BROADCAST INTERFACING

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

One-way Interfacing

There is often a need to take audio from a phone or broadcast in only one direction at a time (newsroom phoners are a common application). If there is no requirement for a two-way conversation, a simple in-terface using a QKT will do. Formerly available from the phone company, this small box was permanently wired into a phone instrument or line and provided a quarter-inch (12.7 mm) phone jack output for feeding a 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 make your own (see Figure 3.10-14). The capacitor provides dc blocking so that the transformer does not become saturated with the phone line’s dc potential. In order for the coupler to hold the line by drawing loop current, eliminate the capacitor and use a transformer that can withstand the loop current without producing distor-tion. (One such a transformer is the SPT117 from Prem Magnetics.) 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 interface if you expect audio levels to get out of hand. Of course, commercial units are available that are a little fancier than the simple device described here. Some offer auto-answer and disconnect capability.

When using a coupler, it is most convenient to have the telephone instrument on-line and equipped with a push-to-talk switch on its receiver. This is because the phone’s receiver has to be off-hook while a feed is coming 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 included in the recording. Since this coupler works in both directions, it can be used to send audio down the phone as well—useful in the production studio for letting

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National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 446 Figure 3.10-14. Simple one-way-at-a-time interface. The capacitor

is for dc isolation and is not required when a transformer which can sink loop current is used. The zeners are chosen to properly limit transmission levels to the required19 dBm.

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

clients hear their commercial masterpieces before they go into the control room.

When hooking up to a multi-line phone, connect to a point where the tip/ring is present after line selection.

The most convenient place is usually right at the phone network. Use headphones to find the spot.

Two-way Interfacing

The simple coupler’s limitations become apparent when it is necessary for the caller to hear the announcer and the audience to hear the caller simultaneously. A more sophisticated method is needed because of the requirement to have isolated send and receive audio signals.

Switching

This is what you get when you connect a speak-erphone to your console input. No commercial broad-cast interface uses this technique, which uses gain switching to keep the send audio from appearing at the receive output. Two electronic switches or voltage controlled amplifiers are used in such a way as to ensure that either the send or the receive path is closed at any given time, but never both simultaneously (see Figure 3.10-15). A decision circuit compares the send and receive levels, with the direction of transmission being determined by the relative signal strengths.

The disadvantage of the switching technique is its uni-directional nature. The caller cannot be heard while the announcer is speaking, and noises in the studio can sometimes cause a caller to disappear momentarily, especially on weak calls.

The Hybrid

Hybrids were invented long ago to separate the send and receive signals from the common two-way phone pair. Early hybrids were made from transformers with multiple windings. Nowadays, most hybrids are made with active components and are known as active hy-brids. Both circuit types use the same principle and achieve 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 two inputs are added out-of-phase (subtracted). If the phone lines and the balancing network have identical charac-teristics, then the send signals at the second differential amp will be identical, and no send audio will appear at the output.

TELEPHONENETWORKINTERFACING

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

Broadcast Hybrid Application

In broadcast application, the studio mixing console combines the output of the hybrid and the announcer’s microphone audio, as illustrated in Figure 3.10-17. As discussed previously, the hybrid output consists of both the desired caller audio and the undesired leakage—

(the announcer audio but phase-shifted because of the phone line’s reactance). If the amount of leakage is too great and the phase shift too extreme, the announcer sound will suffer degradation as the original and leak-age audio combine in and out of phase at the various affected frequencies. When this occurs, the announcer sounds either hollow or tinny as the phase cancellation affects some frequencies more than others. Another effect of too little transhybrid loss is that feedback can result from the acoustic coupling created when callers must be heard on an open loudspeaker. Yet another problem can occur when lines are to be conferenced;

when the gain around the loop of the multiple hybrids is greater than unity, feedback singing will be audible.

So a hybrid will be useful for broadcast only when leakage is kept acceptably and consistently low.

