Unit II :Modulation and

Multiplexing Techniques

Digital-to-digital Conversion: Line Coding, Line Coding Schemes, Block Coding, Scrambling,
Analog to digital Conversion: Pulse Code Modulation (PCM), Delta Modulation (DM),
ADM Transmission modes: parallel transmission, serial transmission,
Analog-to-analog Conversion: Amplitude Modulation, Frequency Modulation, Phase Modulation,
Multiplexing: Frequency-Division Multiplexing (FDM),
Wavelength-Division Multiplexing Synchronous Time-Division Multiplexing,
Statistical Time-Division Multiplexing Spread Spectrum: Frequency Hopping Spread Spectrum (FHSS),
Direct Sequence Spread Spectrum.
4.1 Copyright © The McGraw-Hill Companies, Inc. Permission required for reproduction or display.
 4-1 DIGITAL-TO-DIGITAL CONVERSION

In this section, we see how we can represent digital

data by using digital signals. The conversion involves
three techniques: line coding, block coding, and
scrambling. Line coding is always needed; block
coding and scrambling may or may not be needed.

Topics discussed in this section:

▪ Line Coding
▪ Line Coding Schemes
▪ Block Coding
▪ Scrambling
4.2
 Line Coding

■ Converting a string of 1’s and 0’s

(digital data) into a sequence of signals
that denote the 1’s and 0’s.
■ For example a high voltage level (+V)
could represent a “1” and a low voltage
level (0 or -V) could represent a “0”.

4.3
 Figure 4.1 Line coding and decoding

4.4
 Mapping Data symbols onto
Signal levels
■ A data symbol (or element) can consist of a
number of data bits:
■ 1 , 0 or
■ 11, 10, 01, ……
■ A data symbol can be coded into a single
signal element or multiple signal elements
■ 1 -> +V, 0 -> -V
■ 1 -> +V and -V, 0 -> -V and +V
■ The ratio ‘r’ is the number of data elements
carried by a signal element.
4.5
Channel Capacity

■
Data rate
■
In bits per second
■
Rate at which data can be communicated
■
Bandwidth
■
In cycles per second of Hertz
■
Constrained by transmitter and medium
■
Baud rate
■
Frequency with which the components change

6
Baud Rate

It is number of times signal changes state per second.

(Ex. Ideal digital signal has two states 1 and 0, so per sec baud rate ?? 1 or 2
Here, 1 baud = a bit/sec)

It is measure for data transmission speed.

❑ Data rate rarely the same as baud rate

 Relationship between data
rate and signal rate
■ The data rate defines the number of bits sent
per sec - bps. It is often referred to the bit
rate.
■ The signal rate is the number of signal
elements sent in a second and is measured in
bauds. It is also referred to as the modulation
rate.
■ Goal is to increase the data rate while
reducing the baud rate.

4.8
 Figure 4.2 Signal element versus data element

4.9
 Data rate and Baud rate

■ The baud or signal rate can be

expressed as:
S = c x N x 1/r bauds
where N is data rate
c is the case factor (worst, best & avg.)
r is the ratio between data element &
signal element

4.10
 Example 4.1

A signal is carrying data in which one data element is

encoded as one signal element ( r = 1). If the bit rate is
100 kbps, what is the average value of the baud rate if c is
between 0 and 1?

Solution
We assume that the average value of c is 1/2 . The baud
rate is then

4.11
 Example 4.1

A signal is carrying data in which one data element is

encoded as one signal element ( r = 1). If the bit rate is
100 kbps, what is the average value of the baud rate if c is
between 0 and 1?

Solution
We assume that the average value of c is 1/2 . The baud
rate is then

4.12
 Note

Although the actual bandwidth of a

digital signal is infinite, the effective
bandwidth is finite.

4.13
 Example 4.2

The maximum data rate of a channel is

Nmax = 2 × B × log2 L (defined by the Nyquist formula).
Does this agree with the previous formula for Nmax?

Solution
A signal with L levels actually can carry log2L bits per
level. If each level corresponds to one signal element and
we assume the average case (c = 1/2), then we have

4.14
 Line encoding

■ DC components - when the voltage

level remains constant for long periods
of time, there is an increase in the low
frequencies of the signal. Most channels
are bandpass and may not support the
low frequencies.
■ This will require the removal of the dc
component of a transmitted signal.

