telephone network interfacing - pippin tech network interfacing measuring up from there. the...

to the customer site mostly are not. The vast majorityof users interface to the network via an analog technol-ogy that is little different from that employed in Alex-ander Bell’s days. This is beginning to change withthe introduction of digital last-mile technologies likeISDN, T-1, and an Asynchronous Digital SubscriberLine (ADSL).

Incidentally, in industry jargon, your local phonecompany is a local exchange carrier (LEC) or simplya telco. A long distance company is an inter-exchangecarrier (IEC).

Speech CodingThe bit rate of 64 kbps was chosen to support phone-

grade speech audio encoded using a modified pulsecode modulation (PCM) technique. When we make aplain old telephone (POTS) call, our speech is sampledat an 8 kHz rate and encoded into a digital word 8bits long. Telco engineers call this 64 kpbs bitstreama digital signal level 0 (DS-0) channel.

Theword length iswhatdeterminesdynamic range—and 8 bits would only permit 48 dB were it used in stan-dard PCM linear fashion. A primitive kind of compres-sion is used to stretch thedynamic range:mLaw in NorthAmerica and much of Asia, and A-law in Europe (seeFigure 3.10-1). This is a scheme that equalizes the stepsize in dB terms across the dynamic range—a smallerstep size on low level signals reduces quantization noiseand improves effective dynamic range to the equivalentofabout13bits.Thus, thequantizationnoise(anddistor-tion) is approximately a fixed percentage of the signalamplitude, regardless of its level.

The process of conversion and companding is donein specialized analog-to-digital (A/D) and digital-to-

INTRODUCTION

From earnest political talk presentations to raucousmorning shows, listener involvement via telephone isan important programming element at many radio andtelevision stations. When we want to create a two-wayconnection with our listeners, we will probably beusing the dial-up telephone network.

Radio news departments rely extensively uponphoners to get reporters and newsmakers on the air ina timely fashion. Why are the people who run local TVnews so concerned with avoiding the dreaded talkinghead—that is, the anchor simply reading a story intothe camera? Because they’ve discovered thatbeingthere is better. The same is true for radio.

Today, integrated services digital network (ISDN)lines combined with modern audio compression tech-niques permit instant full fidelity remotes from almostanywhere in the world.

This chapter will explore all of the ways to integratethe ubiquitous telephone network into broadcast opera-tions. First, we’ll learn about the nature of the variousservices available from telephone companies. Thenwe’ll investigate ways to interface them to our sta-tion facilities.

THE TELEPHONE NETWORK

As we transition our broadcast facilities to digital sys-tems, it is interesting to note that the standard voicetelephone network is almost entirely digitized and hasbeen so for many years. The watershed event wasIllinois Bell’s 1962 installation of a T-carrier system—the first widespread commercial application of digitalaudio. Telephone engineers appreciate digital technol-ogy for the same reason broadcasters do: reduced sus-ceptibility to noise and other disturbances, and im-proved ability to switch, monitor and maintain thecircuits.

While the worldwide dial-up telephone network isan amazing achievement, it is mostly made from asimple ubiquitous element: digital circuit-switchedchannels of 64 kbps each.Circuit-switchedmeans thatthe channel is connected end-to-end with the entirecapacity available for the duration of the call. (This isin contrast to packet-switched systems, such as theInternet, where capacity is shared among users andthere is usually no guaranteed bandwidth.)

While most of the network infrastructure is digital,the last-mile copper connections from the central office

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3.10TELEPHONE NETWORK INTERFACING

STEVE CHURCHTELOS SYSTEMS, CLEVELAND, OH

Figure 3.10-1. mLaw PCM coding within the telephone networkcauses the noise to be approximately a fixed percentage regard-less of level.

SECTION 3: AUDIO PRODUCTION FACILITIES

analog (D/A) integrated circuits calledcodecs(CODer/DECoders). The method is specified by the Interna-tional Telecommunications Union (ITU) as standardG.711.

2-Wire and 4-WireBoth speech directions are mixed together on the

usual analog lines with which we are most familiar,but this is not the way signals are handled within thetelephone transmission and switching network. Non-copper transmission media such as microwave radio,satellite and fiber-optic cables are one-way only, sothe paths must be kept independent. Even when copperis used, long-distance links are kept separated so thatamplification can be inserted. A standard analog POTScircuit is 2-wire, because it arrives on two wires. Thenetwork is internally4-wire, so named because in thepast, a 4-wire circuit needed a separate wire pair foreach of the send and receive transmission directions—four wires altogether.

The Traditional Analog LineThe traditional telephone lines provided by the

phone company are known officially as subscriberloops, trunks or simply CO (central office) lines.(Trunks used to refer only to lines destined for privatebranch exchange (PBX) systems and may have in-cluded special signaling as well.)

Because these are 2-wire circuits, the CO uses a 2-to-4-wire converter (also called ahybrid) to interface theanalog lines to its internal 4-wire system, as shown inFigure 3.10-2. This process happens on theline card,which is also responsible for digitization, talk batteryinsertion, off-hook detection, and ring generation.

Talk Battery and RingingThe talk batterydirect current (dc) voltage and the

conversation audio appear together on the phone pair.The talk battery leaves the exchange at148 V and islimited to 20–50 mA by a series resistor. The resistor’s

value is selected to complement the resistance of theloop. The dc resistance of the loop itself varies froma few to 1,300V depending on length. Because of thisseries resistance, when a line is off-hook, its voltageat the customer equipment drops to around112 V.

For ringing, an ac voltage of 90 vrms at 20 Hz issuperimposed on the line. Talk battery is maintainedduring ringing, so that the resulting signal has a sinus-oidal shape shifted 48 V to the negative.

Talk signals are ac coupled with nominal impedanceof 600V. However, some CO equipment uses compleximpedancecoupling,and thenatureof thetelephonenet-work usually results in the actual impedance as pre-sented to the user rarely being the specified simple600V. This turns out to be an important issue for broad-cast interfacing, which we will discuss in detail later.The basic parameters are summarized in Table 3.10-1.

Frequency ResponseFor ordinary subscriber loops, the phone company

specifies a frequency response of 300 Hz to 3.4 kHz.In the not-too-distant past when all local calls wereconnected at the exchange by metallic contacts, better

Table 3.10-1

Phone loop characteristics.

Parameter Typical U.S. Values Operating Limits

Talk Battery Voltage 148 VDC 147 to 1105 VDC

Loop Current 20 to 80 mA 20 to 120 mA

Loop Resistance O to 1300 ohms 0 to 3600 ohms

Loop Loss 8 dB 17 dB

Distortion 150 dB N.A.

Ringing Signal 20 Hz, 90 VRMS 16 to 60 Hz,40 to 130 VRMS

Noise (objective) 169 dBm0 to 180 mi,150 dBm0 to 3000 mi(116 dBm0 talk level)(C msg weight)

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Figure 3.10-2. Two-wire circuits have both directions on a single pair of wires, which are separated for switching and long-distancetransmission into 4-wire signals with hybrids.

TELEPHONE NETWORK INTERFACING

measuringup from there. The reference noise level isone picowatt, which corresponds to190 dBm. Thus,a noise level of160 dB relative to 0 dBm would bereported as 30 dBrn noise (dBrn4 dB above referencenoise). Note that, according to this method, the higherthis number, the worse the noise.

Be aware also that when telephone people measurenoise, they are measuring onlyidle channel noise.Thisis an important difference, since in digital systems idlechannel noise is not the same as the traditional (S/N)measurement in analog systems. Noise in a digitalsystem will generally increase when a signal is present.This effect is calledmodulationor quantizationnoiseand is primarily dependent upon the number of bitsused for quantization.

A C-message weightfilter is employed when meas-uring phone line signal-to-noise ratio (S/N). (See Fig-ure 3.10-5.) The C-message curve was developed yearsago to simulate the frequency response of an old-styletelephone earpiece and, accordingly, it has consider-able low-frequency roll-off. This means that a line canhave significant hum and other low frequency noiseand can still meet the officially mandated noise specs.While this makes life easier for the phone companytechnicians, it can be troublesome when a broadcasteris trying to use phone audio on the air. If noise is aserious problem, try to get the technician to switch thenoise meter to the flat position. The measuring setusually does have this option available.

frequency response was likely to be had on manyconversations. Today almost all calls are digitized andare strictly limited to a 3.4 kHz bandwidth by the sharplow-pass filters required for proper digitization. Thephone network’s 8 kHz sampling rate permits a theoret-ical Nyquist frequency of 4 kHz, but a 600 Hz transitionband is necessary for anti-aliasing and reconstructionfiltering (see Figure 3.10-3).

Noise and LevelA 1971 Bell System survey of the phone network

nationwide determined that the average conversationhad a level of116 dBm. Of course, as anyone whohas wrestled with broadcast-to-telco interfacingknows, incoming level varies tremendously, with arange of perhaps140 to 14 dBm, as illustrated inFigure 3.10-4.

Send audio (that is, audio fed into the telephoneline) must be limited to19 dBm as specified in Part68 of the Federal Communication Commission (FCC)Rules. Audio loss on any given local loop is limitedby tariff to 8 dB or less. This loss limit, however,applies only to the loop from the CO to the subscriberand does not include the rest of the signal path. Also,the 8 dB loss may occur at each end of a conversationpath: once at the calling party end and again at thecalled party end, for a total loss of 16 dB.

The phone engineering people measure noiseupside-down, defining a reference noise floor and then

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Figure 3.10-3. The low-pass filters required for digital transmission restrict frequency response. This response curve is for a codec thatis widely used in the telephone network. (Note also the significant low frequency roll-off).


Figure 3.10-6. DTMF tone keypad frequency assignment. Thefour tone pairs in the last column are for special applications.

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Figure 3.10-4. Signal and noise level references used in telephone engineering.

DTMF Tone DialingDual Tone Multiple Frequency (DTMF) dialing uses

two frequencies for each digit in order to avoid talk-off—that is, the tone detector accidentally sensingvoice as a dial command. In addition, the frequencieswere carefully chosen to avoid problems with har-monic distortion causing false detection. There are four

Figure 3.10-5. C-message weight frequency response curve.

low group frequencies, one for each button row, andfour high group frequencies, with one assigned to eachcolumn as shown in Figure 3.10-6. Tolerance is61.5%for the encoder and62% for the digit receiver. Thetime required to recognize any digit tone is 50 msecwith an interdigit interval of another 50 ms. Low grouptones are supposed to be sent at a level between110


hangs up. Thus, we can use the presence of dial toneas a back up to cause a disconnect when the loop-current detection methods fail. An important consider-ation is to prevent false talk-off from noise, applauseor other spectrally rich audio. Using software basedstatistical methods ensures that the dial tone isreallypresent before terminating the connection.

Caller IDCaller ID (CID) allows you to know the phone num-

ber of the caller. This capability is useful for call-inshows, where it might be desirable to deny access toproblem callers. The technology is simple. Betweenthe first and second ring, the information is sent in apacket using a 1200-baud modem. This is exactly thesame modulation scheme used in normal computermodems operating at this rate. Customer equipmentnormally suppresses the first ring so that the answeringuser does not take the call before the CID informationis fully transmitted.

Loading CoilsA typical #24 gauge phone pair attenuates a 3 kHz

signal 2.5 dB per mile due to capacitive effects. On an8 mile (12.9 km) long line, high-frequency attenuationwould thus be 20 dB, a significant amplitude distortion.Loading coils are toroidal inductors, which counter theeffects of the phone pair’s natural capacitance. Whilethe coils are effective at flattening out the responsewithin the voice band, the roll-off above 3.5 kHz isdevastating, as shown in Figure 3.10-7.

Physically, load coil banks are long cylinders, withthe individual donut-like coils stacked one on top ofthe other inside. They are typically placed at 3,000(.9 km), 4,500 (1.4 km), or 6,000 (1.8 km) ft intervalsalong the phone cables. Generally, loading coils arefound only on cables of greater than 3 miles (4.8 km)in length.

As we shall see, loading coils can create problemsfor the hybrids used in broadcast interfaces.

4-Wire CircuitsIt is possible to purchase analog 4-wire circuits from

telcos. These are used where it is desirable to maintain

and 16 dBm; ideally, tones in the high group aretransmitted with 2 dB greater level in order to compen-sate for high-frequency roll-off in the phone line.

Loop Start and Ground StartCentral office lines come in two basic configura-

tions: loop start and ground start. Loop start is the kindthat is most common. In this kind of circuit, the COprovides talk battery to the line at all times and detectsthat an off-hook condition is occurring when the termi-nal equipment connects and causes current to flowbetween the tip and ring. (Incidentally, the termstipandring originated with the description of the circuitsbeing on the tip and ring of the patch cords that usedto be used by telephone operators.) With ground startcircuits, the CO waits for a connection from the ringwire to ground before connecting talk battery, at whichtime the terminal equipment removes the ground con-nection to establish a balanced talk path. When thecalling party hangs up, a ground start circuit removestalk battery. A loop start circuit may or may not providea momentary interruption or reversal of the talk batterywhen the calling party terminates.