The plots of phone line impedance vs. frequency and phase shift shown in Figure 3.10-18 are the result of measurements performed on phone lines at a radio station 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 impedance characteristics that could be emulated with a simple resistor-capacitor (RC) combination. These lines would work fairly well with a simple hybrid, since an RC balance network would match the impedance characteristic closely enough to make the cancellation of send audio at the hybrid output good enough to prevent coloration of the announcer audio.

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

R1 and the phone line form a voltage divider, as does R2 and the balancing network. If the phone line and balancing network are pure resistances, then, clearly, the phone line and the balancing network must have the same value in order for the signals at the differential amplifier to have the same amplitude and for complete cancellation to occur.

The phone line, however, is not purely resistive, but rather 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 are responsible for significant impedance bumps. Loading coils 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’s impedance curve which are difficult to null out.

Only when the impedance of the balancing network is the same as the phone line, and the signals at the differential amplifier are matched in both amplitude and phase, will full cancellation of the send signal be achieved. Otherwise, leakage results—the scourge of hybrids.

Because the phone company’s requirements are not generally 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 dB rejection, with 6 dB acceptable on difficult lines—just enough to prevent feedback in a system with back-to-back hybrids. When the situation calls for better performance, modules with a number of R and C ele-ments that can be switched in or out are employed,

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National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 447 Figure 3.10-16. Op-amp hybrid. The second op-amp is used as a differential amplifier to perform the required subtraction for nulling.

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Those other lines are quite another story! While it would theoretically be possible to construct a balance network to match the difficult lines, practical consider-ations usually keep this approach from being used.

The impedance characteristic required is too difficult to produce using resistors and capacitors. If the hybrid is to be switched among a number of lines, the line characteristic would have to be consistent from call-to-call and nearly the same impedance curve.

Digital Signal Processing Hybrids

Digital signal processing (DSP) offers a very power-ful and effective technology to improve hybrids. DSP is the process of operating on analog signals that have been converted into the digital domain. Since the sig-nals are numbers, mathematical operations can be

per-formed to manipulate them before being returned to analog. 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 announcer audio. To accomplish this, the announcer and caller audio signals are digitized and processed in a system that makes use of a specialized DSP microprocessor.

The digital hybrid incorporates software programmed to 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 maximum nulling (see Figure 3.10-19). Unlike resistor/capacitor analog schemes, the adaptive filter can create the

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National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 448 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.

TELEPHONENETWORKINTERFACING

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

Another thing to keep in mind—although the two are related, the transhybrid loss is not the same as the observed difference between the caller level and the leakage at the hybrid’s output. That is because the typical phone call is maybe 120 to 125 dBm (on choke lines, even lower) and the send level (to the caller) from the hybrid should be 110 dBm. That means that the hybrid has to use up 10–15 dB of its transhybrid loss just to get even. The remainder becomes the observed difference.

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

Combining the Hybrid and Switching Techniques This 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 or override function causes the dynamic rejection to be greater than the hybrid alone can produce. When send audio is present, the receive gain is reduced. Thus, leakage also is minimized. However, since the level from the phone is also reduced when the announcer is speaking, there is a sacrifice of full-duplex operation.

A user adjustment in the control signal path permits variation of the amount of receive ducking, allowing full duplex operation when the hybrid alone produces sufficient rejection, or speakerphone-like operation whereby the caller is turned almost completely off when the announcer speaks. As a practical matter, this control is usually set to provide the minimum amount of ducking which provides adequate send-to-receive leakage suppression.

ISDN For Studio Call-In Talk Systems

ISDN can provide a direct digital connection to the POTS analog network, so it can be used to enhance the quality of on-air calls. A call set-up message is sent from the customer equipment to the network to tell it to switch into POTS interworking mode. (This is in contrast to when an ISDN line is used with MPEG codecs. In that case, the line may be carrying voice signals but in a format that is incompatible with POTS phones. Instead, the network is providing a transparent end-to-end digital path.)

