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# Multiplexing

## Multiplexing is the name given to techniques, which allow more

than one message to be transferred via the same
communication channel. The channel in this context could be a
transmission line, e.g. a twisted pair or co-axial cable, a radio
system or a fibre optic system etc.

A channel

## will offer a specified bandwidth, which is available for

a time t, where t may . Thus, with reference to the channel
there are 2 degrees of freedom, i.e. bandwidth or frequency
and time.
1

Multiplexing
CHANNEL
BL

BH

freq

BH

## Multiplexing is a technique which allows k

users to occupy the
channel for the duration in time that the
channel is available.
BL
Frequency

Time t

## Now consider a signal

v s (t ) Amp cos(t )

## The signal is characterised by amplitude, frequency, phase and time.

Multiplexing
Various multiplexing methods are possible in terms of the channel bandwidth and time,
and the signal, in particular the frequency, phase or time. The two basic methods are:
1) Frequency Division Multiplexing FDM
FDM is derived from AM techniques in which the signals occupy the same physical
line but in different frequency bands. Each signal occupies its own specific band of
frequencies all the time, i.e. the messages share the channel bandwidth.
2) Time Division Multiplexing TDM
TDM is derived from sampling techniques in which messages occupy all the channel
bandwidth but for short time intervals of time, i.e. the messages share the channel time.
FDM messages occupy narrow bandwidth all the time.
TDM messages occupy wide bandwidth for short intervals of time.

Multiplexing
These two basic methods are illustrated below.
time
time
M1
BL

M2
B

M3

BL
M4

M1

M5

M4 M5
M2 M3
t

BH

BH

freq

freq
t
BL

BH
M1
M2

M1

M3

M2

M3

M4

M5

M4
BH

M5

BL

FDM

TDM

## Frequency Division Multiplexing FDM

FDM is widely used in radio and television systems (e.g.
multichannel telephony (now being superseded by digital
techniques and TDM).
The multichannel telephone system illustrates some
important aspects and is considered below. For speech,
a bandwidth of 3kHz is satisfactory.
The physical line, e.g. a co-axial cable will have a
bandwidth compared to speech as shown next
5

## Frequency Division Multiplexing FDM

3kHz

freq

GHz
From AM we have noted:
m(t)

m(t)

freq

DSBSC

carrier
cos( c t )

DSBSC
freq

fc

## Frequency Division Multiplexing FDM

In order to use bandwidth more effectively, SSB is used i.e.
SSB
Filter

m(t)

SSBSC

carrier
cos( c t )

freq

fc

We have also noted that the message signal m(t) is usually band limited, i.e.

Speech

Band
Limiting
Filter
300Hz 3400Hz

m(t)

SSB
Filter

cos( c t )

SSBSC

## Frequency Division Multiplexing FDM

The Band Limiting Filter (BLF) is usually a band pass filter with a pass band 300Hz to
3400Hz for speech. This is to allow guard bands between adjacent channels.

f
10kHz

Speech

f
300Hz

3400Hz

m(t)

f
300Hz

3400Hz

Convention

## Frequency Division Multiplexing FDM

For telephony, the physical line is divided (notionally) into 4kHz bands or channels, i.e.
the channel spacing is 4kHz. Thus we now have:

Guard Bands

Bandlimited
Speech

4kHz
Note, the BLF does not have an ideal cut-off the guard bands allow for filter roll off
in order to reduce adjacent channel crosstalk.
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## Frequency Division Multiplexing FDM

Consider now a single channel SSB system.
The spectra will be

m(t)

DSBSC
BLF

SSB
Filter

SSBSC

fc

m(t)
freq

300Hz

3400Hz

DSBSC
freq
fc

freq
fc

10

## Frequency Division Multiplexing FDM

Consider now a system with 3 channels
m1(t)
f

SSB
Filter

BLF
fc1

m2(t)

SSB
Filter

BLF
f

fc2

FDM
Signal
M(t)

f2

SSB
Filter

BLF

m3(t)

f1

fc3

f3

Bandlimited

FDM Transmitter
or Encoder

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## Frequency Division Multiplexing FDM

Each carrier frequency, fc1, fc2 and fc3 are separated by the channel spacing
frequency, in this case 4 kHz, i.e. fc2 = fc1 + 4kHz, fc3 = fc2 + 4kHz.
The spectrum of the FDM signal, M(t) will be:

4kHz

4kHz

M(t)

4kHz
show guard bands.