Many PBXs are designed to work with the groundstart circuits because the possibility of collision is re-duced. Collision occurs when the phone system triesto seize a line for an outgoing call just as that line isringing in.

Disconnection: Calling Party ControlLoop-current interruption occurs on most telco lines

when the calling party hangs up. It is sometimes re-ferred to as calling party control (CPC), since thecalling partycontrolsyour equipment when he hangsup. The CPC may turn off an answering machine, forexample, or extinguish the winking light on a held lineon a key phone. The CPC interruption was probablynever intentional, having been a by-product of earlymechanically switched relay-controlled exchanges.Thus, some phone lines do not provide this functionor they provide it unreliably. However, with the prolif-eration of answering machines that rely upon CPC,most central office equipment now has this capabilitydesigned in. In some cases, it is necessary to specifi-cally request this feature from the phone company ona per line basis.

Loop-currentreversal,on the other hand, has longbeen a phone company signaling method. First usedbetween the telco’s own central offices, loop-reversalwas later employed to communicate with some largepremises PBX systems. Thus, lines that are set up forPBX use, or originate at central offices with largeconcentrations of business customers, sometimes usethis method. (However, the preferred and more modernsituation for PBX control is to use ground-start lines.)

While most exchanges do provide CPC, there aresome that do not reliably provide it or provide it aftera variable time delay. Most PBXs do not generate it.However, every telco CO in the United States eventu-ally returns dial tone to its lines when the calling party

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Figure 3.10-7. Frequency response with and without loading coils.


separation in the two speech paths. They are not dial-up, but rather end-to-end hardwired. This service hastraditionally been used by television remote trucks forconnection of remote production intercom systems.With the introduction of digital hybrid interfaces, useof this approach has been in decline. ISDN offers4-wire capability at a lower cost and with fewer has-sles, so it will probably supplant these analog linesover time.

Foreign Exchange (FX) LoopsFX provides local telephone service from a central

office that is outside (foreign to) the subscriber’s ex-change area. If a station is located in the suburbs andthe choke network central office is downtown, FXloops will be needed to connect your lines. When thephone is picked up, you get dial tone not from yourlocal suburban CO, but from the downtown office. FXservice is also sometimes used to extend your coverageinto another city, so that people can call the stationwithout paying a toll charge and calls can be madewithin that city without incurring toll charges. Forinstance, if the studio is in Cleveland and the goal isto serve listeners in Akron as if they were local, FXservice could be the answer.

An FX loop is a 4-wire circuit with hybrids at eachend, at each terminating central office. Since FX loopsadd an extra layer of hardware to the phone audio,they are another source of problems for on-air interfac-ing. They usually are engineered to have a few dBloss and they add to the impedance complexity ofthe line.

FX circuits are usually expensive and pose certaintechnical challenges. Since, as we will learn later inthis chapter, hybrids are imperfect, a potential for aspecial kind of feedback calledsinging exists. Thisresults from the inevitable leakage from the send tothe receive ports at each hybrid. The phone people

solve this problem by inserting a pad—anywhere from5–8 dB is common.

Choke NetworksMost stations need special high volume exchanges

for their contest and request lines. This requirementprobably results from the days when aggressive pro-gram directors (PDs) desired the publicity thatburningout a phone exchange would generate.

The choke network works by diverting calls begin-ning with the unique choke prefix around the localserving central office and sending them directly to thechoke switching exchange, usually located downtown(see Figure 3.10-8). The phone company dedicatesvery few talk paths (wire trunks or special carrierequipment) to the task of connecting the caller’s serv-ing CO choke ports to the choke exchange. The usualswitching and routing process is bypassed. Unfortu-nately, only a very limited number of paths are gener-ally provided. In the densely populated Los Angelesarea, for instance, only three connections exist frommost central offices. In addition, the poorest facilitiesare often given over to the high volume service.

Generally, unless you are near the choke centraloffice, the FX circuits previously described are em-ployed to connect the choke CO to your serving CO.This is one of the reasons why choke circuits oftenhave a lower level than standard lines. Because oftheir higher complexity, choke lines also usually havebumpier impedance curves, making good hybrid per-formance difficult to achieve due to the problem offinding appropriate balancing network values. This isespecially a problem with simple analog hybrids.

In some areas, FX circuits are being replaced byinternal call forwarding. This means that a publishednumber is actually being software forwarded to a realnumber originating from your local serving CO. Themain advantage to this approach is lower cost, since

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Figure 3.10-8. Typical choke network transmission path.


col conversion functions. A traditional TA has an ISDNconnection on one end and one or two bit stream portson the other, usually using the V.25 or X.21 connectors.Modern broadcast equipment combines this capabilitywith the audio encoding equipment into one inte-grated unit.

SPIDsService profile identification numbers (SPIDs) are

only required when you are using the National I-1ISDN protocol in the United States. This number isgiven to the user by the phone company and must beentered into the TA in order for the connection tofunction. SPIDs usually consist of the phone numberplus a few prefix or suffix digits.

The intention of the SPID is to allow the telco equip-ment to automatically adapt to various user require-ments by sensing different SPIDs from each type orconfiguration of user terminal. For instance, multi-button phones could retain function assignments whenmoving from line to line. In this case, the line numberwould probably not be used as the SPID. None of thismatters with our application, but we must enter theSPIDs nevertheless. (Over time, it may be possiblethat a standard SPID could be used for all broadcastcodec applications. A proposal that would allow thisis being considered.)

If you are using the National I-1 protocol, your telcoservice representative must give you one or two SPIDnumbers for each line ordered. You will get one SPIDfor each B channel you need. Upon power-up, connec-tion of the ISDN line or boot, the TA and the telcoequipment go through an initialization/identificationroutine. The TA sends the SPID and, if it is correct,the network signals this fact. Thereafter, the SPID isnot sent again to the switch. You must have this SPIDnumber, and it must be 100% correct, or the systemwill not work. Do not let the installer depart withoutleaving your SPID number(s).

Directory Numbers (DNs)Directory numbers (DNs) are the telephone numbers

assigned to the ISDN line. You may be assigned oneor two, depending upon the line configuration. If youhave two active ISDN B channels, you will usuallyhave two DNs. However, the physical channels areindependent from the logical numbers. A call comingin on the second number will be assigned the firstphysical B channel, if it is not already occupied. There-fore, there must be some way for the TA to sort outwhich call goes to which channel/line. The DN is usedfor this function.

When a call rings in, it contains set-up information,which includes the DN that was dialed by the originat-ing caller. The last seven digits are matched with theDNs programmed into the TA and the proper assign-ment is made. However, it is not usually necessaryto explicitly enter them, as they are almost alwayscontained within the SPID, and most TAs are smartenough to look there first. The only time a DN mustbe entered is in the very rare case where the last seven

you do not have to pay the premium for the FX circuit.However, there usually is a smaller call-forwardingcharge.

ISDN: Basic Rate Interface (BRI)ISDN allows a direct digital connection to the tele-

phone network. In addition to the quality advantagesdigital transmission offers for basic voice service, usersmay bypass the normal POTS speech coding methodsand supply their own much better algorithms, such asthose standardized by Moving Pictures Expert Group(MPEG). MPEG is an organization involved in stand-ardizing audio coding. Another characteristic of ISDNimportant to broadcasters is that the B channels aretrue full-duplex, with absolutely no cross-connectionbetween the send and receive signal paths.

ISDN is now widely available and is growing inpopularity—mostly because of its value for high-speedInternet connectivity. Web surfers may implement di-rect digital links without the bottleneck caused by inef-ficient, slow modems. An ISDN BRI has 128 kbpsraw capacity. Compare this to the speed possible witha 33.6 kbps modem and it becomes evident why thepromise of ISDN creates so much excitement amongpeople who need fast access to the net.

With a BRI line, you get two 64 kbps voice or datachannels, called “B” or bearer channels, and one 16kbps “D” or data channel on a single telephone pair(see Figure 3.10-9). The D data channel is the pathbetween the central office and terminal equipment thatis used for call set-up and status communication andis usually not available to the user.

The S and U InterfacesThe line from the central office is a single copper pair

physically identical to a POTS line. When it arrives atthe subscriber, this is called the “U” interface. The Uinterface converts to an S/T interface with a smallbox called an “NT-1.” In the United States, NT-1functionality is usually included in the terminal equip-ment. In Europe, the telephone company provides theNT-1. Only one NT-1 may be connected to a U inter-face, but as many as eight terminals may be paralleledonto an S bus.

Professional equipment should usually provide ac-cess to the S interface, making it possible to parallelmultiple terminals. You can use either an externalNT-1, or the equipment may have an internal NT-1with both U and S/T connectors.

Terminal AdaptersA terminal adapter (TA) is the equipment that inter-

faces to the ISDN line, providing call set up and proto-

Figure 3.10-9. ISDN termination.

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digits of the DN are not included somewhere withinthe SPID. When DNs are required, only the last sevendigits need be entered.

Digital Long-DistanceLong-distance connectivity is routinely available in

most parts of the United States from the big-threecarriers: AT&T, Sprint and MCI. The “dial 1̀ ” de-fault carrier may be chosen at the time you order theline, just as with traditional voice lines. Also, just aswith voice lines, you may usually choose a carrieron a per call basis by prefixing the number with the1010XXX carrier selection code. You must dial thefull number, including the 1 or 011̀ country codefollowing the prefix.

Here is a hot tip: You can save a lot of money byarranging a special plan with your long-distance (LD)carrier. When you use 1̀ dialing without contactingyour LD carrier, you are generally put into a standardrate plan that has the highest cost of any of the pric-ing tiers.

Some long-distance connections are limited to 56kbps/channel. This arises from a quirk of the oldertelephone infrastructure. The channel banks that havebeen widely employed in the long-distance networkhave a native 64 kbps capability but rob the low orderPCM bit on every sixth frame in order to conveysupervision information (on-hook/off-hook and dialpulses). This limitation is becoming more rare asequipment is upgraded, but there is no way to knowfor sure in advance.

CSD and CSVRecall that each ISDN BRI has two possible B chan-

nels. It is possible to order a line with one or both ofthe B channels enabled, and each may be enabled forvoice and/or data use. Phone terminology for this classof service is circuit switched voice (CSV) and circuitswitched data (CSD). (Both are in contrast to packetswitched data (PSD) which is possible but irrelevantto this discussion.)

CSV is for standard voice phone service and allowsISDN to interwork with analog phone lines and phones.CSD is required for MPEG codec connections. Eventhough you may be sending voice, the codec bit streamoutput looks like computer data to the phone network.

Even for MPEG codec applications, you may wantPOTS speech capability, since some support this fea-ture. Therefore, you may want to order CSV as wellas CSD on one or both B channels. To get a line withone B channel to be used with either hi-fi or speech,you would request an ISDN BRI 1B̀D line withCSV/CSD capability. For both B channels, you wouldorder an ISDN BRI 2B̀ D line with CSV/CSD onboth channels; if you do not need voice possibility onthe channels, you want 2B̀D with only CSD enabled.

ProtocolsIn a perfect world, all ISDN terminal equipment

would work with all ISDN lines, without regard for

such arcana as 5ESS, DMS100, CSV/CSD, SPIDs,etc. Unfortunately, the ISDN standard has beenevolving for years and has only recently begun to settledown. And, sadly, there will remain different standardsfor the Unites States and Europe.

The telco network and the TA communicate via aprotocol—the language the user equipment and thetelephone network use to converse (on the D channel)for setting up calls and the like. This is where youwill find differences, since the protocol depends uponthe central office equipment and the standards thatit follows.

In the United States, telephone companies use eitherAT&T 5ESS, Northern Telecom DMS100, or SiemensEWSD switches. Each of these can support the Na-tional I-1 protocol standard, which has been specifiedby Bellcore. However, both AT&T and Northern Tele-com had versions of ISDN which pre-date the NI-1standard and some switches have not been upgradedto the new format. There is also a newer NI-2 standard,but it is designed to be compatible with NI-1 for allof the basic functions.

In Europe, the common protocol is Euro-ISDN, fol-lowing the ETS300 standards. It is an apparently suc-cessful attempt at having all of the European telephonenetworks use a single, compatible protocol. The telcoauthorities in most countries have adopted it already,with most of the rest planning to do so.

T-1 Digital ServiceAs with ISDN, T-1 is possible because an ordinary

copper phone pair can carry a much wider signal thanthe 3.4 kHz required for a single voice conversation.Indeed, a pure metallic path of reasonable length iseasily capable of passing frequencies in excess of 100kHz. Thus, digitization and multiplexing can be usedto carry a number of voice channels over a single pairof wires.

