz to 400 THz

Optical Frequency Range: 400 THz to 700 THz (700 X 1012 Hz)
22
 Electromagnatic Spectrum

✔The radio wave, microwaves and Optical spectrum are used for
communication.

✔The UV waves, X-Rays, and Gamma Rays would be better

because of higher frequencies, but these are difficult to generate,
modulate and do not propagate well through buildings.

✔Also, these high frequencies are dangerous for health or

living, thatswhy these ways are not used for communication.

23
 Electromagnatic Spectrum (ITU)
S.No. Frequency Band Frequency Range Applications

1 ELF (Extremely low frequency) 30 to 300 Hz Electric power

transmission,Domestic
2 VF (Voice frequency) 300 Hz to 3000 Hz Speech,
telecommunication
3 VLF (Very low frequency) 3 KHz to 30 KHz Submarine (under
sea), navigation
4 LF ( Low frequency) 30 KHz to 300 KHz Marine
communication
5 MF (Medium frequency) 300 KHz to 3000 KHz AM Broadcasting

6 HF (High frequency) 3 MHz to 30 MHz Military, Broadcasting

7 VHF (Very High frequency) 30 MHz to 300 MHz FM, Television

8 UHF (Ultra High Frequency) 300 MHz to 3000 MHz Mobile, Radar

9 SHF (Super High Frequency) 3 GHz to 30 GHz Satellite, WLL

10 EHF (Extremely High Frequency) 30 GHz to 300 GHz WLL

24
 Electromagnatic Spectrum (ITU)

S.No. Frequency Band Frequency Range Applications

11 Infrared 300 GHz to 400 THz Remote control

Infrared LANs

12 Visible light 400 THz to 700 THz Fiber optics

13 X-Rays 1016 to 1022 Hz Medical

14 Gamma Rays 10 22 to 1028 Hz Medical, Nucluear

Science

25
 Electromagnatic Spectrum- ISM Bands

The ISM radio bands are portions of the radio spectrum reserved
internationally for industrial, scientific and medical (ISM) purposes
other than telecommunications

The ISM bands are defined by the ITU Radio Regulations

The ISM bands are Licence free band.

ISM Bands gives three microwave spectrum

902 MHz- 928 MHz (915 MHz Band)

2.4 GHz – 2.4835 GHz (2.4 GHz Band)

5.725 GHz- 5.860 GHz (5.8 GHz Band)

26
 Lecture-5
Principle of Electronic Communication

Topics discussed in the Topics to be discussed in

previous Lecture today’s Lecture

Attenuation
Electromagnatic Spectrum
Gain
ISM Bands
Decibel

Related Numericals

27
 Gain, Attenuation and decibels
TRANSMISSION IMPAIRMENT: The signal at the beginning of
the communication channel is not the same as the signal at the end
of the channel. What is sent is not what is received. Three causes
of impairment are attenuation, distortion, and noise.

Most circuits in communication are used to manupulate the signals

to produce a desired output.

All signal processing circuit involve

------Attenuation

----- Gain

Attenuation and Gain are unit less and these are measured in dB
28
 Attenuation

✔Attenuation is a general term that refers to any reduction in the

strength of a signal. Attenuation occurs with any type of signal,
whether digital or analog.

✔Means loss of energy -> weaker signal

✔When a signal travels through a medium it loses energy

overcoming the resistance of the medium

✔Amplifiers are used to compensate for this loss of energy by

amplifying the signal.

29
Attenuation and Gain

30
 Gain
Gain means amplification and it is the ratio of circuit output to input.

31
 Attenuation and Gain

• To show the loss or gain the unit “decibel” is used.

Attenuation Gain
dB = 10log10P1/P2 dB = 10log10P2/P1
P1 - input signal P1 - input signal
P2 - output signal P2 - output signal

32
 Gain
An amplifier is cascaded when two or more stages are connected
together, the overall gain is the product of individual circuit gain

Ex. Three cascade amplifiers have power gain of 4, 5, and 8. The

input power is 50 mW. What is the ouput power

Sol: Total Gain = 4 x 5 x 8 = 160

Since Gain = P2/P1

P1= 50 miliwatt,

Gain- 160

P2= Gain x P1 = 160 x (50 X 10-3) = 8 W

33
 Attenuation and Gain
Example 2- Suppose a signal travels through a transmission
medium and its power is reduced to one-half. Calculate the
attenuation or Gain

34
 Attenuation and Gain
Example 2- A signal travels through a transmission medium and
its power is reduced to one-half. Calculate the attenuation or
Gain

A loss of 3 dB (–3 dB) is equivalent to losing one-half the power.

