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The transmission channel can be a twisted pair or coaxial cable.9.2.4 Formation of a Basic SupergroupFor higher capacity channels, five basic groups are combined to form a basicsupergrou

SIGNAL PROCESSING IN THE

Signal processing has two major aims:

(1) To improve the quality of signal transmission over the telephone nication channels

commu-(2) To lower the cost of communication by improving the efficiency of channeluse

In general, the quality of a communication channel tends to deteriorate with distance

In addition, long distance channels are expensive to establish and maintain Itfollows that the more messages that can be transmitted in a given time, the lower thecost per message It is therefore on the long-distance channels (trunks or tolls) thatsignal processing techniques have proven to be most successful In this chapter thecommon signal processing techniques used in the telephone system and some of thecircuits employed will be examined

Frequency division multiplex (FDM) is a technique in which a number of signals can

be transmitted over the same channel by using them to modulate carrier signals with

267

different and appropriate frequency so that they do not interfere with each other Theassignment of specific carrier frequencies to radio stations for broadcasting and otherpurposes is, in fact, FDM It can be used with amplitude modulation as well as otherforms of modulation In the context of the telephone, FDM is used in conjunctionwith amplitude modulation.

In normal amplitude modulation, the carrier, upper and lower sidebands aretransmitted When the depth of modulation is 100%, the amplitude of the carriervoltage is twice that of the sidebands The power in the carrier is therefore23of thetotal Unfortunately, the carrier has no information content Each of the sidebandscontains16of the total power In radio, 100% modulation is almost never used so thepower content of the sidebands is much less than described It is noted that theinformation content is duplicated in the two sidebands It is clear that one way tobeat the corrupting influence of noise on the information content of the transmission

is to put as much as possible, if not all, of the available power into one of thesidebands An added advantage to this scheme is that the required bandwidth isreduced to one-half of its original value Clearly, this would allow twice as manymessages to be sent on the same channel as before The transmission of only onesideband in an AM scheme is called single-sideband (SSB) modulation The price to

be paid for this advantage is that to demodulate an SSB signal, it is necessary toreinstate the carrier at the receiver The reinstated carrier has to be in synchronismwith the original carrier, otherwise demodulation yields an intolerably distortedsignal Providing a synchronized local oscillator requires complex equipment at thetransmitter as well as at the receiver In SSB radio, an attenuated form of the carrier

is transmitted with the signal This is used to synchronize a local oscillator in thereceiver In the telephone system, a centrally generated pilot signal is distributed toall offices for demodulation purposes In some cases, a local oscillator withoutsynchronization is used If the frequency error is small (approximately
5 Hz),successful demodulation can be achieved [7]

9.2.1 Generation of Single-Sideband Signals

A block diagram of the SSB generator is shown in Figure 9.1

268 SIGNAL PROCESSING IN THE TELEPHONE SYSTEM

f1ðtÞ ¼A

To eliminate the upper sideband, it is necessary to have a bandpass filter with a verysharp cut-off at the carrier frequency This is not easy to achieve in practice, but thetask is made simpler when the modulating signal os has no low-frequencycomponents Under this condition, crystal and electromechanical filters can bedesigned to suppress the upper sideband This is the case for a telephone voicechannel which is nominally from 300 to 3000 Hz

From Equation (9.2.3) only the lower sideband is transmitted At the receivingend, the signal is demodulated (multiplied) by (the recovered) carrier, cos ot Theresult is

9.2.3 Formation of a Basic Group

In the trunk or toll system, 12 channels form a basic group The basic group isformed by SSB modulation of 12 subcarriers at 64, 68, 72; ; 108 kHz Thesecarriers are generated from a 4 kHz crystal-controlled oscillator and multiplied bythe appropriate factor The upper sidebands are removed and they are added together

to form the group Figure 9.2(a) shows a block diagram for channel 1 Figure 9.2(b)shows the spectrum of the basic group