Introduction to T-1To create the T-1 bit stream, 24 64 kbps DS-0 chan-

nels are assembled serially and the equivalent of an-other 8 kbps channel is added for synchronization (seeFigure 3.10-10). Thus the ultimate data rate becomes1.544 mbps, a rate also called DS-1. The signal is thenconverted into a digital bipolar bit stream in a specialformat called binary 8-zeroes suppression (B8ZS). Thevoltage is modulated between13 V and`3 V.

Most LD carriers offer service on T-1 connecteddirectly to their point of presence (POP). Because theLD carrier does not have to pay the usual fee to thelocal telco for routing over their CO and lines, thecustomer cost can be lower.

Figure 3.10-10. T-1 bit stream. 24 audio channels are transmit-ted sequentially.

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direct digital connection into the telco network. In thiscase, no multiplexer and channel cards are necessary,because the connection is made directly to the CSU/DSU. Some PBX equipment even incorporates theDSU.

T-1 and the Broadcast InterfaceGenerally, T-1 service appears to be a good idea

for broadcasters, and many stations are using it suc-cessfully. However, be aware that some T-1 terminalequipment has problems in its analog conversion sec-tion, which cause the on-air hybrid interface to workvery poorly with bad cancellation the result. Also keepin mind that, since all of your service will depend upona single set of circuits, reliability could be reducedcompared to individual analog lines. Consider havingback-up circuits in place.

Primary Rate ISDN (PRI)Primary rate ISDN has a data rate equivalent to

T-1 circuits, providing 23B̀ D, or 23 64 kbps bearerchannels and a 64 kbps D channel for control. (InEurope, PRIs have 31 bearer channels.) It is expectedto replace T-1 eventually, since it speeds dialing andoffers superior monitoring capabilities.

ADSLAsymmetric digital subscriber lines (ADSL) prom-

ise connections at speeds of up to 3 mbps in the direc-tion from the CO to the user. The upstream speed islimited to some much smaller value which is where theasymmetric part of the name comes from. An importantadvantage is the cost; it appears that this service maybe priced at around the same level as ISDN BRI.

Initially, this technology was viewed by the telcoindustry as a way to compete with cable TV for thedelivery of video services. Combined with an MPEGvideo/audio encoder, the bit rate offered by ADSLwould permit full-quality National Television SystemCommittee (NTSC) television. These projects now ap-

Using T-1: The Customer Provided Equipment(CPE)

Despite the difference in capacity and service, T-1arrives at the end user site as two conventional copperpairs: one for the data send and another for receive.The physical connector used to be a DB-15 type, butthe current standard is the common RJ-48C, an 8-position modular plug. Figure 3.10-11 shows bothtypes.

Here are the usual components of a terminal systemfor a T-1 circuit:

• The CSU and DSU.The T-1 line is first connectedto a piece of equipment called the channel serviceunit (CSU). The CSU used to be considered part ofthe network, but is now almost always customer-provided and may also be merely included as anadjunct section in a complete T-1 interface solution.The CSU contains the last signal regenerator as wellas a number of testing and maintenance featuressuch as provision for loopback testing by the centraloffice. It may also include a system to collect andreport error statistics. The data service unit (DSU)handles the remaining digital housekeeping func-tions and data conversion from the bipolar T-1 formatto standard serial data

• The Multiplexer and Channel Cards.The multi-plexer, sometimes called achannel bank,is wherethe multiple voice (or data) channels are combinedinto the single bit stream required for T-1 transmis-sion. Each voice channel is converted to and fromdigital using codecs. In order to simulate typicaltelco lines, talk battery is added, ringing voltage isgenerated and loop current is detected. Generally,multiplexers are constructed using a modular circuitcard approach so that the available digital bandwidthmay be configured as desired.

Many modern PBX systems and at least one broad-cast on-air system are able to accept T-1 lines directly.This is a near ideal approach, since you get a low cost

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Figure 3.10-11. T-1 connector pin-out. Either DB-15 or RJ48-C modular connectors may be used.


pear to be stalled and current efforts are being focusedon high speed Internet connectivity. Since the Internetis a packet-based system with no bandwidth guarantee,the utility of this service for broadcast audio transmis-sion is unclear.

CentrexThis service goes by various names, but the consis-

tent principle is that the telco’s CO equipment replacescustomer-owned PBXs. Each phone set has a directconnection to the CO. The idea is to eliminate customerup-front costs and transfer maintenance responsibilityto the telco. Varying requirements for numbers of linesor phones can be accommodated without customerequipment upgrades. Centrex is declining in popularitybut seems to remain popular with universities.

Features in Centrex rely upon flashing theswitchhook and the use of the normal dialpad keys,generally an awkward and confusing situation for us-ers. This problem may be solved with ISDN Centrex,as this permits very sophisticated phones to be usedwith all of the usual PBX features.

Cellular TelephoneCellular extends the dial-up network to many places

where a wire connection would not be considered prac-tical. Cellular transceivers operate in the 800 MHzrange and automatically select the appropriate fre-quency from among the 666 FM channels assigned forthis service. Low power is used so that the frequenciescan be re-used in adjacent areas. The mobile phonevaries its power according to the level of signal re-ceived at the base location. A useful feature for on-air use of a cellular phone is the signal strength meterprovided on some units. Some phones also allow youto see the send power value. Often, the antenna’s pat-tern is quite directional due to its position on the vehi-cle, so moving around while observing the level indica-tion can help make remotes sound better. For fixedremotes, a Yagi antenna can be used with its benefitsof higher gain and directionality. At 800 MHz, Yagisare very compact.

Most equipment designed for use with wired phonelines can be connected to cellular phones using anadapter provided by the phone manufacturer. Intendedfor laptop computer modems and portable fax ma-chines, these adapters provide an interface to anybroadcast equipment that can connect to a phone line.Units especially designed for broadcast use have provi-sions for audio input and output for direct connectionto microphone mixers and the like.

Some new digital cellular systems have the capabil-ity to transfer data via a special interface. Unfortu-nately, the bit rate is limited to only 14.4 kbps—notsufficiently fast for digitized audio. The impetus fromthe Internet may cause cellular vendors to offer higherbit rate phones in the future, permitting broadcastersto use them for high-fidelity remotes.

A downside of the new digital phones is that speechquality may be poor. This results from the very low

bit rate used by these systems and the extreme com-pression methods that are required to shoehorn audiointo the channel.

FCC RegulationsFCC requirements for connecting equipment to

phone lines are outlined in Title 47 of the Code ofFederal Regulations (CFR), Part 68: Connection ofTerminal Equipment to the Telephone Network. TheCFR can be ordered from the Government PrintingOffice.

PBX AND KEY SYSTEMS

Now that we know a bit about the nature of the phonenetwork, we can explore what happens after the linesbecome ours. We will want to use some of what thephone people refer to as CPE. That is all of the equip-ment connected to the phone line after the officialdemarcation point. We will survey the various stylesof PBX systems available both for general office andon-air use, followed by a look at systems designedspecifically for studio application.

Private branch exchanges (PBXs) are found wherethere is a need for a large number of extension phones.PBXs are miniature central office exchanges, allowinglocal phones to call each other as well as access trunklines for incoming and outgoing calls. PBX systemsoften have a number of specialized features for callrouting and control. Traditionally, PBX systems haveused only single-line phone sets as terminals, withspecial functions like transferring and conferencingaccessible by flashing the switch hook or by usingthe tone pad in a special way. Most PBXs now haveavailable feature phones, which can button-access indi-vidual lines as well as provide numerous other ad-vanced functions. Sometimes these systems are calledkey systems after the old multi-key 1A2 phones. (Whyphone engineers called buttons keys remains amystery.)

Modern Telephone SystemsWhile the systems are tremendously varied, most

have in common that the cable from each phone setto the common equipment conveys:

• Power to operate the phone• A two-way data path to signal user actions from the

set to the switch and operational and display statusfrom the switch to the set

• The speech audio.

Here are the usual approaches phone manufacturersemploy for wiring and communication:

• All Digital. The most advanced systems use a puredigital bit stream for both voice and data. The phoneset contains the codec for conversion to-and-fromthe analog and digital domains. The pure digitalapproach is used in the AT&T System 85, in the

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line switching clunk is not muted, although this is nota problem when calls are not aired directly and sequen-tially.

Another potential problem is audio quality. The pri-mary impediment is usually noise, most often the resultof the data signals cross-talking into the audio. Buzzfrom the power supply sometimes finds its way intothe audio. Often, frequency response is limited by toosmall line coupling transformers or from other causes.Poorly designed digital systems may suffer from quan-tization and aliasing noise and distortion. Few PBXmanufacturers publish specs on audio performance.Since, clearly, this is of importance to those of us whoneed to get decent quality from phones for on-air use,we’ll want to make sure that the audio is at leastreasonable. When choosing a new PBX, ask the phonesystem dealer for audio performance data or arrangeto conduct at least a few simple tests yourself.

Direct Connection to the Skinny WireWhen the phone system uses the separate-pair ap-

proach previously described, the center two wires onthe modular plug are usually the audio path. Since thephone’s control functions stay active even when theseconnections are broken, it is possible to intercept theaudio signal here for feed to the interface. Most broad-cast interfaces provide a loop-through connection,which feeds the phone line back out when it’s notactive. Thus, the unit may be series connected withthe audio pair. That way, you have normal telephonefunction preserved when the interface is not in use.When the interface is active, the phone serves merelyas a controller, with no audio reaching the phone’snetwork or handset. Wiring the hybrid’s on/off func-tions to the console’s switching logic accomplishesautomatic operation.

When the phone uses the two-pair phantom ap-proach previously described, the audio is again likelyto be present on the inner pair and may be interceptedfor interfacing use if the dc connection is maintained.One way to do this is to provide a bypass for dc withinductors. TwomH has proven acceptable in experi-ments performed on some phone systems As shownin Figure 3.10-12.

Northern Telecom Meridian family, in the newerMitel systems with theSuperset DNphones and inthe digital version of the NEC NEAX, among manyothers. The SiemensOffice Pointsystem claims touse standard ISDN protocol between the sets andthe common equipment

• Separate Pair per Function.The early electronicphones used a separate pair for each of the threefunctions, and thus required three (or more) pairs.The AT&T Merlin system used this design. Thecenter pair is the audio; another pair is for the seriallytransmitted control and display data and another han-dles the phone’s power requirements

• Two-Pair, Phantom Power.This used to be the mostcommon approach, but is now fading, as pure digitaldesigns have become cost-effective. The AT&TSpirit system the popular NEC and TIE systems andmany others use this approach. Talk and data eachuse one of the two pairs. The power is applied be-tween the two pairs similar to the method used forphantom powering condenser microphones in re-cording studios. A transformer at each end of theaudio pair permits the phantom power to be added.The data pair will probably use resistors to obtain acenter tap, rather than transformers since the datasignal has a dc component which could not passthrough a transformer.

• Two-Pair, Power not Phantom.Some two-pair sys-tems put the data on one pair and the audio on theother. Power may be on the data pair or on the audiopair. In the latter case, the audio pair resembles acentral office line so that the phone ports may beuniversal: either single-line sets or feature phonescan be plugged-in without hardware changes in thePBX. At least one of the Panasonic systems usesthis technique. The center pair, again, is generallythe audio

• Data Over Voice.The analog MitelSupersetphonesuse a unique scheme that requires only one pairfor all three functions. The data is amplitude shiftmodulated onto a 32 kHz carrierover voiceand thenthe combined voice and data are ac coupled acrossthe dc power voltage.

Interfacing to PBX PhonesIt is usually possible to interface to PBX phones for

on-air use. However, this is best reserved for casualphone use such as for the occasional request or contestwinner call. For applications where phone calls are asignificant programming element, it is usually betterto consider the specialized on-air systems from thebroadcast-oriented manufacturers.

One reason is that the hybrid interface cannot deter-mine when a new call is selected, so it can not adjustits null to the new line before the conversation starts.(However, since the hybrid can null on voice duringconversation, null will be achieved in perhaps fourseconds. This is acceptable if only a portion of thecall is to be aired, as is common with on-air requests,contest winner calls and the like.) Another shortcomingof the direct-to-electronic phone approach is that the

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Figure 3.10-12. If the center two wires on many electronic phonesystems convey audio, they may be used to feed broadcast equip-ment. The inductors bypass power to the phone set when thestudio interface is active.


