Conclusión

With phones an important part of programming at many stations, systems to enable convenient, high quality on-air integration of phone conversations are essential.

On air phone systems are specifically designed for use in the broadcast studio environment. While many business phone systems offer similar functions—line selection and status indication, conferencing—they are generally awkward to operate in an on-air environment and may have other limitations such as the audio qual-ity flaws described earlier.

While the phone network would not be considered to be a high fidelity source, it clearly does not help to degrade it further by adding additional noise, distortion or frequency response impediments. For that reason, broadcast phone systems are designed with these issues and other specialized requirements in mind. For exam-ple, a broadcast phone system output should be free of inappropriate switching sounds, and air talent should be able to access and manipulate lines live without any pops or clunks being audible to listeners.

Ergonomic Requirements

Line selection and other functions must be per-formed intuitively and with a minimum of hassle. Un-like a telephone set, broadcast line selection panels have large illuminated buttons. To avoid operator cfusion, features are limited to those necessary for on-air application. One such example is panels that drop into an open position in the studio mixing console so that the line selection buttons are located near the channel on/off, fader, and audio switching functions.

Conferencing Capability

Most broadcast systems allow any number of lines to be switched to air, even if only a single hybrid is present. But, unless you are blessed with excellent phone lines, you will want additional hybrids with each connected to the other through a multiple mix-minus arrangement. That way, it will be possible to have amplification between callers. Without multiple hybrids, callers might have difficulty hearing each The ISDN Broadcast Interface

Most of the functions performed by an ISDN inter-face are similar to that of an analog DSP hybrid, but there are some differences, both in the required func-tions and in the implementation of the common fea-tures.

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

Acoustic Coupling Reduction. There is often an acoustic path between the received caller audio and the send audio signal. This results from having a loud-speaker in the studio that produces sound that couples into the microphones. When the talent use headphones for monitoring callers, this is not a problem. But some-times it is not practical to convince guests to wear headphones, and television stations generally do not want talk show talent to wear earplugs. In these cases a combination of adaptive cancellation and dynamic gain reduction will reduce the coupling electronically.

High-grade Digital-to-Analog Conversion. When an analog connection to studio equipment is required, pro-grade converters can be used to provide much better quality than the usual telco conversion. At mini-mum, 16 bits should be used, but 18–20 bits may not be 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 necessary to adapt the sampling rate of the telephone network to the studio rate. telco sampling rate is 8 kHz, and studio equipment will usually operate at 32, 44.1 or 48 kHz. A process is required to perform the required up-and-down sampling, while suppressing aliasing and reconstruction audio components.

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

Dynamic Equalization. With phone sets having a very wide variety of microphone characteristics, a multiband automatic equalizer helps callers have a reasonable spectral consistency.

Caller “Ducking.” As with POTS hybrids, this can serve to reduce residual leakage. However, since ISDN hybrids have much better inherent transhybrid loss, this feature will be used mostly to satisfy a programming aesthetic requirement, reducing the level of the phone audio when the host talks and allowing her an

auto-451

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 451

SECTION3: AUDIOPRODUCTIONFACILITIES

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

Special Features

Desirable features for an on-air phone switching system include:

Busy/unbusy. To prepare for a contest, all lines may be busied-out and then returned to readiness after the contest has been announced.

Automatic next line selection. Pressing the next button picks up the line that has been holding the longest. If no line has been holding, the longest ringing-in lringing-ine 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 PBXs To 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 feed an input or two on the studio system so that that calls taken by the receptionist can be put on the air.

Route all lines through the PBX. The studio lines are programmed in the PBX to be forwarded to the ports 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 be trouble if a line is inadvertently picked up on one system while the other is being used.

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

Improving Phone Audio Quality

Whether 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 quality of our on-air sources. Thus, it generally pays to make telephone audio less of an earsore so that it does not stand out more than is necessary from other pro-gram material.

If the phone network is a digital system, why do phones 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 fidelity broadcast systems in mind. The 8/13 bit quantization scheme used for phone speech coding results in less than high fidelity. Often, the problem lies in the specific implementation rather than in any inherent shortcom-ing in the standard or the technology. One important quality limitation results from the anti-aliasing and reconstruction filters in the codecs. These filters usu-ally have an ultimate roll-off of around 35 dB. Audio above the 4 kHz Nyquist frequency will alias and appear 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 the speech sometimes heard on a telephone circuit is a result of the effects of this kind of distortion combined with audible quantization errors.

Also the codec filters generally use switched-capaci-tor technology, which tends to be fairly noisy. Some newer codecs avoid the switched-capacitor problems by employing the same delta-sigma over-sampling and digital decimation concept used for high performance digital audio conversion, but these are only rarely found in telco central office equipment.