f1
fc1

f3

f2
fc2

fc3

freq

12

## Frequency Division Multiplexing FDM

Note that the baseband signals m1(t), m2(t), m3(t) have been multiplexed into adjacent
channels, the channel spacing is 4kHz. Note also that the SSB filters are set to select
the USB, tuned to f1, f2 and f3 respectively. A receiver FDM decoder is illustrated below:
SSB
Filter
f1

M(t)
FDM
Signal

LPF
fc1

SSB
Filter
f2

Band
Limited
LPF

m2(t)
Back to
baseband

fc2

SSB
Filter
f3

m1(t)

LPF

fc3

m3(t)

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## Frequency Division Multiplexing FDM

The SSB filters are the same as in the encoder, i.e. each one
centred on f1, f2 and f3 to select the appropriate sideband and reject
the others. These are then followed by a synchronous demodulator,
each fed with a synchronous LO, fc1, fc2 and fc3 respectively.

For the 3 channel system shown there is 1 design for the BLF (used
3 times), 3 designs for the SSB filters (each used twice) and 1
design for the LPF (used 3 times).

## A co-axial cable could accommodate several thousand 4 kHz

channels, for example 3600 channels is typical. The bandwidth used
is thus 3600 x 4kHz = 14.4Mhz. Potentially therefore there are 3600
different SSB filter designs. Not only this, but the designs must
range from kHz to MHz.
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## Frequency Division Multiplexing FDM

For designs around say 60kHz, Q

60kHz
= 15 which is reasonable.
4 kHz

## However, for designs to have a centre frequency at around say 10Mhz,

10,000kHz
gives a Q = 2500 which is difficult to achieve.
Q
4 kHz

To overcome these problems, a hierarchical system for telephony used the FDM
principle to form groups, supergroups, master groups and supermaster groups.

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## Basic 12 Channel Group

The diagram below illustrates the FDM principle for 12 channels (similar to 3 channels)
to a form a basic group.
m1(t)
m2(t)
m3(t)

Multiplexer
freq

m12(t)

12kHz

60kHz

i.e. 12 telephone channels are multiplexed in the frequency band 12kHz 60 kHz in
4kHz channels basic group.
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## Basic 12 Channel Group

A design for a basic 12 channel group is shown below:
Band Limiting Filters
DSBSC

4kHz

CH1
m1(t)

8.6 15.4kHz

300Hz

3400kHz

SSB Filter
12.3 15.4kHz

f1 = 12kHz

4kHz

12.6 19.4kHz

CH2
m2(t)
300Hz

16.3 19.4kHz

3400kHz

f1 = 16kHz

## Increase in 4kHz steps

FDM OUT
12 60kHz

4kHz

52.6 59.4kHz

CH12
m12(t)
300Hz

56.3 59.4kHz

3400kHz

f12 = 56kHz

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Super Group
These basic groups may now be multiplexed to form a super group.
12
Inputs

BASIC
GROUP
12 60kHz

SSB
FILTER
420kHz

12
Inputs

BASIC
GROUP
12 60kHz

SSB
FILTER
468kHz

12
Inputs

BASIC
GROUP
12 60kHz

SSB
FILTER

516kHz

12
Inputs

BASIC
GROUP
12 60kHz

SSB
FILTER
564kHz

12
Inputs

BASIC
GROUP
12 60kHz

SSB
FILTER
612kHz

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Super Group
5 basic groups multiplexed to form a super group, i.e. 60 channels in one super group.
Note the channel spacing in the super group in the above is 48kHz, i.e. each carrier
frequency is separated by 48kHz. There are 12 designs (low frequency) for one basic
group and 5 designs for the super group.

612 kHz
12 - which is reasonable
The Q for the super group SSB filters is Q
48kHz
Hence, a total of 17 designs are required for 60 channels. In a similar way, super groups
may be multiplexed to form a master group, and master groups to form super master
groups

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## Time Division Multiplexing TDM

TDM is widely used in digital communications, for example in the form of pulse code
modulation in digital telephony (TDM/PCM). In TDM, each message signal occupies
the channel (e.g. a transmission line) for a short period of time. The principle is
illustrated below:
1

m1(t)
2

m2(t)
m3(t)
m4(t)
m5(t)

m1(t)
2

Tx

4
5

Rx
SW2

SW1

Transmission
Line

m2(t)

4
5

m3(t)
m4(t)
m5(t)

Switches SW1 and SW2 rotate in synchronism, and in effect sample each message
input in a sequence m1(t), m2(t), m3(t), m4(t), m5(t), m1(t), m2(t),
The sampled value (usually in digital form) is transmitted and recovered at the far end
to produce output m1(t)m5(t).
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## Time Division Multiplexing TDM

For ease of illustration consider such a system with 3 messages, m1(t), m2(t) and m3(t),
each a different DC level as shown below.

m1(t)

V1
t

0
m2(t)

V2
0

m3(t)

V3
0

SW1
Sample
t
Position

21

V3
V2
V1
t
m1(t)

m2(t)

m3(t)

m1(t)

m2(t)

m3(t)

m1(t)

Channel
Time
Slots

1
t

Time slot

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## Time Division Multiplexing TDM

In this illustration the samples are shown as levels, i.e. V1, V2 or V3.
Normally, these voltages would be converted to a binary code before
transmission as discussed below.