4.15
 Line encoding

■ Self synchronization - the clocks at the

sender and the receiver must have the
same bit interval.
■ If the receiver clock is faster or slower it
will misinterpret the incoming bit
stream.

4.16
 Figure 4.3 Effect of lack of synchronization

4.17
 Example 4.3

In a digital transmission, the receiver clock is 0.1 percent

faster than the sender clock. How many extra bits per
second does the receiver receive if the data rate is
1 kbps? How many if the data rate is 1 Mbps?
Solution

4.18
 Example 4.3

In a digital transmission, the receiver clock is 0.1 percent

faster than the sender clock. How many extra bits per
second does the receiver receive if the data rate is
1 kbps? How many if the data rate is 1 Mbps?
Solution
At 1 kbps, the receiver receives 1001 bps instead of 1000
bps.

At 1 Mbps, the receiver receives 1,001,000 bps instead of

1,000,000 bps.

4.19
 Line encoding

■ Error detection - errors occur during

transmission due to line impairments.
■ Some codes are constructed such that
when an error occurs it can be
detected.
■ For example: a particular signal
transition is not part of the code. When
it occurs, the receiver will know that a
symbol error has occurred.
4.20
 Line encoding

■ Noise and interference - there are

line encoding techniques that make the
transmitted signal “immune” to noise
and interference.
■ This means that the signal cannot be
corrupted, it is stronger than error
detection.

4.21
 Line encoding

■ Complexity - the more robust and

resilient the code, the more complex it
is to implement and the price is often
paid in baud rate or required
bandwidth.

4.22
 Figure 4.4 Line coding schemes

4.23
 Terminology
■ Unipolar scheme
all the signal levels are on one side of the time axis, either
above or below.
■ Polar scheme

the voltages are on the both sides of the time axis. For
example, the voltage level for 0 can be positive and the
voltage level for 1 can be negative.
■ Bipolar scheme

there are three voltage levels: positive, negative, and zero.

The voltage level for one data element is at zero, while
the voltage level for the other element alternates
between positive and negative.

4.24
 Unipolar

■ All signal levels are on one side of the time

axis - either above or below
■ NRZ - Non Return to Zero scheme is an
example of this code. The signal level does
not return to zero during a symbol
transmission.
■ Scheme is prone to baseline wandering and
DC components. It has no synchronization or
any error detection. It is simple but costly in
power consumption.

4.25
 Figure 4.5 Unipolar NRZ scheme

4.26
 Figure 4.6 Polar NRZ-L and NRZ-I schemes

4.27
 Note

In NRZ-L the level of the voltage

determines the value of the bit.
In NRZ-I the inversion
or the lack of inversion
determines the value of the bit.

4.28
 Figure 4.7 Polar RZ scheme

4.29
 Figure 4.8 Polar biphase: Manchester and differential Manchester schemes

4.30
 Note

In Manchester and differential

Manchester encoding, the transition
at the middle of the bit is used for
synchronization.

4.31
 Example 4.4

A system is using NRZ-I to transfer 1-Mbps data. What

are the average signal rate and minimum bandwidth?

Solution
The average signal rate is S= c x N x R = 1/2 x N x 1 =
500 kbaud. The minimum bandwidth for this average
baud rate is Bmin = S = 500 kHz.

Note c = 1/2 for the avg. case as worst case is 1 and best
case is 0

4.32
 Polar - Biphase: Manchester and
Differential Manchester
■ Manchester coding consists of combining the
NRZ-L and RZ schemes.
■ Every symbol has a level transition in the middle:
from high to low or low to high. Uses only two
voltage levels.
■ Differential Manchester coding consists of
combining the NRZ-I and RZ schemes.
■ Every symbol has a level transition in the middle.
But the level at the beginning of the symbol is
determined by the symbol value. One symbol
causes a level change the other does not.

4.33
 Note

In Manchester and differential

Manchester encoding, there is no DC
component as each bit has positive and
negative voltage contribution.

4.34
 Note

Drawback:Minimum bandwidth for

Manchester and differential Manchester
is 2 times that of NRZ.