For a small-capacity trunk, the basic group may be transmitted without furtherprocessing The transmission channel can be a twisted pair or coaxial cable.9.2.4 Formation of a Basic Supergroup

For higher capacity channels, five basic groups are combined to form a basicsupergroup Figure 9.3(a) shows the block diagram of the basic supergroup 1 Notethat to make the filtering problem easier, the carrier frequency is chosen to be

420 kHz Figure 9.3(b) shows the frequency spectrum of the basic supergroup.Table 9.1 shows the carrier frequencies and bandwidths for each basic super-group

For a 60-channel trunk, the signal can be transmitted in this form Again a twistedpair with coil loading or amplification and coaxial cable may be the medium oftransmission

By organizing the 12 basic groups into a basic supergroup of 5 it clear that thesubcarrier frequencies, the balanced modulators and bandpass filters can all beduplicated five times over If the basic group had been made larger, new subcarrierfrequencies would have had to be generated and bandpass filters of differentcharacteristics would have been necessary

9.2.5 Formation of a Basic Mastergroup

To create a 600-channel trunk, 10 basic supergroups are combined to form a basicmastergroup The frequency spectrum of the basic mastergroup is shown in Figure 9.4

Transmission Systems for Communications, 4th Ed., AT&T, Bell Labs, 1970.

270 SIGNAL PROCESSING IN THE TELEPHONE SYSTEM

Note that there are gaps of 8 kHz between each basic supergroup spectrum Thesegaps are designed to make the filtering problem easier.

The carrier frequencies and bandwidths of the 10 basic supergroups are given inTable 9.2

The basic supergroup can be transmitted over coaxial cable or it can be used tomodulate a 4 GHz carrier for terrestrial microwave transmission or even sent over asatellite link

Transmission Systems for Communications, 4th Ed., AT&T, Bell Labs, 1970.

Transmission Systems for Communications, 4th Ed., AT&T, Bell Labs, 1970.

Other larger groups can be formed, for example 6 mastergroups may be combined

to form a jumbogroup with 3600 voice channels

To recover the original baseband signals from the various groups, the appropriatenumber of filtering=demodulation processes will have to be carried out At eachstage of the demodulation process, the correct carrier will have to be reinstated forthis to be possible

9.3 TIME-DIVISION MULTIPLEX (TDM)

In FDM, voice signals were ‘‘stacked’’ in the frequency spectrum so that many suchsignals could be transmitted over the same channel without interference In time-division multiplex (TDM), each voice signal is assigned the use of the completechannel for a very short time on a periodic basis The theoretical basis of thistechnique is the Sampling Theorem An informal statement of the sampling theoremis:

If the highest frequency in a signal is B Hz, then the signal can be reconstructed fromsamples taken at a minimum rate of 2B samples per second (Nyquist sampling rate orfrequency)

The proof of this theorem is beyond the scope of this book However, there are anumber of practical problems which arise in the application of the theorem:(1) The theorem assumes that the samples have infinitesimally narrow pulsewidths This is clearly not so in a practical circuit The sampling rate isusually chosen to be higher than the Nyquist frequency since it is theminimum; it is discrete to avoid extreme conditions when dealing with animperfect situation

(2) The theorem assumes that an ideal low-pass filter is used to remove allfrequencies above B Hz ahead of the sampler When using a practical filter, it

9.3.1 Pseudodigital Modulation

To code an analog signal in pulse form one can use the height of the pulse, the width(or duration), or the position of the pulse relative to standard position When theheight is used, it is called pulse-amplitude modulation (PAM) When the coding is interms of the width it is called pulse-width modulation (PWM) and when the position

is used it is called pulse-position modulation (PPM) Pulse height, width, andposition are analog quantities which in turn can be quantized and represented by abinary code where the digits are present, 1, or absent, 0 When this has been done themodulation scheme is called pulse-code modulation (PCM) Although PCM isqualitatively different from the other modulating schemes, they are compared inFigure 9.6