35
 Attenuation and Gain
A signal travels through an amplifier, and its power is increased
10 times. This means that P2 = 10P1 . In this case, the
amplification (gain of power) can be calculated as

36
 Decibel (Db)
✔The decibel (dB) is a logarithmic unit used to measure sound level. It
is also widely used in electronics, signals and communication.

✔Decibel is a unit of measure used to express the gain and attenuation

of a circuit.

✔A decible is one tenth of a bel. The dB is a logarithmic way of

describing a ratio. The ratio may be power, sound pressure, voltage or
intensity

On the decibel scale, the smallest audible sound (near total silence) is
0 dB. A sound 10 times more powerful is 10 dB. A sound 100 times
more powerful than near total silence is 20 dB. A sound 1,000 times
more powerful than near total silence is 30 dB

37
Decibel (Db)

38
39
 Lecture-6
Principle of Electronic Communication

Topics discussed in the Topics to be discussed in today’s

previous Lecture Lecture

Attenuation Simple description on Modulation:

Analog Modulation-
Gain
Amplitude Modulation (AM)
Decibel
Equation of AM wave
Related Numericals
Waveforms of AM wave

40
 Analog Modulation
If the modulating signal or information signal or message signal
or baseband signal is in analog form then the modulation is
called analog modulation and communication is known as
analog communication.

An analog signal is continuous in both time and amplitude.

Analog signals in the real world include current, voltage,

temperature, pressure, light intensity, and so

41
 Analog Modulation

Modulation is the process where some characteristics of periodic

waveform, called the carrier signal, is varied according with a
modulating signal that typically contains information to be
transmitted. Modulator is a device that performs modulation

Carrier signal C(t) = A cos ( ωct+ θ )

A= Amplitude of carrier signal (volts)

ωc = Frequency of carrier signal (radian /sec) = 2 π f
θ = Phase of the carrier signal (degree)

42
 Amplitude and Frequency Modulation
■ Amplitude Modulation (AM)
■ Amplitude modulation is the process where the amplitude of
the carrier signal vary or change according the instantaneous
value of the modulation signal.

■ In the process of AM, the frequency and phase of the carrier

remains constant

■ Frequency Modulation (FM)

■ Frequency modulation is the process where the amplitude of
the carrier signal vary or change according the instantaneous
value of the modulation signal.

■ In the process of FM, the amplitude of the carrier remains

constant
43
 Amplitude Modulation

✔Carrier amplitude changes according the modulating signal

✔ Frequency and Phase of carrier remain constant

✔Widely used in video transmission, broadcasting, television

44
 Amplitude Modulation
Amplitude modulation equation:

Carrier equation- C(t) = A cos ( ωct+ θ ) Phase θ = 0

So C(t) = A cos ωct

Modulation Signal x(t)

Then the equation of the amplitude modulated signal will be

S(t) = [A + x(t)]. cos ωct

Here [A + x(t)] is the enevelope of AM wave

45
 Amplitude Modulation-waveforms
Carrier C(t) = A cos ωct Modulation or information Signal x(t)

AM signal s(t) = [A + x(t)]. cos ωct

46
AM AND FM
WAVEFORM

47
 Single Tone- Amplitude Modulation
Modulation Signal x(t)

✔Till now, we discussed amplitude modulation in which we assumed

that baseband or modulating signal is a random signal which
contains a large number of frequency components.

✔This means that a carrier signal (fixed frequency signal) is modulated

by a large number of frequency components.

✔Here, we will discuss amplitude modulation in which the modulating

or baseband signal consists of only one (single) frequency i.e.
modulation is done by a single frequency or tone. This type of
modulation is known as single tone amplitude modulation .