4.35
 Terminology
■ Unipolar scheme
all the signal levels are on one side of the time axis, either
above or below.
■ Polar scheme

the voltages are on the both sides of the time axis. For
example, the voltage level for 0 can be positive and the
voltage level for I can be negative.
■ Bipolar scheme

there are three voltage levels: positive, negative, and zero.

The voltage level for one data element is at zero, while
the voltage level for the other element alternates
between positive and negative.

4.36
 Bipolar(Multilevel binary) Schemes

■ In bipolar encoding (sometimes called multilevel

binary), there are three voltage levels: positive,
negative, and zero. The voltage level for one data
element is at zero, while the voltage level for the
other element alternates between positive and
negative.

■ AMI : alternate mark inversion

■ Pseudoternary

4.37
 Note

In bipolar encoding, we use three levels: positive, zero, and

negative.

4.38
 AMI
■
A neutral zero voltage represents binary 0.
■
Binary 1s are represented by alternating positive
and negative voltages.
Pseudoternary
■
A variation of AMI encoding is called pseudoternary
in which the 1 bit is encoded as a zero voltage
■
0 bit is encoded as alternating positive and
negative voltages.
4.39
 Figure 4.9 Bipolar schemes: AMI and
pseudoternary

4.40
 Multilevel Schemes
■ The desire to increase the data speed or decrease
the required bandwidth has resulted in the creation
of many schemes. The goal is to increase the number
of bits per baud by encoding a pattern of m data
elements into a pattern of n signal elements.
■ We only have two types of data elements (0s and
1s), which means that a group of m data elements
can produce a combination of 2^m data patterns.
■ We can have different types of signal elements by
allowing different signal levels.

4.41
 Multilevel Schemes

■ If 2^m < L^n, data patterns occupy only a subset of

signal patterns.

■ If we have L different levels, then we can produce

L^n combinations of signal patterns.

■ If 2^m =L^n, then each data pattern is encoded into

one signal pattern.

■ Data encoding is not possible if 2^m > L^n because

some of the data patterns cannot be encoded.

4.42
 Multilevel: terminology
The code designers have classified these types of coding as
mBnL, where
■m is the length of the binary pattern, B means binary data

■ n is the length of the signal pattern, and L is the number

of levels in the signaling.

■A letter is often used in place of L: B (binary) for L =2, T

(ternary) for L =3, and Q (quaternary) for L =4.

■Note that the first two letters define the data pattern, and

the second two define the signal pattern.

4.43
 Note

In mBnL schemes, a pattern of m data elements is encoded

as a pattern of n signal elements in which 2^m <= L^n.

4.44
 2B1Q scheme
■ Two binary, one quaternary (2B1Q), uses data patterns
of size 2 and encodes the 2-bit patterns as one signal
element belonging to a four-level signal.
■ In this type of encoding m =2, n =1, and L =4
(quaternary), Gig 4.10
■ The average signal rate of 2B1Q is S =N/4. This means
that using 2B1Q, we can send data 2 times faster than
by using NRZ-L.
■ 2B1Q is used in DSL (Digital Subscriber Line) technology
to provide a high-speed connection to the Internet by
using subscriber telephone lines.
■ (Numerical ….)
4.45
 Figure 4.10 Multilevel: 2B1Q scheme

11 00 01

4.46
 8B6T scheme
■ Eight binary, six ternary (8B6T), used with 100BASE-4T
cable.
■ The idea is to encode a pattern of 8 bits as a pattern of
6 signal elements, where the signal has three levels
(ternary).
■ we can have 2^8 =256 different data patterns and 3^6
=478 different signal patterns.
■ There are 478 - 256 =222 redundant signal elements
that provide synchronization and error detection.

■ (Numerical ….)

4.47
 8B6T scheme
Figure 4.11 shows an example of three data patterns encoded as three
signal patterns.
■The three possible signal levels are represented as -,0, and +.

■The first 8-bit pattern 00010001 is encoded as the signal pattern

-0-0++ with weight 0;

■the second 8-bit pattern 010 10011 is encoded as - + - + + 0 with
weight +1.
■The third bit pattern should be encoded as + - - + 0 + with weight +1.