These schemes would work equally well in a noiseless environment When noise

is present, and it always is, PCM has a clear advantage over the others In the case ofPAM, PWM and PPM the receiver has to determine what the original amplitude,width, and position were respectively in order to reconstruct them In PCM, thedecision is simplified to whether the digit sent was a 1 or a 0 In all cases, it isnecessary to transmit timing information with the signal so that the receiver knowswhere the bit stream starts and stops

9.3.2 Pulse-Amplitude Modulation Encoder

To illustrate the design principle of a PAM communication channel, a four-channelPAM system has been chosen The coder or commutator is shown in Figure 9.7.The master clock drives the four-phase ring counter The ring counter drives foursampling gates on and off in the correct sequence When one of the four outputs is

on, 1, all the others are off, 0; so only the sampling gate with the 1 is connected tothe adder Note that the second input to the fourth sampling gate are connected to themaster clock This means that channel 4 will always produce a positive pulse Theamplitude of this pulse is adjusted to be higher than the most positive value of theanalog input voltage This is called the synchronization pulse or sync pulse for short

It is used to identify and time the other channels

The design of the component circuits now follows

9.3.2.1 Four-Phase Ring Counter The four-phase ring counter and its timingdiagram are shown in Figure 9.8

Figure 9.6 A comparison of PAM, PWM, PPM and PCM Note that PAM, PWM and PPM are not truly digital since they convey information by the variation of analog quantities, that is, amplitude, duration and position in time Reprinted with permission from B P Lathi, Modern Digital and Analog Communication Systems, CBS College Pub., New York, 1983.

It can be seen from the diagram that, in the time taken by one frame, the outputpulses go through one cycle The outputs are used to drive the sampling gates.

9.3.2.2 Series Sampling Gate The configuration of the series sampling gate

is shown in Figure 9.9

The transistor is an open-circuit when the control signal is a 0 and a short-circuitwhen it is a 1 The output is as shown.

9.3.2.3 Shunt Sampling Gate The shunt sampling gate is shown in Figure9.10

The transistor acts as a switch and short-circuits the output when the gate voltage

is a 1 When the gate voltage is a 0, it is an open-circuit and a path exists between theinput and the output

9.3.2.4 Series-Shunt Sampling Gate The two circuits shown above have aninherent deficiency because, when the transistor is on, its source-to-drain impedance

is low but not equal to zero To improve the performance, the action of the two gatescan be combined as shown in Figure 9.11

278 SIGNAL PROCESSING IN THE TELEPHONE SYSTEM

9.3.2.5 Operational Amplifier Sampling Gate The circuit is shown inFigure 9.12.

The operational amplifier is connected to give a gain of R2=R1when the transistor

is in the off state When the transistor is on, R2 is short-circuited and the gain isreduced to unity

9.3.2.6Multiplier Sampling Gate A PAM sampler can be seen as a plication of the analog signal and a train of pulses The process is illustrated inFigure 9.13

multi-One of the best methods for accomplishing analog multiplication is use the quadrant analog multiplier This circuit was described in Section 2.6.3 A practicalintegrated circuit realization of this is the MC1595 (four-quadrant multipliermanufactured by Motorola Semiconductor Products, Inc.)

four-9.3.2.7 The Adder The adder was discussed in Section 4.4.3.5

9.3.3 Pulse-Amplitude Modulation Decoder

The first step in the recovery of the original three signals is to reverse the action ofthe commutator by separating them into their respective channels Low-pass filters

are then used to reconstruct the analog waveform from the PAM pulses The PAMdecoder system is shown in Figure 9.14.