Special System Ports: Faux CO LinesSince fax machines and modems need connections

that look like central office lines, many systems pro-vide ports for this use. They may be connected tobroadcast interfacing gear as if they were CO lines.Sophisticated PBXs have programming features thatallow these ports to be configured in various and poten-tially useful ways. For example, they may be set upfor private line ringing (when a given incoming COline rings, the call may be directly sent to the selectedport). Unfortunately, with most PBX systems, awk-ward operation may result, since the only way to movea call from a phone set to the port may be to transferit using multiple button punches, rather than the usualsimple place-on-hold-and-pick-up-elsewhere opera-tion. Taking calls in sequence on-air may be extremelydifficult or impossible. Figure 3.1013 illustrates onepossible solution.

Speakerphone Tap-OffOne way to get low cost interfacing is to take advan-

tage of the switching-type interface that many phoneset internal speakerphones provide. The procedure isto tap off the speaker with a transformer and pad tothe console’s required input level. You may continueto use the phone’s internal microphone or you canprovide an external send audio source to substitutefor the phone’s internal microphone. Again, you willcertainly need a pad and probably a transformer. Theinput feed must be set so that appropriate switchingaction and proper send levels are obtained.

Handset AdaptersAdapters are available that plug into the phone set’s

handset modular jack and convert the microphone andearpiece signals into a signal that emulates a standardCO line. While useful in some applications, this ap-proach is likely to offer a lower quality feed because

the phone set’s network remains in the signal pathcausing impedance bumps and other problems.

Intercepting the Serial Data StreamWhy can’t we just emulate an electronic phone set

by generating and decoding the phone system’s serialdata? It does seem that this would be a good solution.However, phone system manufacturers insist on keep-ing their data protocols a deep secret. That means thatbroadcast manufacturers are unable to design directemulation equipment. Of course, even if we had theprotocols, there is the problem of accommodating thedozens of communication methods employed byPBX designers.

1A2 Key SystemsWhile nearly all stations have gone to high-tech

PBXs for the business office, many on-air installationscontinue to rely upon 1A2 key systems. Key systemsoffer the advantage of providing a direct metallic con-nection to the CO line. That means that no frequencyresponse error, noise, distortion or time delay is intro-duced. Often, these issues are not fully considered inthe design of the more complex business phone sys-tems. In addition, costs are favorable, and full schemat-ics and other documentation are readily available.

Leading from the key service unit (KSU) to eachphone is a thick cable with 50 conductors (25 pairs).The tip/ring pair carries the telephone audio. As men-tioned, these are direct connections to the telco COlines. TheA leadstell the key system which lines arein use and also signal a hold condition. Selecting aline causes a connection to be made in the phone setfrom the A lead to another wire, theA-common.TheA lead is normally at124 Vdc and A-common is atground potential, so when a line is selected, the A leadgoes from124 Vdc to ground. If the A lead is brokenbefore the tip/ring is disconnected, the system puts the

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Figure 3.10-13. One way to integrate the on-air system with the station business phone system. Ports intended for single-line phonesets are used as input to the on-air system.


Computer Telephony Integration (CTI)With such a system, the PBX manufacturer pro-

vides complete documentation on an interface thatcan provide control of all of the important aspectsof phone switching, including call set-up and routingfunctions. A standard data port is provided so that out-side vendors may supply systems to work in concertwith the phone equipment. These open PBXs mayeventually offer a universal method for broadcastequipment to coordinate with the station’s officephone system.

Another approach is to build a PBX using spec-ial cards and software installed in a standard PC.Systems of this type would use the Windows NT oper-ating system along with other standard PC software

line on hold. Thelamp-leadslight the phone’s linebuttons with 10 Vac from the KSU’s power supplyand are returned via thelamp grounds.The standardcolor codes and pinout are given in Table 3.10-2.

The Evolving PhoneAs time goes on, probably all but the most inexpen-

sive systems will use the purely digital approach. Aswe’ve seen, these systems are difficult to interfaceto, but perhaps over time protocols will become stand-ardized and maybe even based on ISDN. If this hap-pens, broadcast interface manufacturers may be ableto provide equipment that could directly connectto the PBX in place of, or in series with, the studiophone set.

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Table 3.10-2

Telephone color code and 1A2 key system assignments. The pin numbers indicated are forthe Amphenol “Blue Ribbon” connectors used to terminate 25-pair cables.

Pin # Wire Color 9 Line 1A2 5 Line 1A2

26 WHITE/BLUE Line 1 tip Line 1 tip1 BLUE/WHITE Line 1 ring Line 1 ring27 WHITE/ORANGE Line 1 A Line 1 A2 ORANGE/WHITE A circuit common(gnd) A circuit common(gnd)28 WHITE/GREEN Line 1 lamp ground Line 1 lamp ground3 GREEN/WHITE Line 1 lamp Line 1 lamp29 WHITE/BROWN Line 2 tip Line 2 tip4 BROWN/WHITE Line 2 ring Line 2 ring30 WHITE/SLATE Line 2 A Line 2 A5 SLATE/WHITE Line 9 A A circuit common(gnd)31 RED/BLUE Line 2 lamp ground Line 2 lamp ground6 BLUE/RED Line 2 lamp Line 2 lamp32 RED/ORANGE Line 3 tip Line 3 tip7 ORANGE/RED Line 3 ring Line 3 ring33 RED/GREEN Line 3 A Line 3 A8 GREEN/RED Line 8 A A circuit common(gnd)34 RED/BROWN Line 3 lamp ground Line 3 lamp ground9 BROWN/RED Line 3 lamp Line 3 lamp35 RED/SLATE Line 4 tip Line 4 tip10 SLATE/RED Line 4 ring Line 4 ring36 BLACK/BLUE Line 4 A Line 4 A11 BLUE/BLACK Line 7 A A circuit common(gnd)37 BLACK/ORANGE Line 4 lamp ground Line 4 lamp ground12 ORANGE/BLACK Line 4 lamp Line 4 lamp38 BLACK/GREEN Line 5 tip Line 5 tip13 GREEN/BLACK Line 5 ring Line 5 ring39 BLACK/BROWN Line 5 A Line 5 A14 BROWN/BLACK Line 6 A A circuit common(gnd)40 BLACK/SLATE Line 5 lamp ground Line 5 lam ground15 SLATE/BLACK Line 5 lamp Line 5 lamp41 YELLOW/BLUE Line 6 tip16 BLUE/YELLOW Line 6 ring42 YELLOW/ORANGE BL, AG, or spare BL, AG, or spare17 ORANGE/YELLOW SG, LK, or spare SG, LK, or spare43 YELLOW/GREEN Line 6 lamp ground18 GREEN/YELLOW Line 6 lamp44 YELLOW/BROWN Line 7 tip19 BROWN/YELLOW Line 7 ring45 YELLOW/SLATE B or B1 B or B120 SLATE/YELLOW R or R1 R or R146 VIOLET/BLUE Line 7 lamp ground21 BLUE/VIOLET Line 7 lamp47 VIOLET/ORANGE Line 8 tip22 ORANGE/VIOLET Line 8 ring48 VIOLET/GREEN Line 9 lamp ground23 GREEN/VIOLET Line 9 lamp49 VIOLET/BROWN Line 8 lamp ground24 BROWN/VIOLET Line 8 lamp50 VIOLET/SLATE Line 9 tip25 SLATE/VIOLET Line 9 ring


components such as a database server to provide avery sophisticated package of features. It is possible,for instance, to dial using database name look-up ona networked PC. Ironically, most CTI systems useanalog phone sets.

BROADCAST INTERFACING

This section describes the techniques necessary toachieve the best possible result from the phone-to-broadcast shotgun marriage.

One-way InterfacingThere is often a need to take audio from a phone

or broadcast in only one direction at a time (newsroomphoners are a common application). If there is norequirement for a two-way conversation, a simple in-terface using a QKT will do. Formerly available fromthe phone company, this small box was permanentlywired into a phone instrument or line and provided aquarter-inch (12.7 mm) phone jack output for feedinga line-level signal to a console or recorder input.

Since the QKT is nothing more than a transformer,a capacitor and a zener diode limiter, you can makeyour own (see Figure 3.10-14). The capacitor providesdc blocking so that the transformer does not becomesaturated with the phone line’s dc potential. In orderfor the coupler to hold the line by drawing loop current,eliminate the capacitor and use a transformer that canwithstand the loop current without producing distor-tion. (One such a transformer is the SPT117 from PremMagnetics.) When sending audio into the phone line,remember audio level should be limited to19 dBm.The QKT had back-to-back zeners for this purpose;you may want to add them to your homemade interfaceif you expect audio levels to get out of hand. Of course,commercial units are available that are a little fancierthan the simple device described here. Some offerauto-answer and disconnect capability.

When using a coupler, it is most convenient to havethe telephone instrument on-line and equipped with apush-to-talk switch on its receiver. This is because thephone’s receiver has to be off-hook while a feed iscoming in; the switch turns off the receiver’s mouth-piece microphone when it is not depressed, thus insur-ing that noise from the studio side will not be includedin the recording. Since this coupler works in bothdirections, it can be used to send audio down the phoneas well—useful in the production studio for letting

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Figure 3.10-14. Simple one-way-at-a-time interface. The capacitoris for dc isolation and is not required when a transformer whichcan sink loop current is used. The zeners are chosen to properlylimit transmission levels to the required 19 dBm.

Figure 3.10-15. Switching interface allows two-way conversation,but only one way at a time.

clients hear their commercial masterpieces before theygo into the control room.

When hooking up to a multi-line phone, connect toa point where the tip/ring is present after line selection.The most convenient place is usually right at the phonenetwork. Use headphones to find the spot.

Two-way InterfacingThe simple coupler’s limitations become apparent

when it is necessary for the caller to hear the announcerand the audience to hear the caller simultaneously. Amore sophisticated method is needed because of therequirement to have isolated send and receive audiosignals.

SwitchingThis is what you get when you connect a speak-

erphone to your console input. No commercial broad-cast interface uses this technique, which uses gainswitching to keep the send audio from appearing atthe receive output. Two electronic switches or voltagecontrolled amplifiers are used in such a way as toensure that either the send or the receive path is closedat any given time, but never both simultaneously (seeFigure 3.10-15). A decision circuit compares the sendand receive levels, with the direction of transmissionbeing determined by the relative signal strengths.

The disadvantage of the switching technique is itsuni-directional nature. The caller cannot be heard whilethe announcer is speaking, and noises in the studio cansometimes cause a caller to disappear momentarily,especially on weak calls.

The HybridHybrids were invented long ago to separate the send

and receive signals from the common two-way phonepair. Early hybrids were made from transformers withmultiple windings. Nowadays, most hybrids are madewith active components and are known asactive hy-brids. Both circuit types use the same principle andachieve the same effect.

In Figure 3.10-16, the first op-amp is simply a buffer.The second is used as a differential amplifier; the twoinputs are added out-of-phase (subtracted). If the phonelines and thebalancing networkhave identical charac-teristics, then the send signals at the second differentialamp will be identical, and no send audio will appearat the output.


the switches being set to match the network to a partic-ular line.

Broadcast Hybrid ApplicationIn broadcast application, the studio mixing console

combines the output of the hybrid and the announcer’smicrophone audio, as illustrated in Figure 3.10-17. Asdiscussed previously, the hybrid output consists of boththe desired caller audio and the undesired leakage—(the announcer audio butphase-shifted because of thephone line’s reactance). If the amount of leakage istoo great and the phase shift too extreme, the announcersound will suffer degradation as the original and leak-age audio combine in and out of phase at the variousaffected frequencies. When this occurs, the announcersounds either hollow or tinny as the phase cancellationaffects some frequencies more than others. Anothereffect of too little transhybrid loss is that feedback canresult from the acoustic coupling created when callersmust be heard on an open loudspeaker. Yet anotherproblem can occur when lines are to be conferenced;when the gain around the loop of the multiple hybridsis greater than unity, feedback singing will be audible.So a hybrid will be useful for broadcast only whenleakage is kept acceptably and consistently low.

The plots of phone line impedance vs. frequencyand phase shift shown in Figure 3.10-18 are the resultof measurements performed on phone lines at a radiostation in the Midwest. They indicate the wide varia-tion seen on typical telco lines as provided to broad-casters. The lines with smooth curves have impedancecharacteristics that could be emulated with a simpleresistor-capacitor (RC) combination. These lineswould work fairly well with a simple hybrid, sincean RC balance network would match the impedancecharacteristic closely enough to make the cancellationof send audio at the hybrid output good enough toprevent coloration of the announcer audio.

The balancing network is a circuit consisting ofcapacitance, resistance and sometimes inductance,forming an impedance network. Depending on the hy-brid’s application, this circuit can be very simple orit can be comprised of a large number of componentsand have a very complex impedance characteristic.

R1 and the phone line form a voltage divider, asdoes R2 and the balancing network. If the phone lineand balancing network are pure resistances, then,clearly, the phone line and the balancing network musthave the same value in order for the signals at thedifferential amplifier to have the same amplitude andfor complete cancellation to occur.