■To create DC balance, the sender inverts the actual signal. The receiver

can easily recognize that this is an inverted pattern because the weight
is -1. The pattern is inverted before decoding

4.48
 Figure 4.11 Multilevel: 8B6T scheme

*Along with data bits weights are also considered to generate the digital pulse

4.49
 Table 4.1 Summary of line coding schemes

4.50
 Block Coding
■ For a code to be capable of error detection, we need
to add redundancy, i.e., extra bits to the data bits.
Ex. Hamming distance
■ Synchronization also requires redundancy -
transitions are important in the signal flow and must
occur frequently.
■ Block coding is done in three steps: division,
substitution and combination.
■ It is distinguished from multilevel coding by use of
the slash - xB/yB.
■ The resulting bit stream prevents certain bit
combinations that when used with line encoding
would result in DC components or poor sync. quality.

4.51
 Note

Block coding is normally referred to as

mB/nB coding;
it replaces each m-bit group with an
n-bit group.

4.52
 Figure 4.14 Block coding concept

4.53
 Figure 4.15 Using block coding 4B/5B with NRZ-I line coding scheme

4.54
 Figure 4.16 Substitution in 4B/5B block coding

4.55
 Redundancy

Let’s assume
■A 4 bit data word can have 24
combinations.
■A 5 bit word can have 25=32
combinations.
■We therefore have 32 - 24 extra words.

■Some of the extra words are used for

control/signalling purposes.

4.56
 Figure 4.17 8B/10B block encoding

4.57
 More bits - better error detection

■ The 8B10B block code adds more

redundant bits and can thereby choose
code words that would prevent a long
run of a voltage level that would cause
DC components.

4.58
 Scrambling
■ Deliberate distortion or encoding of audio/video signals, or a data
stream, through an electronic device (scrambler) to prevent
unauthorized reception in 'plain' or 'readable' form. It is like
protective layer.

• DC- direct current

• DC components: After line coding, the signal may have zero
frequency component in the spectrum of the signal, which is
known as the direct-current (DC) component.DC component in a
signal is not desirable because the DC component does not pass
through some components of a communication system such as
a transformer.

4.59
 DC Component : Flat line found in spectrum which is close to the
time axis. It should be minimum or not available for good signal.

4.60
 Scrambling
■ The best code is one that does not increase
the bandwidth for synchronization and has no
DC components.
■ Scrambling is a technique used to create a
sequence of bits that has - self clocking, no
low frequencies, no wide bandwidth.
■ It is implemented at the same time as
encoding, the bit stream is created on the fly.
■ It replaces ‘unfriendly’ runs of bits with a
violation code that is easy to recognize and
removes the unfriendly c/c.

4.61
 Figure 4.18 AMI used with scrambling

4.62
 Figure 4.20 Different situations in HDB3 scrambling technique

4.63
 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. We describe two
techniques, pulse code modulation and delta
modulation.
Topics discussed in this section:
▪ Pulse Code Modulation (PCM)
▪ Delta Modulation (DM)

4.64
 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.

4.65
 Figure 4.21 Components of PCM encoder

4.66
 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
4.67
 Figure 4.22 Three different sampling methods for PCM

4.68
 69

PCM Sampling

Natural sampling Flat-top sampling

 Note

According to the Nyquist theorem, the

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

4.70
 Terminologies
■ Passband
A passband is the range of frequencies or wavelengths
that can pass through a filter.
■ Bandpass

It is the range of frequencies which are transmitted

through a bandpass filter.
■ A bandpass filter

is an electronic device or circuit that allows signals

between two specific frequencies to pass, but that
discriminates against signals at other frequencies. ... The
range of frequencies between f1 and f2is called the filter
passband.

4.71
 Signals
■ Bandpass signal is a signal containing a band of frequencies not adjacent
to zero frequency, such as a signal that comes out of a bandpass filter.
The bandwidth of the filter is simply the difference between the upper and
lower cutoff frequencies
In telecommunications and signal processing, baseband signals are transmitted
without modulation, that is, without any shift in the range of frequencies of
the signal.
Simply we can say original signal is passed as it is.
■ Low pass:
A baseband channel or lowpass channel (or system, or network) is a
communication channel that can transfer frequencies that are very near
zero.
■ High pass: high-pass filter (HPF) is an electronic filter that
passes signals with a frequency higher than a certain cutoff frequency and
attenuates signals with frequencies lower than the cutoff frequency. The
amount of attenuation for each frequency depends on the filter design

4.72
 Summary
To be exact, systems (filters) are low pass and bandpass, signals are
baseband and passband.
You can consider this signal as a low pass signal.
A low pass signal can be output of a low pass filter.
A baseband signal is centerd around DC (zero) frequency.

