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7.0
Multiplexing
In a communication system, the costliest element is the transmission medium. To make the
best use of the medium, we have to ensure that the bandwidth of the channel is utilized to its
fullest capacity.
Multiplexing is the technique used to combine a number of channels and send them over the
medium to make the best use of the transmission medium. We will discuss the various
multiplexing techniques in this lesson.
7.1
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 fiber 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 two degrees of freedom, i.e. bandwidth
(frequency) and time.
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Use of multiplexing technique is possible if the capacity of the channel is higher than the data
rates of the individual data sources. Consider the example of a communication system in
which there are three data sources.
As shown in Figure 7.1, the signals from these three sources can be combined together
(multiplexed) and sent through a single transmission channel. At the receiving end, the
signals are separated (de-multiplexed).
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7.2
One of FDM's most common applications is cable television. Only one cable reaches a
customer's home but the service provider can send multiple television channels or signals
simultaneously over that cable to all subscribers. Receivers must tune to the appropriate
frequency (channel) to access the desired signal.
FDM is widely used in radio and television systems (e.g. broadcast radio and TV) and was
widely used in multichannel telephony. However, it is prone to noise problems, and has been
overtaken by Time Division Multiplexing which is better suited for digital data.
The multichannel telephone system illustrates some important aspects and is considered
below. For speech, a bandwidth of 3 kHz is satisfactory. The physical line, e.g. a co-axial
cable will have a bandwidth compared to speech as shown below.
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where
Carrier frequency = fc
DSB-SC Double Sideband Suppressed Carrier
In order to use bandwidth more effectively, Single Sideband (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
m(t)
SSB
Filter
SSBSC
300Hz 3400Hz
cos( c t )
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.
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f
300Hz
3400Hz
f
300Hz
3400Hz
10kHz
Convention
m(t)
Speech
For telephony, the physical line is divided (notionally) into 4 kHz bands or channels, i.e. the
channel spacing is 4 kHz. Thus we now have:
Guard Bands
Bandlimited
Speech
f
4kHz
Note, the BLF does not have an ideal cut-off the guard bands allow for filter roll off inorder to reduce adjacent channel crosstalk.
DSBSC
m(t)
BLF
fc
300Hz
3400Hz
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SSB
Filter
SSBSC
m(t)
freq
DSBSC
freq
fc
freq
fc
m1(t)
f
SSB
Filter
BLF
fc1
m2(t)
f1
SSB
Filter
BLF
FDM
Signal
fc2
M(t)
f2
SSB
Filter
BLF
m3(t)
fc3
f3
FDM Transmitter
or Encoder
Bandlimited
Each carrier frequency, fc1, fc2 and fc3 are separated by the channel spacing frequency, in this
case 4 kHz, i.e
fc2 = fc1 + 4 kHz,
fc3 = fc2 + 4 kHz.
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4kHz
4kHz
M(t)
4kHz
Shaded areas are to
show guard bands.
f1
fc1
f3
f2
fc2
freq
fc3
Note that:
The baseband signals m1(t), m2(t), m3(t) have been multiplexed into adjacent channels,
the channel spacing is 4 kHz.
The SSB filters are set to select the USB, tuned to f1, f2 and f3 respectively.
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
m3(t)
fc3
The SSB filters are the same as in the encoder, i.e. each one centered on f1, f2 and f3 to select
the appropriate sideband and reject the others.
These are then followed by a synchronous demodulator, where each fed with a synchronous
local oscillator (LO), fc1, fc2 and fc3 respectively.
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For the 3 channel system (Transmitter and Receiver) shown, there is:
1 design for the BLF (used 3 times)
3 designs for the SSB filters (each used twice)
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 4 kHz = 14.4 MHz. Therefore there
are 3600 different SSB filter designs. Not only this, but the designs must range from kHz to
MHz.
Consider also the Q of the filter, where Q is defined as Q =
centre frequency
.
bandwidth
60kHz
= 15 which is reasonable. However, for designs
4 kHz
10,000kHz
gives a Q = 2500 which is
to have a centre frequency at around say 10 MHz, Q =
4 kHz
difficult to achieve.
For designs around say 60 kHz, Q =
To overcome these problems, a hierarchical system for telephony used the FDM principle to
form groups, super-groups, master-groups and super-master groups.