What can we do? An ISDN connection solves half of the problem, since at least one of the telco’s codecs is bypassed. We still have the other end to contend with, and the majority of broadcast connections will remain analog. Fortunately, there are some remediation possibilities. Filtering, equalization, gating and dy-namics compression are the primary tools. Most of the commercial hybrid interfaces have at least some of these processes built-in.

Filtering

On a dial-up phone line, there is very little audio above 3.4 kHz—but there is noise. Thus, a filter with a very steep roll-off above the telephone passband will reduce phone line noise significantly without affecting conversation audio. The low-end can be improved as well. Low-frequency hum is often a problem—usually 60 Hz mixed with its second harmonic, 120 Hz. Thus, it is often a good idea to have a sharp roll-off starting at 200 Hz or so.

Equalization

An equalizer used to shape the frequency response of the phone line within its audio bandwidth can result in marked improvements in perceived quality. A typi-cal phone line has an excess of energy at around 400 Hz and considerable roll-off at both the top and bottom ends of its passband, so the idea is to compensate by adding gain at both. Boosts at 2.5 kHz and 250 Hz and a cut at 400—500 Hz with a parametric equalizer will help achieve better sound. Since every phone line is different, the ear is usually the best instrument to evaluate the results.

When it is not possible or practical to make custom adjustments, an adaptive multiband EQ can be an ef-fective tool. The principle is much the same as imple-mented in broadcast transmission processors. Audio is filtered into multiple bands, and an automatic gain adjustment is performed on each spectral segment.

Given the limited frequency range of telephone calls, three bands are sufficient.

Noise Gating

Another effective processing device is the expander or noise gate. These devices may be used to reduce gain between the words of a conversation, thus making phone line noise less objectionable. On extremely noisy lines, however, the gating action can make noise more distracting by causing it to come and go with the words. In such cases, it might sound better to 452

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 452

TELEPHONENETWORKINTERFACING

a feed taken from the main announce microphone may be all that is necessary. The patch send output available on many consoles is precisely what you need. In instal-lations where multiple microphones are to be used, a combiner of some sort is required. This may be a small outboard mixer or a homemade op-amp summer or even a resistive combiner. Better consoles offer special purpose busses that may be used for mix-minus, often with provision for selective switching of sources into the phone feed. If you need to modify an older console that does not have special buses, a device (made by Henry Engineering) accomplishes the mix-minus by subtracting the hybrid audio from the console program output with a differential amp scheme. This unit gener-ates a mix-minus signal true to its name—all sources except the phone itself will feed the phone.

Recording Phone Calls

Some stations may want to record calls for later playback. One technique is to have the mix-minus go to one track of a stereo tape machine, while the other channel gets the hybrid output with the caller audio.

The result is a two-track tape with the announcer and caller separated. To play back, the console’s input mode is set to mono; the relative balance, if need be, can be adjusted upon playback. The production department can use its tape to facilitate extraction of contest squeals.

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

Dynamics Compression

Levels on phone calls vary widely, and it is not uncommon to see levels range from140 to near 14 dBm as calls are switched into a given line. A compres-sor helps to smooth the levels. An AGC that maintains a constant compression ratio regardless of average gain reduction produces more consistency. Freeze gating is also important, so that gain does not increase during caller speech pauses.

Mix-Minus: Getting the Send Audio Feed The feed-to-caller signal has come to be referred to as mix-minus, so called because it is often the mix of all of the console’s active inputs minus the phone hybrid’s output (see Figure 3.10-20). A mix-minus feed is necessary because the hybrid will create a feed-back path if it is forced to chase its tail. Usually, the mix-minus is a mix of only the studio microphones, but it may sometimes include other audio that is to be sent to the phone such as contest sound effects from cart machines.

To create the required signal in simple installations,

453

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 453 Figure 3.10-20. Simplified studio block diagram shows the mix-minus required for hybrid feed.

SECTION3: AUDIOPRODUCTIONFACILITIES

Talk-Show Screening Software

In its simplest form this personal computer software lets a talk show screener/producer communicate to the air talent who’s on the line waiting to talk. It replaces the paper pieces on the window system employed for years at many talk stations. The better packages offer a number of convenient features: display of liner mes-sages and other information, storage of caller data for demographic analysis and remote operation via modem.

An Ethernet or serial port on the broadcast system can let the computer display reflect current line status.

New software enables laptop computers to extend full control capability and status display to a remote site, and modern systems even permit this function to be conveyed over the Internet.

ISDN: HI-FI REMOTES ON DIAL-UP LINES