■ What is low pass and band pass?

■ A low-pass filter (LPF) is a filter that passes signals with a
frequency lower than a certain cutoff frequency and attenuates signals
with frequencies higher than the cutoff frequency. A low-pass filter is the
complement of a high-pass filter

4.73
 Figure 4.23 Nyquist sampling rate for low-pass and bandpass signals
Below Case1: Near to zero frequency and case2 far from zero frequency

4.74
 75

Sampling Rate
Sampling is a process of taking samples of information signal at a rate based
on the Nyquist Sampling Theorem.

Nyquist Sampling Theorem – the original information signal can be

reconstructed at the receiver with minimal distortion if the sampling rate
in the pulse modulation signal is equal or greater than twice the
maximum information signal frequency.
 76

Sampling Rate
n If fs is less than 2 times fm(max) an impairment called as alias or fold-over distortion occurs.
 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.
4.77
 Figure 4.24 Recovery of a sampled sine wave for different sampling rates

4.78
 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.
4.79
 Figure 4.25 Sampling of a clock with only one hand

4.80
 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.

4.81
 Example 4.9

Telephone companies digitize voice by assuming a

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

4.82
 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.

4.83
 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.

4.84
 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
4.85
 86

Quantization
Quantization – process of assigning the analog signal samples to a pre-determined
discrete level.

The number of quantization levels, L depends on the number of bits per sample, n
where

where L = number of quantization level

n = number of bits in binary to represent the value of the samples

The quantization levels are separated by a value of ΔV that can be defined as

 87

5.3.3 Quantization
n Ex :
Vmax= 3.5 Vmin=-3.5 No of
levels L=8
DeltaV= 3.5-(-3.5)/(8-1)= 1 V
 88

5.3.3 Quantization
n Ex (continue) :
 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.

4.89
 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:

4.90
 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.

4.91
 Figure 4.27 Components of a PCM decoder

4.92
 Example 4.15

We have a low-pass analog signal of 4 kHz. If we send the

analog signal, we need a channel with a minimum
bandwidth of 4 kHz. If we digitize the signal and send 8
bits per sample, we need a channel with a minimum
bandwidth of 8 × 4 kHz = 32 kHz.

4.93
 Delta Modulation
■ 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.

4.94
 Figure 4.28 The process of delta modulation

4.95
 Figure 4.29 Delta modulation components

4.96
 Figure 4.30 Delta demodulation components

4.97
 4-3 TRANSMISSION MODES

The transmission of binary data across a link can be

accomplished in either parallel or serial mode. In
parallel mode, multiple bits are sent with each clock
tick. In serial mode, 1 bit is sent with each clock tick.
While there is only one way to send parallel data, there
are three subclasses of serial transmission:
asynchronous, synchronous, and isochronous.

Topics discussed in this section:

▪ Parallel Transmission
▪ Serial Transmission
4.98
 Figure 4.31 Data transmission and modes

4.99
 Figure 4.32 Parallel transmission

4.100
 Figure 4.33 Serial transmission

4.101
 Note

In asynchronous transmission, we send

1 start bit (0) at the beginning and 1 or
more stop bits (1s) at the end of each
byte. There may be a gap between
each byte.

4.102
 Note

Asynchronous here means

“asynchronous at the byte level,”
but the bits are still synchronized;
their durations are the same.

4.103
 Figure 4.34 Asynchronous transmission

4.104
 Note

In synchronous transmission, we send

bits one after another without start or
stop bits or gaps. It is the responsibility
of the receiver to group the bits. The bits
are usually sent as bytes and many
bytes are grouped in a frame. A frame is
identified with a start and an end byte.

4.105
 Figure 4.35 Synchronous transmission

4.106
 Isochronous

■ In isochronous transmission we cannot

have uneven gaps between frames.
■ Transmission of bits is fixed with equal
gaps.