The cost of ISDN service is not a barrier. With ISDN lines costing about the same as analog in most parts of the United States. An ISDN BRI, with two channels, costs about twice as much as a POTS line.

(Pricing varies depending on the telco but ranges from a 20% discount to a 30% premium. The average is probably around a 10% premium.

Broadcast interfaces may use either BRI or PRI. A simple interface for the newsroom could use a single BRI. Even sophisticated multiline systems could use plex multiple break-point impedance vs. frequency

curves required by difficult-to-match phone lines. The send and receive signals are constantly compared in a feedback loop with the leakage becoming an error control signal which drives adjustment of the digital balancing network.

The performance advantage of the digital hybrid technology is striking. On a typical phone line with a fairly smooth impedance curve, an analog hybrid might attain 15–20 dB transhybrid loss. A digital hybrid will likely produce 40 dB or better transhybrid loss. On lines with difficult impedance curves, the analog hy-brid’s performance will usually be so poor as to prevent its 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 system to adjust to the phone line prior to the call going on air. The caller hears a noisy tone, but none of this tone is heard on the air since the output is muted. This has the incidental benefit of removing the line switching clunk. Adaption continues as the conversation pro-ceeds, using voice as the reference signal.

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

Evaluating Hybrid Performance

The amount of hybrid rejection—the transhybrid loss—directly affects the on-air audio and is the most critical measure of hybrid quality. The true test of hybrid performance is determined by measuring the amount of rejection across the entire audio frequency range, preferably with pink noise as a test signal at the send input. Any hybrid with an adjustable R and C balance network can produce high rejection at a

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National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 449 Figure 3.10-19. In the DSP hybrid, the digital balancing network

continuously adjusts to the phone line impedance characteristic.

When the adaptive network transfer function is identical to that of the phone line, perfect cancellation is achieved. Since the adaptive network is a digital filter than can create almost any required curve, performance is superior to the analog hybrid alone.

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

ISDN Lines Are Inherently 4-Wire

As we have learned, analog lines use a single pair of wires for both signal directions, mixing the send and receive audio. This causes the famous leakage problem—where the announcer’s audio is present on the interface output, instead of the desired caller only audio. Digital circuits inherently offer independent and separated signal paths.

While DSP based hybrids applied to the problem of separating the send/receive signals are dramatic im-provement over analog systems, ISDN enables further improved performance. This is because it offers a fully independent path for each speech direction. In the case where both ends of a connection are digital, there is no mixing whatsoever. In the call-in application, the far-end from the studio will still be 2-wire, so the audio paths will not be fully independent and a digital hybrid function will still be necessary to cancel residual leakage. But moving the studio side connection away from mixed analog can help tremendously because it provides the hybrid a much better starting point.

Better Digital-Analog Conversion Quality

The codecs used in telephone central offices are not as good as the converters commonly used in audio equipment. Fidelity is not an important consideration when designers choose parts for this function. In a professional interface for studio application, we are able to design with much better converters than avail-able in the telco’s equipment. Noise-shaping functions permit a larger word-length converter to provide sig-nificantly better distortion and S/N performance.

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

Lower Noise

As digital circuits, ISDN lines are not susceptible to induced noise. Analog lines are exposed to a wide variety of noise and impulse trouble-causers as they move across town on poles and through your building.

Hum is the main one, given the line’s proximity to transformers and ac power lines, but there are also sources of impulse noise from motors, switches and other sources. Digital lines convey the bits precisely and accurately from the network to your studio equip-ment without any perturbation—so the audio re-mains clean.

Call Setup and Supervision are Better

Analog lines use a strange mix of signaling to convey call status. Loop current drop and returned dialtone signal that a far-end caller has disconnected;

blasts of 100 volts at 20 Hz mean someone wants you to answer. Why should we be using a mechanism

blasts of 100 volts at 20 Hz mean someone wants you to answer. Why should we be using a mechanism