The phone line, however, is not purely resistive, butrather is complex impedance, causing both the ampli-tude and phase to vary as the send signal frequency var-ies. Two-to-four wire converters, transformers, repeat-ers, T-carrier systems and other telco systems areresponsible for significant impedance bumps. Loadingcoils also usually have a deleterious effect on the perfor-mance of hybrid interfaces since the coils can create res-onant peaks and phase anomalies in the phone line’simpedance curve which are difficult to null out.

Only when the impedance of the balancing networkis the same as the phone line, and the signals at thedifferential amplifier are matched in both amplitudeand phase, will full cancellation of the send signalbe achieved. Otherwise, leakage results—the scourgeof hybrids.

Because the phone company’s requirements are notgenerally too stringent, they usually use a simple net-work with compromise values of resistance and capaci-tance. Their goal is to get an average of about 12 dBrejection, with 6 dB acceptable on difficult lines—justenough to prevent feedback in a system with back-to-back hybrids. When the situation calls for betterperformance, modules with a number of R and C ele-ments that can be switched in or out are employed,

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Figure 3.10-16. Op-amp hybrid. The second op-amp is used as a differential amplifier to perform the required subtraction for nulling.


Those other lines are quite another story! While itwould theoretically be possible to construct a balancenetwork to match the difficult lines, practical consider-ations usually keep this approach from being used.The impedance characteristic required is too difficultto produce using resistors and capacitors. If the hybridis to be switched among a number of lines, the linecharacteristic would have to be consistent from call-to-call and nearly the same impedance curve.

Digital Signal Processing HybridsDigital signal processing(DSP) offers a very power-

ful and effective technology to improve hybrids. DSPis the process of operating on analog signals that havebeen converted into the digital domain. Since the sig-nals are numbers, mathematical operations can be per-

formed to manipulate them before being returned toanalog. Complex processing functions either impracti-cal or impossible to be done with analog circuit ele-ments are achievable in DSP.

With the DSP hybrid, natural simultaneous conver-sation is possible without distortion of the announceraudio. To accomplish this, the announcer and calleraudio signals are digitized and processed in a systemthat makes use of a specialized DSP microprocessor.The digital hybrid incorporates software programmedto perform the hybrid cancellation function. The tech-nique,convolutional least mean square adaptive filter-ing, is capable of very accurate synthesis of the re-quired balancing transfer function for maximumnulling (see Figure 3.10-19). Unlike resistor/capacitoranalog schemes, the adaptive filter can create the com-

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Figure 3.10-17. Block diagram of typical studio arrangement with telephone hybrid. Announcer audio is combined with hybrid output,potentially causing problems with announces voice distortion. The acoustic path is a possible source of audible feedback.

Figure 3.10-18. Impedance vs. frequency curves for some typical phone lines.


single frequency, since both phase and amplitude at asingle frequency can be adjusted for good cancellation.

Another thing to keep in mind—although the twoare related, the transhybrid loss is not the same as theobserved difference between the caller level and theleakage at the hybrid’s output. That is because thetypical phone call is maybe120 to 125 dBm (onchoke lines, even lower) and the send level (to thecaller) from the hybrid should be110 dBm. Thatmeans that the hybrid has to use up 10–15 dB ofits transhybrid loss just to get even. The remainderbecomes the observed difference.

Other important performance characteristics includeS/N ratio, distortion and (for a digital unit) number ofbits in the audio path. The operation of the dynamicfunctions—the AGC, noise gate and override duck-ing—make a significant contribution to a hybrid’s ef-fective performance.

Combining the Hybrid and Switching TechniquesThis is the method used in nearly all commercial

interfaces. The hybrid produces as much send-to-re-ceive isolation as can be achieved. Then a ducking oroverride function causes thedynamicrejection to begreater than the hybrid alone can produce. When sendaudio is present, the receive gain is reduced. Thus,leakage also is minimized. However, since the levelfrom the phone is also reduced when the announceris speaking, there is a sacrifice of full-duplex operation.A user adjustment in the control signal path permitsvariation of the amount of receive ducking, allowingfull duplex operation when the hybrid alone producessufficient rejection, or speakerphone-like operationwhereby the caller is turned almost completely offwhen the announcer speaks. As a practical matter, thiscontrol is usually set to provide the minimum amountof ducking which provides adequate send-to-receiveleakage suppression.

ISDN For Studio Call-In Talk SystemsISDN can provide a direct digital connection to the

POTS analog network, so it can be used to enhancethe quality of on-air calls. A call set-up message issent from the customer equipment to the network totell it to switch into POTS interworking mode. (Thisis in contrast to when an ISDN line is used with MPEGcodecs. In that case, the line may be carrying voicesignals but in a format that is incompatible with POTSphones. Instead, the network is providing a transparentend-to-end digital path.)

The cost of ISDN service is not a barrier. WithISDN lines costing about the same as analog in mostparts of the United States. An ISDN BRI, with twochannels, costs about twice as much as a POTS line.(Pricing varies depending on the telco but ranges froma 20% discount to a 30% premium. The average isprobably around a 10% premium.

Broadcast interfaces may use either BRI or PRI. Asimple interface for the newsroom could use a singleBRI. Even sophisticated multiline systems could use

plex multiple break-point impedance vs. frequencycurves required by difficult-to-match phone lines. Thesend and receive signals are constantly compared in afeedback loop with the leakage becoming an errorcontrol signal which drives adjustment of thedigitalbalancing network.

The performance advantage of the digital hybridtechnology is striking. On a typical phone line with afairly smooth impedance curve, an analog hybrid mightattain 15–20 dB transhybrid loss. A digital hybrid willlikely produce 40 dB or better transhybrid loss. Onlines with difficult impedance curves, the analog hy-brid’s performance will usually be so poor as to preventits use, while a digital hybrid would perform ac-ceptably.

When a call is initially established, a brief mute/adaption period provides an opportunity for the systemto adjust to the phone line prior to the call going onair. The caller hears a noisy tone, but none of this toneis heard on the air since the output is muted. This hasthe incidental benefit of removing the line switchingclunk. Adaption continues as the conversation pro-ceeds, using voice as the reference signal.

While in the digital domain, other operations inaddition to the hybrid adaptive balancing can be per-formed. Automatic gain control (AGC) can take advan-tage of digital techniques to significantly improve uponthe functions implemented in analog. For instance,cross coupling to the hybrid section is possible in orderto avoid the output AGC, confusing hybrid leakagewith low level caller audio and inappropriately increas-ing gain. AGC may be smartened in other ways, aswell. An adaptive floating expansion threshold, forexample, improves noise-gating quality.

Evaluating Hybrid PerformanceThe amount of hybrid rejection—the transhybrid

loss—directly affects the on-air audio and is the mostcritical measure of hybrid quality. The true test ofhybrid performance is determined by measuring theamount of rejection across the entire audio frequencyrange, preferably with pink noise as a test signal atthe send input. Any hybrid with an adjustable R andC balance network can produce high rejection at a

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Figure 3.10-19. In the DSP hybrid, the digital balancing networkcontinuously adjusts to the phone line impedance characteristic.When the adaptive network transfer function is identical to thatof the phone line, perfect cancellation is achieved. Since theadaptive network is a digital filter than can create almost anyrequired curve, performance is superior to the analog hybridalone.


BRIs, with enough of them to achieve the desirednumber of lines. While PRIs would seem to be a moretechnically appropriate solution for a multiline system,BRIs may be more cost effective, more readily avail-able and able to provide a measure of redundancy. Asystem using PRI or T-1s may be able to share linesamong a number of studios, with connections to bothhybrids and codecs.

ISDN Lines Are Inherently 4-WireAs we have learned, analog lines use a single pair

of wires for both signal directions, mixing the sendand receive audio. This causes the famous leakageproblem—where the announcer’s audio is present onthe interface output, instead of the desired caller onlyaudio. Digital circuits inherently offer independent andseparated signal paths.

While DSP based hybrids applied to the problemof separating the send/receive signals are dramatic im-provement over analog systems, ISDN enables furtherimproved performance. This is because it offers a fullyindependent path for each speech direction. In the casewhere both ends of a connection are digital, there isno mixing whatsoever. In the call-in application, thefar-end from the studio will still be 2-wire, so theaudio paths will not be fully independent and a digitalhybrid function will still be necessary to cancel residualleakage. But moving the studio side connection awayfrom mixed analog can help tremendously because itprovides the hybrid a much better starting point.

Better Digital-Analog Conversion QualityThe codecs used in telephone central offices are not

as good as the converters commonly used in audioequipment. Fidelity is not an important considerationwhen designers choose parts for this function. In aprofessional interface for studio application, we areable to design with much better converters than avail-able in the telco’s equipment. Noise-shaping functionspermit a larger word-length converter to provide sig-nificantly better distortion and S/N performance.

In all digital installations, the phone interface canmaintain a digital path all the way. Audio EngineeringSociety/European Broadcasting Union (AES/EBU)can be provided on the interface to accomplish theconnection to the studio gear.

Lower NoiseAs digital circuits, ISDN lines are not susceptible

to induced noise. Analog lines are exposed to a widevariety of noise and impulse trouble-causers as theymove across town on poles and through your building.Hum is the main one, given the line’s proximity totransformers and ac power lines, but there are alsosources of impulse noise from motors, switches andother sources. Digital lines convey the bits preciselyand accurately from the network to your studio equip-ment without any perturbation—so the audio re-mains clean.

Call Setup and Supervision are BetterAnalog lines use a strange mix of signaling to

convey call status. Loop current drop and returneddialtone signal that a far-end caller has disconnected;blasts of 100 volts at 20 Hz mean someone wantsyou to answer. Why should we be using a mechanismdesigned to bang a gong against a metal bell totransmit network status information in the 1990s?ISDN uses a modern digital approach to controllingcalls and conveying status information about them.The sophisticated transactions on the D channel areable to keep both ends of a call accurately informedabout what is happening.

ISDN call set-up times are often a few tenths ofmilliseconds, enhancing production of a fast-pacedshow. Perhaps more importantly, when a caller discon-nects while waiting on hold, the ISDN channel commu-nicates this status change instantly. This contrasts withthe usual 11 second delay on most analog lines. Oneof the most common complaints of talk hosts is thatwhen they go to a line where they expect a caller to bewaiting, they are met instead with a blaring, annoyingdialtone. The chance of this happening with an ISDNline is reduced to near zero.

Another common error is when a talent goes topunch up a line that looks free, but is actually justabout to begin ringing and connects to a surprisedcaller. This condition results from the delay in the ringsignaling, which comes from the nature of the analogline’s ringing cadence. This is much less likely withISDN because the ambiguous status period is elimi-nated.

LevelsISDN does not have the FCC-mandated19 dBm

send level limit. Audio may be adjusted to fill upthe digital word, resulting in higher send signalvolume.

Reduced Feedback During Multi-lineConferencing

When conferencing is required on 2-wire circuits,very good hybrids are needed to separate the two audiopaths to add gain in each direction. When the gainaround the loop exceeds unity, there is the possibilityof feedback singing. Since the conference path usuallyincludes four AGC functions, the hybrid must be suf-ficiently good to cover the additional gain that maybe dynamically inserted. Because of the 4-wire natureof ISDN, the hybrid function is more effective andmore reliably so across a variety of calls. That meansmore gain can be inserted between calls before feed-back becomes a problem.

Line MonitoringSince there is a full-time connection between the

central office and the terminal on the D channel, it ispossible to detect when a line is a not working. On ananalog line, one discovers a problem only from a failedattempt to use the line.

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matic control over a caller who wants to carry on. Thisis a matter of taste; some talent and programmers preferno ducking so that hosts and callers can conduct heatedexchanges without impediment, while others want toexercise control.

Caller ID. ISDN naturally conveys caller ID infor-mation. This is transmitted instantly in the setup mes-sage and is much faster than the 1200-baud modemmethod used in analog caller ID.

Conference Linking. With two B channels avail-able on one BRI line, broadcast interfaces will bedual units, making possible high quality conferencingbetween the two potential callers. Some systems willprobably support larger numbers of conferencedcallers.

BROADCAST ON-AIR SYSTEMS

With phones an important part of programming atmany stations, systems to enable convenient, highquality on-air integration of phone conversations areessential.

On air phone systems are specifically designed foruse in the broadcast studio environment. While manybusiness phone systems offer similar functions—lineselection and status indication, conferencing—they aregenerally awkward to operate in an on-air environmentand may have other limitations such as the audio qual-ity flaws described earlier.