4.107
4-4 ANALOG-TO-ANALOG CONVERSION

?
 Amplitude Modulation

■ A carrier signal is modulated only in

amplitude value
■ The modulating signal is the envelope of the
carrier
■ The required bandwidth is 2B, where B is the
bandwidth of the modulating signal
■ Since on both sides of the carrier freq. fc, the
spectrum is identical, we can discard one half,
thus requiring a smaller bandwidth for
transmission.

5.109
 Figure 5.16 Amplitude modulation

5.110
 Note

The total bandwidth required for AM

can be determined
from the bandwidth of the audio
signal: BAM = 2B.

5.111
 Figure 5.17 AM band allocation

5.112
 Frequency Modulation

■ The modulating signal changes the freq.

fc of the carrier signal
■ The bandwidth for FM is high
■ It is approx. 10x the signal frequency

5.113
 Note

The total bandwidth required for FM can be determined from the

bandwidth
of the audio signal: BFM = 2(1 + β)B. Where β is usually 4.

5.114
 Figure 5.18 Frequency modulation

5.115
 Figure 5.19 FM band allocation

5.116
 Phase Modulation (PM)

■ The modulating signal only changes the

phase of the carrier signal.
■ The phase change manifests itself as a
frequency change but the instantaneous
frequency change is proportional to the
derivative of the amplitude.
■ The bandwidth is higher than for AM.

5.117
 Figure 5.20 Phase modulation

5.118
 Note

The total bandwidth required for PM can be determined from the

bandwidth
and maximum amplitude of the modulating signal:
BPM = 2(1 + β)B.
Where β = 2 most often.

5.119
 Bandwidth Utilization:
Multiplexing and Spreading

6.120 Copyright © The McGraw-Hill Companies, Inc. Permission required for reproduction or display.
 Note

Bandwidth utilization is the wise use of

available bandwidth to achieve
specific goals.

Efficiency can be achieved by multiplexing; i.e., sharing of the

bandwidth between multiple users.

6.121
 6-1 MULTIPLEXING

Whenever the bandwidth of a medium linking two devices is greater than the
bandwidth needs of the devices, the link can be shared.
Multiplexing is the set of techniques that allows the (simultaneous) transmission of
multiple signals across a single data link.
As data and telecommunications use increases, so does traffic.

Topics :
❑ Frequency-Division Multiplexing
❑ Wavelength-Division Multiplexing
❑ Synchronous Time-Division Multiplexing
❑ Statistical Time-Division Multiplexing
6.122
 Figure 6.1 Dividing a link into channels

6.123
 Figure 6.2 Categories of multiplexing

6.124
 Figure 6.3 Frequency-division multiplexing (FDM)

6.125
 Note

FDM is an analog multiplexing technique that combines analog

signals.
It uses the concept of modulation.

6.126
 Figure 6.4 FDM process

6.127
 FM

6.128
 Figure 6.5 FDM demultiplexing example

6.129
 Example 6.1

Assume that a voice channel occupies a bandwidth of 4 kHz. We need to combine three
voice channels into a link with a bandwidth of 12 kHz, from 20 to 32 kHz. Show the
configuration, using the frequency domain. Assume there are no guard bands. ( Not for
Exam)

Solution
We shift (modulate) each of the three voice channels to a different bandwidth, as shown
in Figure 6.6. We use the 20- to 24-kHz bandwidth for the first channel, the 24- to 28-kHz
bandwidth for the second channel, and the 28- to 32-kHz bandwidth for the third one.
Then we combine them as shown in Figure 6.6.

6.130
 Figure 6.6 Example 6.1

6.131
 Example 6.2

Five channels, each with a 100-kHz bandwidth, are to be multiplexed together. What is
the minimum bandwidth of the link if there is a need for a guard band of 10 kHz between
the channels to prevent interference?

Solution
For five channels, we need at least four guard bands. This means that the required
bandwidth is at least
5 × 100 + 4 × 10 = 540 kHz,
as shown in Figure 6.7.

6.132
 Figure 6.7 Example 6.2

6.133
 Figure 6.9 Analog hierarchy

6.134
 Wavelength
■ Definition:
The distance between successive crests of a wave,
especially points in a sound wave or electromagnetic
wave.