Note that the channel is divided into time slots and in this example, 3
messages are time-division multiplexed on to the channel. The sampling
process requires that the message signals are a sampled at a rate fs 2B,
where fs is the sample rate, samples per second, and B is the maximum
frequency in the message signal, m(t) (i.e. Sampling Theorem applies). This
sampling process effectively produces a pulse train, which requires a
bandwidth much greater than B.

Thus in TDM, the message signals occupy a wide bandwidth for short
intervals of time. In the illustration above, the signals are shown as PAM
(Pulse Amplitude Modulation) signals. In practice these are normally
converted to digital signals before time division multiplexing.
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## Time Division Multiplexing TDM

A schematic diagram to illustrate the principle for 3 message signals is shown below.
m1(t)

S/H

BLF

PAM
1

fs1
m2(t)

BLF

S/H

PAM
2

S/H

PAM
3

fs2
m3(t)

BLF

Multiplexing
Analogue
To
Digital
Convertor

Serial output
Binary digital
data d(t)

fs3
Band limiting
Filter 0 B Hz

## Sample and Hold

Sample rate fs
fs 2B Hz

Converts each input
in turn to an n bit code.

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25

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## Time Division Multiplexing TDM

Each sample value is converted to an n bit code by the ADC. Each n bit code fits into
the time slot for that particular message. In practice, the sample pulses for each
message input could be the same. The multiplexing ADC could pick each input
(i.e. a S/H signal) in turn for conversion.

For an N channel system, i.e. N message signals, sampled at a rate fs samples per
second, with each sample converted to an n bit binary code, and assuming no
additional bits for synchronisation are required (in practice further bits are required) it is
easy to see that the output bit rate for the digital data sequence d(t) is

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## School of Electrical, Electronics and

Computer Engineering
University of Newcastle-upon-Tyne

## Baseband digital Modulation

Prof. Rolando Carrasco

Lecture Notes
University of Newcastle-upon-Tyne
2005

## Bit-rate, Baud-rate and

Bandwidth
B denotes the duration of the 1 bit
Hence Bit rate =

## bits per second

All the forms of the base band signalling shown transfer data at the same bit rate.

## denotes the duration of the shortest signalling element.

Baud rate is defined as the reciprocal of the duration of the shortest signalling element .
1
Baud Rate =
baud

In general
For

NRZ :
RZ :
Bi-Phase:
AMI:

## Baud Rate = Bit Rate

Baud Rate = 2 x Bit Rate
Baud Rate = 2 x Bit Rate
Baud Rate = Bit Rate

i.e.

## This sequence produces a square wave with periodic time 2 E

Fourier series for a square wave,

If we pass this signal through a LPF then the maximum bandwidth would be 1/T
Hz, i.e. to just allow the fundamental (1st harmonic) to pass.

(Contd)
The data sequence 1010
could then be completely
recovered

Bmin

1
1 Baud Rate
1

## Since Baud Rate

T 2 E
2
E

Considering RZ signals, the max frequency occurs when continuous 1s are transmitted.

## This produces a square wave with periodic time

Bmin

2 E

Baud Rate
fU
2

If the sequence was continuous 0s, the signal would be V continuously, hence

f L ' DC '

Bi-Phase
Maximum frequency occurs when continuous
1s or 0s transmitted.
This is similar to RZ with
Baud Rate =

= 2 x Bit rate

Baud Rate
2
The minimum frequency occurs when the sequence is 10101010.
e.g.
Bmin f U

In this case

B = E

Bmin f L

Baud Rate
2

## Digital Modulation and

Noise
The performance of Digital Data Systems is dependent on the bit error rate, BER, i.e.
probability of a bit being in error.
Prob. of Error or BER,
No of Errors E
P
as N
Total bits N
Digital Modulation
There are four basic ways of sending
digital data
The BER (P) depends on several factors
the modulation type, ASK FSK or
PSK
the demodulation method
the noise in the system
the signal to noise ratio

Noise

Noise

Noise

Analysis

## DEMODULATOR DETECTOR DECISION

Demodulator-Detector-Decision
FOR FSK

Demodulator

Demodulator Contd)

1
V IN dt

RC
Hence design RCT

Vout

Detector-Decision
V1 - V0 is the voltage difference
between a 1 and 0.