Channel Grouping
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 4 kHz
channels basic group. A design for a basic 12 channel group is shown below:
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SSB Filter
DSBSC
4kHz
CH1
m1(t)
8.6 15.4kHz
300Hz
12.3 15.4kHz
3400kHz
f1 = 12kHz
4kHz
16.3 19.4kHz
12.6 19.4kHz
CH2
m2(t)
300Hz
3400kHz
f1 = 16kHz
4kHz
52.6 59.4kHz
CH12
m12(t)
300Hz
56.3 59.4kHz
3400kHz
f12 = 56kHz
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
5 basic groups multiplexed to form a super group, i.e. 60 channels in one super group.
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Note the channel spacing in the super group in the above is 48 kHz, i.e. each carrier
frequency is separated by 48 kHz. There are 12 designs (low frequency) for one basic group
and 5 designs for the super group.
612 kHz
12 - which is reasonable. Hence, a
48kHz
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
7.3
TDM is a digital technology derived from sampling techniques. It is widely used in digital
communications. In TDM, messages occupy all the channel bandwidth but for short time
intervals of time, i.e. the messages share the channel time.
In comparison with FDM,
FDM messages occupy narrow bandwidth all the time.
TDM messages occupy wide bandwidth for short intervals of time.
TDM involves sequencing groups of a few bits or bytes from each individual input stream,
one after the other, and in such a way that they can be associated with the appropriate
receiver. If done sufficiently and quickly, the receiving devices will not detect that some of
the circuit time was used to serve another logical communication path.
In TDM, each message signal occupies the channel (e.g. a transmission line) for a short
period of time. The principle is illustrated as follow:
1
1
m1(t)
2
m2(t)
m3(t)
m4(t)
m5(t)
m1(t)
2
Tx
4
Rx
SW2
SW1
5
Transmission
Line
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m2(t)
3
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). For ease of illustration consider such a system with 3 messages,
m1(t), m2(t) and m3(t), each at different DC level as shown below.
m1(t)
V1
t
0
m2(t)
V2
0
m3(t)
V3
0
SW1
Sample
t
Position
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
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 timedivision multiplexed on to the channel.
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The sampling process requires that the message signals are 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.
This process is illustrated below.
A schematic diagram to illustrate the principle for 3 message signals is shown below.
m1(t)
S/H
BLF
PAM
1
Multiplexing
Analogue
To
Digital
Convertor
fs1
m2(t)
S/H
BLF
PAM
2
Serial output
Binary digital
data d(t)
fs2
m3(t)
S/H
BLF
PAM
3
fs3
Band limiting
Filter 0 B Hz
Multiplexing ADC
Converts each input
in turn to an n bit code.
Again for simplicity, each message input is assumed to be a DC level. (see next page)
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
synchronization 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
Output bit rate = Nnfs bits/second.
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m1(t) V1
t
m2(t) V2
t
m3(t) V3
t
fs1
Sample
pulses
fs2
fs3
PAM 1
S/H
PAM 2
S/H
PAM 3
S/H
001
011
110
001
011
110
d(t)
e.g. n = 3 bits
m1(t) m2(t) m3(t) m1(t) m2(t) m3(t)
7.4
Page 15 of 21
Theoretically, the full data transmission capacity of a fiber could be exploited with a single
data channel of very high data rate, corresponding to a very large channel bandwidth.
However, given the enormous available bandwidth (tens of terahertz) of the low-loss
transmission window of silica single-mode fibers, this would lead to a data rate which is far
higher than what can be handled by optoelectronic senders and receivers.
Also, various types of dispersion in the transmission fiber would have very detrimental
effects on such wide-bandwidth channels, so that the transmission distance would be strongly
restricted.
WDM solves these problems by keeping the transmission rates of each channel at reasonably
low levels (e.g. 10 Gbit/s) and achieving a high total data rate by combining several or many
channels.
Two different versions of WDM, defined by standards of the International
Telecommunication Union (ITU), are distinguished:
Coarse wavelength division multiplexing (CWDM, ITU standard G.694.2)
Dense wavelength division multiplexing (DWDM, ITU standard G.694.1)
Page 16 of 21
7.5
CDM allows signals from a series of independent sources to be transmitted at the same time
over the same frequency band.