While the phone network would not be consideredto be a high fidelity source, it clearly does not help todegrade it further by adding additional noise, distortionor frequency response impediments. For that reason,broadcast phone systems are designed with these issuesand other specialized requirements in mind. For exam-ple, a broadcast phone system output should be freeof inappropriate switching sounds, and air talent shouldbe able to access and manipulate lines live withoutany pops or clunks being audible to listeners.

Ergonomic RequirementsLine selection and other functions must be per-

formed intuitively and with a minimum of hassle. Un-like a telephone set, broadcast line selection panelshave large illuminated buttons. To avoid operator con-fusion, features are limited to those necessary for on-air application. One such example is panels that dropinto an open position in the studio mixing console sothat the line selection buttons are located near thechannel on/off, fader, and audio switching functions.

Conferencing CapabilityMost broadcast systems allow any number of lines

to be switched to air, even if only a single hybrid ispresent. But, unless you are blessed with excellentphone lines, you will want additional hybrids witheach connected to the other through a multiple mix-minus arrangement. That way, it will be possible tohave amplification between callers. Without multiplehybrids, callers might have difficulty hearing each

The ISDN Broadcast InterfaceMost of the functions performed by an ISDN inter-

face are similar to that of an analog DSP hybrid, butthere are some differences, both in the required func-tions and in the implementation of the common fea-tures.

Send/Receive Separation.This is the traditionalhybrid function provided by broadcast telephone inter-faces. Despite the fact that ISDN lines naturally havetwo independent send and receive paths, it is still nec-essary to provide additional functions to further reduceleakage. The reason is that almost all calls will origi-nate with telephone sets connected via 2-wire analoglines, and so there will still be a mixing of bothspeech directions.

Acoustic Coupling Reduction. There is often anacoustic path between the received caller audio andthe send audio signal. This results from having a loud-speaker in the studio that produces sound that couplesinto the microphones. When the talent use headphonesfor monitoring callers, this is not a problem. But some-times it is not practical to convince guests to wearheadphones, and television stations generally do notwant talk show talent to wear earplugs. In these casesa combination of adaptive cancellation and dynamicgain reduction will reduce the coupling electronically.

High-grade Digital-to-Analog Conversion.Whenan analog connection to studio equipment is required,pro-grade converters can be used to provide muchbetter quality than the usual telco conversion. At mini-mum, 16 bits should be used, but 18–20 bits may notbe excessive.

Sampling-rate Conversion.When the studio con-nection is via a digital AES/EBU channel, no analog-digital conversion is required, but it will be necessaryto adapt the sampling rate of the telephone networkto the studio rate. telco sampling rate is 8 kHz, andstudio equipment will usually operate at 32, 44.1 or48 kHz. A process is required to perform the requiredup-and-down sampling, while suppressing aliasing andreconstruction audio components.

Automatic Gain Control. As with POTS hybrids,this function should be provided on both the send andreceive audio paths. On the send side, it is necessaryto smooth the wide level variations that arise fromusual studio practices. Talent are used to having on-air processing take care of level variations and aregenerally not very careful at riding gain. On the receiveside, AGC is essential to deal with the very differentlevels that can result from the many types of phonesets and telco analog network components.

Dynamic Equalization. With phone sets havinga very wide variety of microphone characteristics, amultiband automatic equalizer helps callers have areasonable spectral consistency.

Caller “Ducking.” As with POTS hybrids, this canserve to reduce residual leakage. However, since ISDNhybrids have much better inherent transhybrid loss, thisfeature will be used mostly to satisfy a programmingaesthetic requirement, reducing the level of the phoneaudio when the host talks and allowing her an auto-

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other, since you are at the mercy of the telco-deliveredline level.

Special FeaturesDesirable features for an on-air phone switching

system include:

Busy/unbusy. To prepare for a contest, all linesmay be busied-out and then returned to readiness afterthe contest has been announced.

Automatic next line selection.Pressing the nextbutton picks up the line that has been holding thelongest. If no line has been holding, the longest ringing-in line is selected.

Call length timer. Displays call duration time.Held caller timer. This tells which line has been

holding longest and for how long.

Integration of On-Air Systems with PBXsTo interconnect the on-air system with the front

office PBX, there are a number of possibilities.Segregate the studio and office phone lines.Ports

from the PBX configured to look like CO lines feedan input or two on the studio system so that that callstaken by the receptionist can be put on the air.

Route all lines through the PBX.The studio linesare programmed in the PBX to be forwarded to theports that feed the studio system. Some audio degrada-tion may result.

Simply parallel the two systems.With no cross-coupling of line status information, there could betrouble if a line is inadvertently picked up on onesystem while the other is being used.

Route the on-air lines through the broadcast sys-tem. Possible if the broadcast system brings out aloop-through connection. This scheme prevents PBXphones from picking up active on-air lines.

Improving Phone Audio QualityWhether extracted from analog or digital lines, due

to its limited frequency response and fairly high distor-tion, the audio from the phone has the poorest qualityof our on-air sources. Thus, it generally pays to maketelephone audio less of an earsore so that it does notstand out more than is necessary from other pro-gram material.

If the phone network is a digital system, why dophones still sometimes sound pretty awful on the air?The main problem is that phone engineers never de-signed the systems with a connection to full fidelitybroadcast systems in mind. The 8/13 bit quantizationscheme used for phone speech coding results in lessthan high fidelity. Often, the problem lies in the specificimplementation rather than in any inherent shortcom-ing in the standard or the technology. One importantquality limitation results from the anti-aliasing andreconstruction filters in the codecs. These filters usu-ally have an ultimate roll-off of around 35 dB. Audioabove the 4 kHz Nyquist frequency will alias andappear in the 300 Hz–3.4 kHz band as distortion. Thus,typical codecs have distortion of 2–3% from aliasing.

The strange raspy noise that seems correlated with thespeech sometimes heard on a telephone circuit is aresult of the effects of this kind of distortion combinedwith audible quantization errors.

Also the codec filters generally use switched-capaci-tor technology, which tends to be fairly noisy. Somenewer codecs avoid the switched-capacitor problemsby employing the samedelta-sigmaover-sampling anddigital decimation concept used for high performancedigital audio conversion, but these are only rarelyfound in telco central office equipment.

What can we do? An ISDN connection solves halfof the problem, since at least one of the telco’s codecsis bypassed. We still have the other end to contendwith, and the majority of broadcast connections willremain analog. Fortunately, there are some remediationpossibilities. Filtering, equalization, gating and dy-namics compression are the primary tools. Most of thecommercial hybrid interfaces have at least some ofthese processes built-in.

FilteringOn a dial-up phone line, there is very little audio

above 3.4 kHz—but there is noise. Thus, a filter witha very steep roll-off above the telephone passband willreduce phone line noise significantly without affectingconversation audio. The low-end can be improved aswell. Low-frequency hum is often a problem—usually60 Hz mixed with its second harmonic, 120 Hz. Thus,it is often a good idea to have a sharp roll-off startingat 200 Hz or so.

EqualizationAn equalizer used to shape the frequency response

of the phone line within its audio bandwidth can resultin marked improvements in perceived quality. A typi-cal phone line has an excess of energy at around 400Hz and considerable roll-off at both the top and bottomends of its passband, so the idea is to compensate byadding gain at both. Boosts at 2.5 kHz and 250 Hzand a cut at 400—500 Hz with a parametric equalizerwill help achieve better sound. Since every phone lineis different, the ear is usually the best instrument toevaluate the results.

When it is not possible or practical to make customadjustments, an adaptive multiband EQ can be an ef-fective tool. The principle is much the same as imple-mented in broadcast transmission processors. Audio isfiltered into multiple bands, and an automatic gainadjustment is performed on each spectral segment.Given the limited frequency range of telephone calls,three bands are sufficient.

Noise GatingAnother effective processing device is the expander

or noise gate. These devices may be used to reducegain between the words of a conversation, thus makingphone line noise less objectionable. On extremelynoisy lines, however, the gating action can make noisemore distracting by causing it to come and go withthe words. In such cases, it might sound better to

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a feed taken from the main announce microphone maybe all that is necessary. The patch send output availableon many consoles is precisely what you need. In instal-lations where multiple microphones are to be used, acombiner of some sort is required. This may be a smalloutboard mixer or a homemade op-amp summer oreven a resistive combiner. Better consoles offer specialpurpose busses that may be used for mix-minus, oftenwith provision for selective switching of sources intothe phone feed. If you need to modify an older consolethat does not have special buses, a device (made byHenry Engineering) accomplishes the mix-minus bysubtracting the hybrid audio from the console programoutput with a differential amp scheme. This unit gener-ates a mix-minus signal true to its name—all sourcesexcept the phone itself will feed the phone.

Recording Phone CallsSome stations may want to record calls for later

playback. One technique is to have the mix-minus goto one track of a stereo tape machine, while the otherchannel gets the hybrid output with the caller audio.The result is a two-track tape with the announcer andcaller separated. To play back, the console’s inputmode is set to mono; the relative balance, if needbe, can be adjusted upon playback. The productiondepartment can use its tape to facilitate extraction ofcontest squeals.

leave the gate off and let the noise remain present ata constant level. A unit with variable threshold andduck ratio can be adjusted so that the optimum compro-mise may be achieved between the benefit of reducednoise and audibility of the effect.

Dynamics CompressionLevels on phone calls vary widely, and it is not

uncommon to see levels range from140 to near14dBm as calls are switched into a given line. A compres-sor helps to smooth the levels. An AGC that maintainsa constant compression ratio regardless of average gainreduction produces more consistency. Freeze gating isalso important, so that gain does not increase duringcaller speech pauses.

Mix-Minus: Getting the Send Audio FeedThe feed-to-caller signal has come to be referred to

asmix-minus,so called because it is often themix ofall of the console’s active inputsminus the phonehybrid’s output (see Figure 3.10-20). A mix-minusfeed is necessary because the hybrid will create a feed-back path if it is forced to chase its tail. Usually, themix-minus is a mix of only the studio microphones,but it may sometimes include other audio that is to besent to the phone such as contest sound effects fromcart machines.

To create the required signal in simple installations,

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Figure 3.10-20. Simplified studio block diagram shows the mix-minus required for hybrid feed.


Talk-Show Screening SoftwareIn its simplest form this personal computer software

lets a talk show screener/producer communicate to theair talent who’s on the line waiting to talk. It replacesthe paper pieces on the window system employed foryears at many talk stations. The better packages offera number of convenient features: display of liner mes-sages and other information, storage of caller data fordemographic analysis and remote operation viamodem.

An Ethernet or serial port on the broadcast systemcan let the computer display reflect current line status.New software enables laptop computers to extend fullcontrol capability and status display to a remote site,and modern systems even permit this function to beconveyed over the Internet.

ISDN: HI-FI REMOTES ON DIAL-UP LINES

ISDN makes high quality remotes possible with dial-up convenience. Convenient, reasonably priced studio-quality audio from almost anywhere in the world isnow possible. The enabling technologies are digitaltelephone services like ISDN and audio compressionor coding algorithms. Products offering this capabilityhave burst onto the market in the last few years, andbroadcasters have enthusiastically embraced the pro-gramming possibilities created by the new capability.

BROADCAST CODECS

Broadcast codecs have evolved rapidly over the pastfew years. Most are now single-box solutions thatinclude an ISDN TA interface and a number of se-lectable coding algorithms. Some portable units eveninclude a mixer for multiple audio inputs and outputs.Most are full duplex, with provision for transmittingand receiving simultaneously, and most offer theADPCM G.722 and perceptual MPEG Layers 2 and3 coding algorithms. Some offer a feature to allowdialing to POTS phones for low-grade voice communi-cations. State-of-the-art systems include an auto-dialfeature that adjusts the codec section settings, such asbit rate and transmit and receive coding choices, aswell as the numbers for the codec you wish to dial.

In MPEG modes, many codecs permit bidirectionalserial data at 9.6 kbps to be transmitted simultaneouslywith the audio. End-to-end parallel contact-closuresoffered by many codecs may be used to control re-corders and other devices. Since codecs are inherentlydigital devices, it is only natural that AES/EBU digitalinputs and outputs are usually available. Sample rateconversion is generally available on both input andoutput paths.

For many remotes, a receive-side mixer is requiredto combine the mix-minus signal from the studio withthe local audio (seeDealing with Delay,below). Insome cases, you will want to have two outputs: onefor the talent, which can include cueing audio, andanother for the public, who listens to PA loudspeakers.

This is a common feature in portable units that aredesigned to be used in the field. Another valuablefeature for remote applications is an input audio limiterto prevent digital nasties when the program signalpeaks instantaneously, as might be the case on remoteshosted by excitable sports announcers.

The J.52 ProtocolWhile codec manufacturers have been remarkably

successful at making their products inter-operate, it isoften necessary to manually adjust a unit at one endor the other to a compatible mode. The J.52 standardaddresses this problem by including information in thetransmitted bit stream which identifies the details ofthe encoding method being used, allowing the receiverto automatically conform. J.52 also standardizes chan-nel bonding.