(V REF

V1 V2

)
2
2

Detector-Decision (Contd)
ND is the noise at the Detector input.
Probability of Error,

1 erf
2
2 2 N D
Hence

v0

v1

P(v0)

vn

P0 (vn )

1 1
e
2 2

2 ND

( v0 v1 ) 2
2 2

P1 (vn )

Pe1

vn

v1

v0

v0 v1
2

2
x

( v n v0 ) 2
2 2

v n v0
2

dv n

(*)

Pe1

This becomes

x 2 dx

(**)

v1 v0

2 2

## The incomplete integral cannot be evaluated analytically but can be recast as a

complimentary error function, erfc(x), defined by

erfc ( z )
Equations (*) and (**) become

1
v1 v0
Pe1 erfc

2
2 2

e
z

x2

dx
erfc( z ) 1 erf ( z )
Pe1

Pe 0

1
v1 v0
1

erf

2
2

v0 v1
2

1
e
2

( vn v1 ) 2
2 2

dvn

It is clear from the symmetry of this problem that Pe0 is identical to Pe1 and the
probability of error Pe, irrespective of whether a one or zero was transmitted, can
be rewritten in terms of v = v1 v0

1
v
Pe 1 erf

2
2 2
for unipolar signalling (0 and v)

v
for polar signalling (symbol represented by voltage
2

Detector-Decision (Contd)

1
e 1 erf
2

S IN
4 N IN

OOK

FSK

PSK
PRK

1
e 1 erf
2
1

e 1 erf
2

S IN
2 N IN

## For Optimum ASK , FSK , PSK

S IN
N IN

Detector-Decision (Contd)

## FM/ FSK Demodulation

One form of FM/FSK demodulator is shown below

## In general VIN (t) will be

VIN (t ) Vc Cos IN t
Where IN is the input frequency (rad/sec) IN 2 f IN
V x V IN t V IN t
V x Vc Cos IN t .Vc Cos IN (t )
Since CosA CosB

1
Cos A B Cos A B
2

Vc2
Vx
Cos IN t IN t Cos IN t IN t
2

## FM/ FSK Demodulation (Contd)

i.e

Vc2
Cos IN t IN IN t Cos IN t IN IN t
Vx
2
Vc2
Cos 2 IN IN t Cos IN
Vx
2

Vc2
Cos 2 IN t
(1)
2
2
Vc2
and
Cos IN t
( 2)
2

## Component (1) is at frequency 2 fIN Hz and component (2) is effectively a DC voltage if

IN is constant.

The cut-off frequency for the LPF is designed so that component (1) is removed and
component (2) is passed to the output.
VOUT

Vc2

Cos IN t
2

## FM/ FSK Demodulation (Contd)

The V/F characteristics and inputs are shown below
Analogue FM

f c Vm
ym xc
f out VIN f 0
VIN VDC m(t )
VIN VDC Vm Cos mt
i.e. f out VDC Vm Cos mt f 0
f c VDC ,

Tc

Modulation Index

1
fc

f c Vm

fm
fm

## FM/ FSK Demodulation (Contd)

The spectrum of the analogue FM signal depends on

and is given by

FM Vs (t ) Vc J n ( ) Cos c n m t
n 1

Digital FSK
ym xc
f out V IN f 0
V IN V DC m(t )
V IN V DC V1

for 1' s

V IN V DC V0

for 0' s

f 1 V DC V1 f 0

for 1' s

f 0 V DC V0 f 0

for 0' s

f c V DC ,

Tc

1
fc

## Normalized frequency Deviation ratio

h

f1 f 0
Rb

i.e. Modulus f1 f 0

## FM/ FSK Demodulation (Contd)

Consider again the output from the demodulator VOUT
The delay
Hence

VOUT

is set to Tc
4

where Tc

2 f IN
Vc2

Cos
2
4 fc

1
fc

VOUT

and

Vc2

Cos IN
2

fc

f IN
Vc2

Cos
2
2 fc

## FM/ FSK Demodulation (Contd)

The curve shows the demodulator F/V characteristics which in this case is non linear.

## Practical realization of F/V process

The comparator is LIMITER which is a zero crossing detector to give a digital input to
the first gate.
This is form of delay and multiply circuit where the delay

= CR

Consider now

f IN f c

VOUT

AE f IN

4 fc

f IN

(Assuming A=1)