This is accomplished by using orthogonal Code division multiplexing codes to spread each
signal over a large, common frequency band. At the receiver, the appropriate orthogonal code
is then used again to recover the particular signal intended for a particular user.
The key principle of CDM is spread spectrum. Spread spectrum is a means of communication
with the following features:
1. Each information-bearing signal is transmitted with a bandwidth in excess of the
minimum bandwidth necessary to send the information.
2. The bandwidth is increased by using a spreading code that is independent of the
information.
3. The receiver has advance knowledge of the spreading code and uses this knowledge to
recover the information from the received, spread-out signal.
Spread spectrum seems incredibly counterintuitive. Weve spent quite some lecturing hours
studying ways to transmit information using a minimum of bandwidth. Why should we now
study ways to intentionally increase the amount of bandwidth required to transmit a signal?
If you understand CDM, you will see that spread spectrum is a good technique for providing
secure, reliable, private communication in an environment with multiple transmitters and
receivers. In fact, spread spectrum and CDM are currently being used in a number of
commercial cellular telephone systems and satellite communications.
Application of CDM (in this case CDMA) in cellular telephony
Each mobile device is assigned unique 64-bit code (called Chip spreading code)
To send a binary 1, mobile device transmits the unique code
To send a binary 0, mobile device transmits the inverse of code
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At the receiver
Gets summed signal
Multiplies it by receiver code
Adds up resulting values
Interprets as a binary 1 if sum is near +64
Interprets as a binary 0 if sum is near 64
Summed signal received by base station: +3, -1, -1, +1, +1,
-1, -3, +3
7.6
DMT is a form of multicarrier modulation that encodes bits in the frequency domain. It is
now used in certain wireless communication systems (802.11) and Digital Subscriber Line
(DSL) technologies.
Through the use of DMT, DSL technology enables very high speed connections from
individual computers to switching stations over a standard copper telephone line.
DMT places the data onto 247 separate sub-channels, each 4 KHz wide. This is like
having 247 different dial-up lines connected to a computer all at the same time!
On top of that DSL allows a subscriber to be able to receive phone calls over the same line at
the same time, without risk of disconnection or data loss.
The existing local loop*1 can handle up to 1.1 MHz bandwidth
The first 4 KHz bandwidth is used for regular telephone voice service
Rest of the bandwidth is divided into 256 channels each occupying a bandwidth of
4.312 KHz
Each sub-channel can carry up to 60 Kbps data rate. (4 KHz 15 bits/Signal Change) = 60
Kbps
Allocation of the 1.1 MHz bandwidth (refer to Figure 7.6 below)
Channel 0 is used for voice
Channel 1 to 5 are not used to allow a gap between voice and Data
Channels 6 to 30 (25 channels) are used for up stream transmission and control. One
for Control and 24 for data. Thus upstream data rate is: 24 4 KHz 15 bits/ Signal
change = 1.44 Mbps
Channel 31 to 255 (225 channels) are used for downstream transmission, one for
control and 224 for data. Thus the downstream data rate is : 224 4 KHz 15 bits/
Signal change = 13.4 Mbps
Voice
Ch 0
0
Upstream
Ch 6 to 30
4
26
Downstream
Ch 31 to 255
108 138
1104
KHz
Figure 7.6
Note that these are theoretical maximum bandwidth. The actual practical data rate is usually
lower and depends on:
The S/N Ratio of the link
The distance between the customer location and the Central Office*2 (CO)
Note: 1.Central Office (CO) is the physical building where the local telephone switching
equipment is located. All telephone lines in a town lead to the CO.
Note: 2. the pair of wires (twisted pair) connecting a telephone subscriber to a CO.
Page 19 of 21
Being an adaptive technology, the ADSL modems, when turned ON, check the quality of line
and automatically adjust the data rate.
The noise level of each sub-channel is monitored; if a sub-channel with a particular
frequency becomes too noisy, data will be reallocated from the noisy sub-channel to
others with less noise.
As the frequency response of the channel changes with time, the ADSL system
constantly shifts data from one sub-channel to another, searching for the frequency
distribution that allows for an optimal data rate.
7.7
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7.8
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