The incoming signal is fed into the Schmitt trigger The trip level of the Schmitttrigger is set so that only the large sync pulse will trigger it The output of theSchmitt trigger is then used to synchronize an astable multivibrator The astablemultivibrator then runs in synchronism with the master clock in the PAM coder Theoutput of the astable multivibrator drives a four-phase ring counter which producesfour sequential output pulses in synchronism with the ring counter in the encoder.These pulses are used to drive the control gates of the sample-and-hold (S=H)circuits The incoming signal is also fed to the analog inputs of the sample-and-holdcircuits Since the control gate of only the S=H-1 is open during the period allotted tochannel 1, the pulse amplitude of the signal in channel 1 is passed onto S=H-1 Theother channels follow in sequence The low-pass filters remove the high-frequencycomponents of the PAM signals, producing a replica of the original analog signals.9.3.3.1 Schmitt Trigger Design The circuit diagram of the Schmitt trigger isshown in Figure 9.15

The circuit is designed so that, with no input, Q1 has no base current and it istherefore off Q2is supplied with base current from the resistive chain Rc1, R1and R2and it is therefore on Current flows in Q2and with the correct choice of Rc2, Q2will

multi-plication of two signals.

280 SIGNAL PROCESSING IN THE TELEPHONE SYSTEM

be in saturation The emitter current of Q2flows in REand sets up a voltage VE If anincreasing positive voltage is applied to the input, when it reaches the value

VEþVBE, Q1 will start to conduct Current is drawn through Rc1 causing thevoltage on the base of Q2to drop Q2conducts less vigorously and the voltage across

RE tends to drop But this drop in voltage at the emitter causes the base-emittervoltage of Q1 to increase rapidly This is a form of regeneration and proceeds veryfast, ending with Q1 conducting and in saturation and Q2 cut off

When the input voltage is decreasing, there comes the point when it is slightlybelow the value VEþVBE Q1 conducts less current, causing its collector voltage totend to rise and its emitter voltage to tend to drop The rising trend at the collector of

Q1 is passed onto the base of Q2by R1 and C The combined effect of a decreasing

VE and a rising base voltage causes Q2 to switch on regeneratively and go intosaturation The Schmitt trigger reacts to a slowly changing input voltage byproducing a voltage step when its trip level is exceeded This happens for increasing

as well as decreasing voltages The design of the Schmitt trigger is best illustrated by

an example

Example 9.3.1 Schmitt Trigger Design Design a Schmitt trigger circuit so that ittriggers when a voltage in excess of 3.0 V is applied at the input The dc supplyvoltage is 10 V and two NPN silicon bipolar transistors with b ¼ 100 are provided

VCEðsatÞ¼0:5 V and VBE¼0:7 V The load driven by the Schmitt trigger requires acurrent of 1 mA

Solution The transistors are made of silicon; therefore, VBE¼0:7 V For the circuit

to trigger at 3.0 V, VEmust be designed to be equal to ð3:0  0:7Þ ¼ 2:3 V Since theSchmitt trigger is to drive a load that requires 1.0 mA, it is good design practice to

282 SIGNAL PROCESSING IN THE TELEPHONE SYSTEM

instanta-not an ideal short-circuit when closed and the voltage source has an internalresistance Rs This means that C charges up with a time-constant t ¼ CRs Solong as the pulse width of the driving source is significantly longer than t the errorcan be regarded as insignificant Another practical consideration is that, when S isopen, C can lose its charge due to leakage in the dielectric of the capacitor and thefinite impedance of the load driven by the S=H.

A simple but practical circuit for the S=H is shown in Figure 9.17

The operational amplifiers are connected as voltage followers with a gain of unity.The output impedance of A1 is low enough for it to drive the required charge into thecapacitor The N -channel JFET is switched on by a pulse applied to the gate and thecapacitor charges up to the value of the input voltage When the JFET switch isturned off, the high input impedance of A2 drains minimal current from C Thedesign of the S=H circuit is best illustrated by an example

Example 9.3.2 Sample-and-Hold Circuit The S=H circuit shown in Figure 9.17uses a JFET as a series sampling gate The voltage follower A2 takes a current of

500 nA The JFET has a source-to-drain resistance of 25 O when it is in the on stateand may be considered to be an open-circuit when it is in the off state The signal

284 SIGNAL PROCESSING IN THE TELEPHONE SYSTEM