ADPCM CodingAdaptive delta pulse code modulation (ADPCM)

pre-dates MPEG perceptual coding. It has been aroundas an international standard the longest and is probablythe most widely used system. ADPCM is much simplerthan the perceptual methods but suffers from pooreraudio performance. It has the benefit of low cost andthe unique advantage of low delay.

The most popular method, G.722, was invented inthe late 1970s and adopted as a standard in 1984 bythe Consultative Committee for International Tele-phony and Telegraphy (CCITT), a division of theUnited Nations. The technique used is Sub-BandADPCM, which achieves data reduction by transmit-ting only the difference between successive samples.G.722 does this in two audio frequency sub-bands:50 Hz–4 kHz and 4 kHz–7 kHz.

G.722 has a frequency response extending to 7 kHzat 56 or 64 kbps. Unless there is no alternative, it shouldbe used only for voice feeds, as music transmitted viaG.722 has a distinct fuzzy quality. It is good alsofor cueing and intercom channels. Only two bits areallocated per sample for audio frequencies above 4kHz—sufficient for conveying the sibilance in voicesignals but not very good for intricate musical sounds.Also, thepredictor modelused to determine the stepsize in the adaptive function is designed only forspeech.

G.722 has the lowest delay of all popular codingmethods, about 20 msec. For this reason, it often usedas a return channel so that the round-trip delay isreduced, even when a higher fidelity method is usedfor the on-air feed.

Statistical Recovery TimingG.722 uses a procedure calledstatistical recovery

timing (SRT) orstatistical framingto lock the decoderto the data stream. This procedure is specified in ANSIstandard T1.306-1989.) The process usually happensinstantaneously but can take up to 30 seconds.

The locking can be sensitive to audio present on theG.722 path, as it relies upon the properties of the audio

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mous layers. In 1997, another algorithm, advancedaudio coding (AAC) was added to the MPEG standard.

Acoustic MaskingAll of the MPEG codecs rely upon the celebrated

acoustic masking principle—an amazing property ofthe human aural perception system. When a tone—called a masker—is presented at a particular fre-quency, we are unable to perceive audio at nearbyfrequencies that are sufficiently low in volume. As aresult, it is not necessary to use precious bits to encodethese inaudible, masked frequencies. In perceptualcoders, a filter bank divides the audio into multiplebands. When audio in a particular band falls belowthe masking threshold, few or no bits are devoted toencoding that signal, resulting in a conservation ofbits that can then be used for the bands where theyare needed.

MPEG Layer 2MPEG Layer 2 is the world’s most popular percep-

tual coding method. It is the preferred choice for appli-cations where greater than 120 kbps/channel is avail-able, such as satellite links and high capacity terrestrialpaths such as Primary ISDN or T1 channels. Layer 2is the method used for satellite television audio andfor many other applications such as hard disk storage.It is also used for European Eureka 147 terrestrialdigital broadcasting.

Layer 2 offers ajoint stereotechnique to improvecoding efficiency with stereo signals. The Layer 2 jointstereo mode uses an intensity coding method. Thisprocess has high coding power and is quite effective;however, it may impair stereo separation on someprogram material as audio above about 3 kHz is com-bined to mono and panned to one of seven positionsacross the stereo stage, at lower bit rates.

MPEG Layer 3MPEG Layer 3 is perfectly matched to the bit rates

available on ISDN BRI lines, permitting full FMbroadcast quality. Full fidelity 15 kHz mono is possibleon a single ISDN B channel and very near CD-quality20 kHz stereo is achievable using both ISDN B chan-nels. Until equipment supporting the new MPEG AACstandard arrives, Layer 3 is the most powerful methodavailable to broadcasters. It is widely supported inbroadcast codec equipment from a number of manufac-turers.

MPEG Layer 3 uses a number of advanced tech-niques to achieve its power:

Psychoacoustic Masking.The audio is divided into576 frequency bands. First, a polyphase filter bankperforms a division into the 32 main bands, whichcorrespond in frequency to those used by the less com-plex Layer 2. Filters are then used to further subdivideeach of the main bands into 18 more. The resultingbandwidth of each sub-band is 27.78 Hz. A 32 kHzsampling rate allows very accurate calculation of themasking threshold values. Sufficient frequency resolu-tion is available to exceed the width of the ear’s critical

bit stream itself. Some audio material and tones canprevent lock from ever happening. Silence is the mostreliable signal for locking, and undistorted voice isusually acceptable. The most common problems arewith sine tones and distorted voice or music signals,in which case, removing or lowering (to around112dB) the audio for a few seconds will generally causelock to occur. In very rare cases, it may be necessaryto disconnect and redial. Other strange effects may beobserved. Tones and noises may be present beforelocking occurs, and some continuous audio tones maycause momentary unlocking.

Perceptual CodingThe broadcast world has been transformed by the

introduction of perceptual audio coding techniques.Applying perceptual coding methods, it is possible topass studio-quality 15 or 20 kHz bandwidth audio overISDN channels.

Demystifying MPEGBy far, the most popular perceptual coders rely upon

techniques developed under the MPEG umbrella.About a decade ago, when the CD had just been intro-duced, the first proposals for audio coding were greetedwith suspicion and disbelief. There was widespreadagreement that it would not be possible to satisfygolden ear listeners while deleting 80% or more of thedigital audio data. In response, the MPEG was formed,and since 1988 the group has been working on thestandardization of high quality low bit rate audio cod-ing. Two standards have been completed: MPEG-1(coding of mono and stereo signals at sampling ratesof 32, 44.1 and 48 kHz) and MPEG-2 (ISO/MPEGIS-11172: coding of 5̀ 1 multi-channel sound signalsand low bit rate coding of mono and stereo audio atsampling rates of 16, 22.05 and 24 kHz). Today almostall agree not only that audio bit rate reduction is ef-fective and useful, but that the MPEG process hasbeen successful at picking the best technology andencouraging compatibility across a wide variety ofequipment.

The MPEG process is open and competitive. A com-mittee of industry representatives and researchers meetto determine goals for target bit rate, quality levels,application areas etc. Interested organizations that havesomething to contribute are invited to submit their bestwork. A careful, double blind listening test series isthen conducted to determine which of the entrant’stechnologies delivers the highest performance. Thesubjective listening evaluations are done at variousvolunteer organizations around the world that haveaccess to both experienced and inexperienced test sub-jects. Broadcasters are the most common participantswith recent test series conducted at the BBC, the CBC,NHK. Finally, results are tabulated, a report is draftedand a standard is issued.

In 1992, this process resulted in the selection ofthree related audio coding methods, each targeted todifferent bit rates and applications. These are the fa-

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bands (100 Hz below 500 Hz; 20% of the center fre-quency at higher frequencies) across the audible spec-trum, resulting in better hiding of noise than wouldotherwise be possible.

Redundancy Reduction.A Huffman coding proc-ess accomplishes redundancy reduction. Values thatappear more frequently are coded with shorter words,whereas values that appear only rarely are coded withlonger words. This results in an overall decrease inthe data rate with no degradation, since it is a losslessreduction scheme.

Bit Reservoir Buffering. Often, there are some crit-ical parts in a piece of music that cannot be encodedat a given data rate without audible noise. These se-quences require a higher data rate to avoid artifacts.Layer 3 uses a short time bit reservoir buffer to addressthat need.

Ancillary Data. The bit reservoir buffer offers aneffective solution for the inclusion of such ancillarydata as text or control signaling. The data is held in aseparate buffer and gated onto the output bit streamusing the bits allocated for the reservoir buffer whenthey are not required for audio.

Joint Stereo. A joint stereo mode different fromthat in Layer 2 permits advantage to be taken from theredundancy in stereo program material. The encoderswitches from discrete L/R to a matrixed L̀R/L1Rmode dynamically, depending upon the program ma-terial.

MPEG AACThe MPEG-2 AAC system is the newest audio

coding method selected by MPEG and become aninternational standard in April 1997. It is a fully state-of-the-art audio compression tool kit that provides per-formance superior to any known approach at bit ratesgreater than 64 kbps and excellent performance rela-tive to the alternatives at bit rates reaching as lowas 16 kbps.

The development of AAC began when researchersbecameconvinced thatsignificant improvementswouldbe possible by abandoning backward compatibility tothe earlier MPEG layers. The idea was to start fresh andtakethebestwork fromtheworld’s leadingaudiocodinglaboratories. Fraunhofer Institute, Dolby, Sony andAT&T were the primary collaborators. The hoped forresult wasInternational Telecommunications Union(ITU)-R indistinguishable qualityat 64 kbps per monochannel.Thiswasa fairlydaunting requirementbecauseit requires that no test item fall below the perceptible,but not annoying threshold in controlled listening tests.Thetest itemsinclude themostdifficult-to-encodeaudioknown to researchers—isolated pitch pipe, harpsichordand glockenspiel recordings, among others. The think-ingwasthat ifacodingsystempasses this requirement, itwill almost certainly perform well with normal programmaterial.Poporwesternclassicalmusic is tremendouslyeasier to encode.

Compared to the previous layers, AAC takes advan-tage of such new tools as temporal noise shaping,backward adaptive linear prediction and enhanced joint

stereo coding techniques. AAC supports a wide rangeof sampling rates (8–96 kHz), bit rates (16–576 kbps)and from one to 48 audio channels.

The AAC system uses a modular approach. An im-plementer may pick and choose among the componenttools to produce a system with appropriate perfor-mance-to-complexity ratios. Three default profileshave been defined, using different combinations of theavailable tools:

Main Profile. Uses all tools except the gain controlmodule. Provides the highest quality for applicationswhere the amount of random accessory memory(RAM) needed is not constrained.

Low-complexity Profile. Deletes the prediction tooland reduces the temporal noise-shaping tool in com-plexity.

Sample-rate Scaleable (SRS) Profile.Adds thegain control tool to the low complexity profile. Allowsthe least complex decoder.

AAC is the first codec system to fulfill the ITU-R/EBU requirements for indistinguishable quality at 128kbps/stereo. It has approximately 100% more codingpower than Layer 2 and 30% more power than theformer MPEG performance leader, Layer 3.

Choosing the Coding Method Most Appropriateto Your Application

The following chart compares some of the importantcharacteristics of G.722, Layer 2 and Layer 3.

One thing that should be apparent from Table 3.10-3

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Table 3.10-3

Audio Coding Comparisons

Layer 3 Layer 2 G.722

Method Perceptual`Huffman Perceptual ADPCM

Audio Freq. 15/20 kHz* 8/10 kHz** 7 kHzResponse/mono

Audio Freq. 15/20 kHz* 20 kHz 7 kHzResponse/stereo

Delay at 32 280 msec — 20 mskHz/mono

Delay at 48 240 msec 150 msec 20 mskHz/mono

Delay at 32 450 ms — 20 mskHz/stereo

Delay at 48 kHz/ 340 ms 220 ms 20 msstereo

20 ms MS Matrix “Intensity —Coding”

ISO Target Bit Rate 64 kbps/channel 128 N/Akbps/channel

Coding “Power” 12:1 6-8:1*** 4:1

Bands 576 32 2

Frequency 42 Hz 750 Hz —Resolution (48 kHz)

* 15 kHz at 32 kHz sample rate; 20 kHz at 48 kHz sample rate.** 8 kHz at a 56 kbps network rate; 10 kHz at 64 kbps.

*** 12:1 in intensity joint stereo mode.


Some of what we do know:

• Some International Consultative Committee for Ra-dio (CCIR) tests have demonstrated that one pass ofLayer 3 at 56/64 kbps can be cascaded with two-five passes of Layer 2 operating at high (112 kbps`/per channel mono; 192 kbps̀joint stereo) bit rateswith good results

• Informal tests at Telos with two passes of Layer 3transmitted via Zephyr codecs have proven success-ful, with listeners noticing no audible degradation—even on difficult CDs

• One user has reported that two passes of Layer 3,followed by one pass of sedat, is acceptable (L3 injoint stereo mode).

SEDATThe goal is to get as much coding headroom as

possible at each stage. This is achieved by:

• Using the most possible bits at each stage—the leastcrunching—and/or

• Using the more powerful coding method of thoseavailable at each stage.

Here is some practical advice:

• Use coders only where necessary. Consider the alter-natives at each stage. With the cost of hard diskcapacity falling, is it really necessary to crunch atthis point?

• Use the maximum bit rate you can afford at eachstage. Hard disk recorders and other studio systemsoften have an option to adjust this. For very criticalwork, remember that some codecs may be used ina mode where a mono program is split over twodigital network channels

• Use Layer 3 or AAC on low bit rate channels.