✔So Modulation Signal x(t) = Vm cos ωmt ωm = Frequency of

modulating signal.
48
 Equation of -Single Tone Amplitude Modulation
Modulation Signal x(t) = Vm cos ωmt eq. 1

Carrier Signal C(t) = A cos ωct eq. 2

AM signal s(t) = [A + x(t)]. cos ωct eq.3

For single tone AM, putting eq.1 in eq.3

Single tone AM, S(t) = [A + Vm cos ωmt ]. cos ωct eq.4

S(t)= A cos ωct + Vm .cos ωmt . cos ωct

S(t) = A cos ωct + Vm /2 cos (ωc + ωm ) + Vm /2 cos (ωc -ωm )

49
 Equation of -Single Tone Amplitude Modulation
S(t)= A cos ωct + Vm .cos ωmt . cos ωct

S(t)= A cos ωct ( 1+ ma .cos ωmt )

= A cos ωct + A ma .cos ωmt . cos ωct

50
Equation of -Single Tone Amplitude Modulation

AM Signal has three components

51
Equation of -Single Tone Amplitude Modulation

52
 Pulse Modulation
Pulse modulation is a type of modulation in which the signal is
transmitted in the form of pulses. It can be used to transmit analogue
information. In pulse modulation, continuous signals are sampled at
regular intervals.

53
54
55
Any Query

56
 ANALOG TO DIGITAL CONVERSION-PCM
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.

✔The occurrence of cross-talk is very rare in digital communication.

So, the processing of digital Signal is eaiser compare to analog

communication 57
 ANALOG TO DIGITAL CONVERSION
Analog Signal: Continuous in time and Amplitude

Digital Signal: Discrete in time and Amplitude

To convert the analog singal in Digital, we need to discrete the time and
amplitude of the signal .

The process to Discrete the time- SAMPLING

The process to Discrete the Amplitude- QUANTIZATION 58

 PULSE CODE MODULATION

Pulse code modulation is a method that is used to convert an analog

signal into a digital signal so that a modified analog signal can be
transmitted through the digital communication network.

59
 ANALOG TO DIGITAL CONVERSION- PCM

Sampling Theorem: Sampling theorem states that “A continues time

signal can be represented in discrete time singal (samples) and
recovered back, if the sampling frequency is greater than equals to
maximum frequency of modulating signal”. Fs>=2Fm.

When sampling frequency equals twice the input signal frequency is

known as “Nyquist rate or minimum sampling rate”. Fs=2Fm

60
 ANALOG TO DIGITAL CONVERSION

Sampling Theorem: Sampling theorem states that “A continues time

signal can be represented in discrete time singal (samples) and
recovered back, if the sampling frequency is greater than equals to
maximum frequency of modulating signal”. Fs>=2Fm.

When sampling frequency equals twice the input signal frequency is

known as “Nyquist rate or minimum sampling rate”. Fs=2Fm

61
 ANALOG TO DIGITAL CONVERSION

Quantization: The digitization of analog signals involves the rounding

off of the values which are approximately equal to the analog values.
The method of sampling chooses a few points on the analog signal and
then these points are joined to round off the value to a near stabilized
value. Such a process is called as Quantization.

Quantization is the process of converting a continuous range of values

into a finite range of discreet values.

62
 4-2 ANALOG-TO-DIGITAL
CONVERSION
A digital signal is superior to an analog signal because it
is more robust to noise and can easily be recovered,
corrected and amplified. For this reason, the tendency
today is to change an analog signal to digital data. In
this section we describe two techniques, pulse code
modulation and delta modulation.

Topics discussed in this section:

▪ Pulse Code Modulation (PCM)
▪ Delta Modulation (DM)

4.63
 PCM
• PCM consists of three steps to digitize an
analog signal:
1. Sampling
2. Quantization
3. Binary encoding
▪ Before we sample, we have to filter the signal
to limit the maximum frequency of the signal as
it affects the sampling rate.
▪ Filtering should ensure that we do not distort
the signal, ie remove high frequency
components that affect the signal shape.

4.64
Figure 4.21 Components of PCM encoder

4.65
 Sampling
• Analog signal is sampled every TS secs.
• Ts is referred to as the sampling interval.
• fs = 1/Ts is called the sampling rate or sampling
frequency.
• There are 3 sampling methods:
– Ideal - an impulse at each sampling instant
– Natural - a pulse of short width with varying
amplitude
– Flattop - sample and hold, like natural but with single
amplitude value
• The process is referred to as pulse amplitude
modulation PAM and the outcome is a signal with
analog (non integer) values
4.66
Figure 4.22 Three different sampling methods for PCM

4.67
 Note

According to the Nyquist theorem, the

sampling rate must be
at least 2 times the highest frequency
contained in the signal.

4.68
Figure 4.23 Nyquist sampling rate for low-pass and bandpass signals

4.69
 Example 4.6

For an intuitive example of the Nyquist theorem, let us

sample a simple sine wave at three sampling rates: fs = 4f
(2 times the Nyquist rate), fs = 2f (Nyquist rate), and
fs = f (one-half the Nyquist rate). Figure 4.24 shows the
sampling and the subsequent recovery of the signal.