■ Mostly term used for electromagnetic waves passed

through fiber optic cables.

6.135
 Figure 6.10 Wavelength-division multiplexing (WDM)

6.136
 Note

WDM is an analog multiplexing technique to combine optical

signals.

6.137
 Figure 6.11 Prisms in wavelength-division multiplexing and demultiplexing

6.138
 Figure 6.12 Time Division Multiplexing (TDM)

6.139
 Note

TDM is a digital multiplexing technique for combining several

low-rate digital
channels into one high-rate one.

6.140
 Figure 6.13 Synchronous time-division multiplexing

6.141
 Note

In synchronous TDM, the data rate

of the link is n times faster, and the unit duration is n times
shorter.

6.142
 Figure 6.13 Synchronous time-division multiplexing

6.143
 Data Rate Management

■ Not all input links maybe have the same

data rate.
■ Some links maybe slower. There maybe
several different input link speeds
■ There are three strategies that can be
used to overcome the data rate
mismatch: multilevel, multislot and
pulse stuffing

6.144
 Data rate matching

■ Multilevel: used when the data rate of the

input links are multiples of each other.
■ Multislot: used when there is a GCD between
the data rates. The higher bit rate channels
are allocated more slots per frame, and the
output frame rate is a multiple of each input
link.
■ Pulse Stuffing: used when there is no GCD
between the links. The slowest speed link will
be brought up to the speed of the other links
by bit insertion, this is called pulse stuffing.
6.145
 Figure 6.19 Multilevel multiplexing

6.146
 Figure 6.20 Multiple-slot multiplexing

6.147
 Figure 6.21 Pulse stuffing

6.148
 Synchronization
■ To ensure that the receiver correctly reads
the incoming bits, i.e., knows the incoming bit
boundaries to interpret a “1” and a “0”, a
known bit pattern is used between the
frames.
■ The receiver looks for the anticipated bit and
starts counting bits till the end of the frame.
■ Then it starts over again with the reception of
another known bit.
■ These bits (or bit patterns) are called
synchronization bit(s).
■ They are part of the overhead of
transmission.
6.149
 Figure 6.22 Framing bits

6.150
 Figure 6.23 Digital hierarchy

6.151
 Table 6.1 DS and T line rates

6.152
 Inefficient use of Bandwidth

■ Sometimes an input link may have no

data to transmit.
■ When that happens, one or more slots
on the output link will go unused.
■ That is wasteful of bandwidth.

6.153
 Figure 6.18 Empty slots

6.154
 Figure 6.26 TDM slot comparison

6.155
 SPREAD SPECTRUM

In spread spectrum (SS), we combine signals from different sources to fit into a
larger bandwidth, but our goals are to prevent eavesdropping and jamming.

To achieve these goals, spread spectrum techniques add redundancy.

Topics discussed in this section:

▪ Frequency Hopping Spread Spectrum (FHSS)
▪ Direct Sequence Spread Spectrum (DSSS)

6.156
 Terminilogies

■ Jamming Signal
A signal that intentionally introduces interference
into a communication channel, either to intentionally
prevent error-free reception or as a means of advising
stations of some event.

Spying ::

6.157
 Spread Spectrum

■ A signal that occupies a bandwidth of B, is

spread out to occupy a bandwidth of Bss
■ All signals are spread to occupy the same
bandwidth Bss
■ Signals are spread with different codes so
that they can be separated at the receivers.
■ Signals can be spread in the frequency
domain or in the time domain.

6.158
 Figure 6.27 Spread spectrum

6.159
 Figure 6.28 Frequency hopping spread spectrum (FHSS)

6.160
 Figure 6.29 Frequency selection in FHSS

6.161
 Figure 6.30 FHSS cycles

6.162
 Figure 6.31 Bandwidth sharing

6.163
 Figure 6.32 DSSS

6.164
 Figure 6.33 DSSS example

6.165
 REFERENCEs

1. Fourauzan B., "Data Communications and Networking",

5th edition, McGraw-Hill Publications
2. Stallings William., "Data and Computer Communications",
Sixth Edition, Prentice Hall of India

4.166