The staff at Fraunhofer Institute who developed theLayer 3 algorithm have introduced a computer basedperceptual coding analyzer. This device has the poten-tial of making objective measurements a reality andmay help us learn about the effects of cascading withvarious coding methods and bit rates.

Mixed MPEG Layer 2 and Layer 3 Signal ChainsWhat about the case where you will be using L2

and L3 together in a signal chain? It turns out that thetwo methods are nicely complementary.

At low bit rates, Layer 3 gets more signal-to-maskmargin than Layer 2. This is why it performs betterin the low bit rate tests. It accomplishes this by usinga filter bank with more bands, 576 vs. 32. One effectof this is time spread. (More frequency resolution re-quires a longer time window. This is a fundamentalphysical law.) For one or two passes, this is good, asthe ear has masking in both the time and the frequencydomains and L3 naturally exploits this additional di-mension. The downside is that when many stages ofL3 are used at low bit rates, the time spread can become

is the trade-off between delay and audio performance.Layer 3’s excellent audio performance requires a sig-nificant delay, because some of its power comes fromthe ability to analyze the audio over a relatively longperiod. Layer 2 requires the next longest delay, andG.722 has minimal delay.

The most flexible broadcast codecs permit the cod-ing mode for the send and receive paths to be indepen-dently chosen, so the choice may be optimized for thespecific requirement of each direction.

Dealing with DelayAll perceptual coders have too much delay for talent

on remote to hear themselves via a round-trip loop.Therefore, a special mix-minus arrangement is re-quired—exactly the same as has been used with satel-lite linked remotes for years. The principle is this: Theremote talent does not hear himself via the studio cuereturn. Rather, his microphone is mixed locally witha studio feed that has everything but the remote audiothus the mix-minus designation. The announcer getsin his headphones a non-delayed version of himself anda slightly delayed version of all of the studio pieces.

To save money and hassle, callers are usually re-ceived at the studio, rather than at the remote site. Inthis situation, phones need to be fed to the remotetalent so that they can hear and respond to callers, andthe phone callers can hear them. In many cases, theremotes are sufficiently distant that the station cannotbe monitored for the caller feed. Even if it could, theprofanity delay would be a problem, since the talentneeds to hear the phone pre-delay. Instead, the talenthears callers via the return path. As before, this returnis fed with mix-minus: a mix of everything on theprogram bus minus the remote audio.

As for the second half of the equation, the callershear the talent because the remote feed is added to thetelephone mix-minus bus. This should be no problemif you have a setup that permits selective assignmentto the phone mix-minus. The most common problemwith this arrangement is a result of a phone hybrid withtoo much leakage combined with the system delay. Ifthe hybrid isn’t doing a good job of preventing thesend audio from leaking to its output, the special re-mote send mix-minus is corrupted. Remember, if anyof the announcer audio from the remote site is returnedvia the monitor feed, it will be delayed by the digitallink, causing an echo effect. The answer is to make sureyou have the best possible hybrid with the maximumtranshybrid loss. If it has variable override (caller duck-ing), you could increase the amount when these re-motes are in progress.

The round-trip delay in a typical remote broadcastmay be reduced by using the G.722 algorithm forthe return cueing path and MPEG for only the on-air direction.

Cascading CodecsCoder cascading is an active field of investigation

among algorithm designers, standards organizationsand users.

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audible (softening of transients and pre-echoes,mostly), and this is a bad thing. L2 does not have thisproblem. However because it is closer to the edge forS/N, multiple generations result in unmasking (noiseand grit, mostly).

International Standards Organization (ISO)/MPEGproponents do not propose that a lot of passes of L3be used. They advocate that L3 be used at ISDN bitrates for remotes and that L2 be used at higher bitrates in other parts of the signal chain. This is whyISO decided to recommend the layers as they did: L3for 64 kbps/channel and L2 for equal to or greaterthan 120 kbps/mono channels.

My own experiments with codec cascading confirmthat this is the right approach—the two coding methodsseem to complement each other. Two passes of L3sound noticeably better than two of L2; a pass of L3followed by a pass of L2 also sounds better than twoof L2.

DIAL-UP REMOTES ON POTS LINES

We can not always have an ISDN line at a remotesite. Sometimes they are not available from the telco,or we just prefer not to use them because of the costor the delay and trouble of getting one installed. Sincecheap POTS dial-up lines are everywhere it makessense to find ways to use them for program remotes.The problem is that the 300–3.4 kHz frequency re-sponse and limited dynamic range that the dial networkprovides are not generally adequate for modern broad-cast needs.

Modem ~ Coding $ Broadcast Audio onPOTS Lines

A method that has emerged in the past few years isthe so-called POTS codec. This piece of gear combinesa high power coding algorithm with a fast modem. Ofcourse fast is relative here. Recall that ISDN can supplya minimum 56 kbps bit rate, while the fastest modemis limited to 33.3 kbps—and very rarely achieves thisspeed, usually settling at around 24–26 kbps. (56 kbpsmodems have the fast rate only in the downstreamdirection; the upstream remains limited to 33.3 kbps.)Because our goal is to achieve something approachingbroadcast quality, this is a very challenging bit ratefor audio coding technology.

Generally, we employ a kind of coding that is opti-mized for speech and very low bit rates. The mostcommon are taken from the code excited linear predic-tion (CELP) family, which have better audio quality forspeech at very low bit rates than the MPEG perceptualcoders used in ISDN equipment. They also have muchlower delay, a critical characteristic when live interac-tion is required. Perceptual coders work by using anunderstanding of how the human ear works, whileCELP algorithms model how the mouth produces voicesounds. Not surprisingly, CELP coders do a fairly poorjob with musical signals. There can also be problemswith background noises, such as applause.

Some equipment has the possibility to select fromeither CELP or an MPEG algorithm so that the usercan decide which trade-off to make.

Frequency ShiftingNow that the digital systems previously described

are available, the frequency shifting technique is usedmuch less often. Nevertheless, it has its place, andmany units are still in service. Frequency shifting of-fers a way to squeeze more high frequencies into aline than it will normally pass. More accurately, theprocess allows different frequencies than the usual300–3.4 kHz to be passed through a POTS phone line.

Frequency-shifting units using a single phone linemove all frequencies up by 250 Hz at the encode sideand down by 250 Hz at the decoder as illustrated inFigure 3.10-21. The result is a 250 Hz improvementat the low end at the cost of a 250 Hz loss at the highend. This means a typical phone line’s response willbe changed after the shifting process to 50–3150 Hz.The 250 Hz loss at the top is not very significant dueto the logarithmic nature of audio perception.

The shifting function is accomplished by heterodyn-ing the input audio with a low-frequency carrier. Thephasing single sideband (SSB) generation method isemployed to allow only one sideband to emerge at theoutput—the carrier and other sideband having beencancelled in the SSB process. Encoding and decod-ing can easily be accomplished in the same unit,since only a simple signal path change is requiredfor an encoder to decode, and vice versa. (See Figure3.10-22.)

Subjectively, the resulting frequency-shifter-pro-cessed audio sounds less telephone like. However, theresult of improvement at the low end without high endenhancement is often a somewhat muddy or flat sound.You can sometimes improve subjective quality byboosting the high end with a sharp EQ rise above 2kHz. A parametric EQ or custom filter is preferred sothat a high-Q curve can be obtained.

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Figure 3.10-21. A frequency shifting bandwidth extender allowsimproved low end response at the expense of a small loss ofhigh-frequency audio.


bution process. While it would be possible to use specialfour-wire telco circuits (or two standard loops) to main-tain independent signal paths to remote sites, it is moreeconomical and convenient to be able to use a singlephone line. To accomplish this, we must create effectiveconversion between the 2-wire phone line and a porton the 4-wire intercom matrix. It will be necessary toseparate the intermingled send and receive speech sig-nals on the phone line with a 2-to-4-wire converter, orhybrid. One approach is given in Figure 3.10-23.

Transhybrid loss performance will be importantwhen intercom stations with open loudspeakers andmics are to be used and when conferencing of multipletelephone lines is desired. In the first case, the acousticcoupling between the speaker and mic completes afeedback path which includes the hybrid. Clearly, thebetter the hybrid’s isolation, the higher the feedbackmargin. In the second case, a feedback path exists fromeach active hybrid through all of the others that areconferenced to it. When the total gain exceeds unity,feedback results. The goal is to have the best possibletranshybrid loss so that the maximum line-to-line gainmay be achieved.

An auto-answer and disconnect function may be re-quired for unattended operation. This circuit respondsto a phone line ringing signal by activating the hybridand de-activates the hybrid when the calling party hangsup. As discussed in the section on calling party control(CPC) a dial tone detector may be necessary to ensurereliable operation. The tone detectors are connected soas to respond to signals on the hybrid’s separated telcoreceive audio signal. Were this not the case, and the de-tector was merely connected across the phone line, therewould be a major problem when multiple lines are usedtogether inaconference.Why?Because the toneswouldbe conveyed to each line in use (through the intercomswitching matrix) from every other line, causing all of

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Figure 3.10-22. Single-line frequency extender uses SSB tech-niques to shift the audio 250 Hz at each end. A decode systemis shown; signals at X and Y are reversed for encoding.

INTERFACING PRODUCTION INTERCOMSYSTEMS

To aid communication with the field crew during re-mote broadcast projects, connecting the production in-tercom system to dial-up telephone lines is often re-quired. Smooth integration of live news remote feeds,for instance, requires that production personnel at alllocations be able to communicate with each other ina simple, trouble-free fashion. This is especially truewhen multiple remote sites are involved, as for electioncoverage, major sporting events and telethons. Ideally,crews at each location would use the intercom systemwithout regard for the distances involved. Most often,access to the dial-up phone network is available bywire or cellular, so an interconnection of the intercomsystem to the telephone network may be the solution.

4-Wire Intercom SystemsFour-wire systems are those in which the two speech

directions are kept separated in the switching and distri-

Figure 3.10-23. An arrangement which integrates a four-wire switching matrix with telco lines, an interruptible fold-back (IFB) feed, anda two-wire party-line intercom system for field production work.


the detectors to respond to the tones from all of the otherlines as well as its own. When one line’s interface getsa disconnect, all of the others would turn off as well!Therefore, there is a critical requirement in this setupthat transhybrid loss must be sufficient in order to becertain that any cross coupling is below the threshold ofthe tone detectors. The same situation applies with anyDTMF detection that is used on a per-line basis.

2-Wire SystemsThese are the popular party-line systems. Here, the

interface to requires two hybrids. The hybrids are con-nected back-to-back so that the intercom hybrid’s re-ceive output is fed to the phone line hybrid’s send inputand vice versa. Appropriate gain and processing stagesare inserted in the 4-wire path. This system is whattelephone engineers call a 2-wire-to-2-wire repeater.

High quality hybrids are required to prevent feed-back. As should be evident from Figure 3.10-24, thesignals can feed around the loop and feedback couldbuild up. This happens when the combined transhybrid

loss of the hybrids is not at least as great as the gainin the two amplifiers.

As telephone circuits have widely varying and un-predictable end-to-end transmission characteristics, in-terfacing intercom systems to phone lines without gainand without AGC is not likely to work very well.

ISDNBecause ISDN circuits are inherently 4-wire, they

are perfect for the intercom application. Used with a4-wire intercom system, speech paths may be keptseparated end-to-end. Applied to a 2-wire intercomsystem, the problem of maintaining sufficient hybridbalance is eased. ISDN lines are cheaper and easierto get than the special 4-wire lines sometimes usedfor intercom interconnection. Yet another benefit isthat ISDN offers two channels so that production andtalent feeds may be kept separate. Finally, a low delaycoding method such as G.722 can be used to improveaudio bandwidth in order to correspond more closelyto the fidelity users are accustomed to on local links.

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Figure 3.10-24. Two-wire intercom-to-telephone line interface (Telos “Link”).


tally, the Telecom Library is a very useful sourceof all kinds of material on telephones, from tutorialson 1A2 key systems to the latest on ISDN and digitaltechnology. Call for their free catalog.

Newton, Harry,Telecom Dictionary,Telecom Library,Gilroy, CA,. Available directly from Telecom Li-brary by calling (800) LIBRARY.

Teleconnect.Call (888) 824-9795 for subscription in-formation to this monthly magazine. Edited byHarry Newton.

Telos Systems’ website: www.telos-systems.com.

REFERENCES

Fike, John L., and George E. Friend,UnderstandingTelephone Electronics,Howard W. Sams & Co.,1984.

Flanagan, William A.,The Guide to T-1 Networking,Telecom Library, Gilroy, CA, 1990. A superblywritten and very complete description of the tech-nology and use of T-1 service. Highly recommendedif you need to learn about T-1. Available directlyfrom Telecom Library at (800) LIBRARY. Inciden-

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