It can be seen that sampling at the Nyquist rate can create

a good approximation of the original sine wave (part a).
Oversampling in part b can also create the same
approximation, but it is redundant and unnecessary.
Sampling below the Nyquist rate (part c) does not produce
a signal that looks like the original sine wave.
4.70
Figure 4.24 Recovery of a sampled sine wave for different sampling rate

4.71
 Example 4.7

Consider the revolution of a hand of a clock. The second

hand of a clock has a period of 60 s. According to the
Nyquist theorem, we need to sample the hand every 30 s
(Ts = T or fs = 2f ). In Figure 4.25a, the sample points, in
order, are 12, 6, 12, 6, 12, and 6. The receiver of the
samples cannot tell if the clock is moving forward or
backward. In part b, we sample at double the Nyquist rate
(every 15 s). The sample points are 12, 3, 6, 9, and 12.
The clock is moving forward. In part c, we sample below
the Nyquist rate (Ts = T or fs = f ). The sample points are
12, 9, 6, 3, and 12. Although the clock is moving forward,
the receiver thinks that the clock is moving backward.
4.72
Figure 4.25 Sampling of a clock with only one hand

4.73
 Example 4.8

An example related to Example 4.7 is the seemingly

backward rotation of the wheels of a forward-moving car
in a movie. This can be explained by under-sampling. A
movie is filmed at 24 frames per second. If a wheel is
rotating more than 12 times per second, the
under-sampling creates the impression of a backward
rotation.

4.74
 Example 4.9

Telephone companies digitize voice by assuming a

maximum frequency of 4000 Hz. The sampling rate
therefore is 8000 samples per second.

4.75
 Example 4.10

A complex low-pass signal has a bandwidth of 200 kHz.

What is the minimum sampling rate for this signal?

Solution
The bandwidth of a low-pass signal is between 0 and f,
where f is the maximum frequency in the signal.
Therefore, we can sample this signal at 2 times the
highest frequency (200 kHz). The sampling rate is
therefore 400,000 samples per second.

4.76
 Example 4.11

A complex bandpass signal has a bandwidth of 200 kHz.

What is the minimum sampling rate for this signal?

Solution
We cannot find the minimum sampling rate in this case
because we do not know where the bandwidth starts or
ends. We do not know the maximum frequency in the
signal.

4.77
 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

4.78
 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.

4.79
 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
4.80
 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.

4.81
Figure 4.26 Quantization and encoding of a sampled signal

4.82
 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

4.83
 Quantization Error and SNQR
• Signals with lower amplitude values will suffer
more from quantization error as the error range:
Δ/2, is fixed for all signal levels.
• Non linear quantization is used to alleviate this
problem. Goal is to keep SNQR fixed for all sample
values.
• Two approaches:
– The quantization levels follow a logarithmic curve.
Smaller Δ’s at lower amplitudes and larger Δ’s at
higher amplitudes.
– Companding: The sample values are compressed at
the sender into logarithmic zones, and then expanded
at the receiver. The zones are fixed in height.

4.84
 Bit rate and bandwidth requirements of
PCM
• The bit rate of a PCM signal can be calculated form the
number of bits per sample x the sampling rate
Bit rate = nb x fs
• The bandwidth required to transmit this signal depends
on the type of line encoding used. Refer to previous
section for discussion and formulas.
• A digitized signal will always need more bandwidth than
the original analog signal. Price we pay for robustness
and other features of digital transmission.

4.85
 Example 4.14

We want to digitize the human voice. What is the bit rate,

assuming 8 bits per sample?

Solution
The human voice normally contains frequencies from 0
to 4000 Hz. So the sampling rate and bit rate are
calculated as follows:

4.86
 PCM Decoder

• To recover an analog signal from a digitized signal

we follow the following steps:
– We use a hold circuit that holds the amplitude value
of a pulse till the next pulse arrives.
– We pass this signal through a low pass filter with a
cutoff frequency that is equal to the highest
frequency in the pre-sampled signal.
• The higher the value of L, the less distorted a
signal is recovered.

4.87
Figure 4.27 Components of a PCM decoder

4.88
Pulse code Modulation (PCM)

89
 Quantization

To keep the quantization error small relative to the message signal

level, use smaller quantization steps.. 90