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By
Robert Orban and Greg Ogonowski
Orban/CRL

System Considerations 

Headroom
The single most common cause of distorted air sound is probably clipping—intentional (in the audio processing chain) or unintentional (in the program chain). In order to achieve the maximum benefit from processing, there must be no clipping before the processor! The gain and overload point of every electronic component in the station must therefore be critically reviewed to make sure they are not causing clipping distortion or excessive noise. In media with limited dynamic range (like magnetic tape), small amounts of peak clipping introduced to achieve optimal signal-to-noise ratio are acceptable. Nevertheless, there is no excuse for any clipping at all in the purely electronic part of the signal path, since good design readily achieves low noise and wide dynamic range. Check the following components of a typical FM audio plant for operating level and headroom:
• Analog-to-digital converters
• Studio-to-transmitter link (land-line or microwave)
• Microphone preamps
• Console summing amplifiers
• Line amplifiers in consoles, tape recorders, etc.
• Distribution amplifiers (if used)
• Signal processing devices (such as equalizers)
• Specialized communications devices (including remote broadcast links and telephone interface devices)
• Phono preamps
• Tape and cart preamps
• Record amplifiers in tape machines
• Computer sound cards VU meters are worthless for checking peak levels. Even peak program meters (PPMs) are insufficiently fast to indicate clipping of momentary peaks (their integration time is approximately 10ms). While PPMs are excellent for monitoring operating levels where small amounts of peak clipping are acceptable, the peak signal path levels should be monitored with a true peak-reading meter or oscilloscope. Particularly, if they are monitoring pre-emphasized signals, PPMs can under-read the true peak levels by 5dB or more. Adjust gains so that peak clipping never occurs under any reasonable operating conditions (including sloppy gain riding by the operator). For older equipment with very soft clipping characteristics, it may be impossible to see a well-defined clipping point on a scope. Or, worse, audible distortion may occur many dB below the apparent clip point. In such a case, the best thing to do is to determine the peak level that produces 1% THD, and to arbitrarily call that level the clipping level. Calibrate the scope to this 1% THD point, and then make headroom measurements, Engineers should also be aware that certain system components (like microphone or phono preamps) have absolute input overload points. Difficulties often arise when gain controls are placed after early active stages, because the input stages can be overloaded without clipping the output. Many broadcast mic preamps are notorious for low input overload points, and can be easily clipped by high-output mics and/or screaming announcers. Similar problems can occur inside consoles if the console designer has poorly chosen gain structures and operating points, or if the “master” gain controls are operated with unusually large amounts of attenuation. When operating with nominal line levels of +4 or +8dBu, the absolute clipping point of the line amplifier becomes critical, The headroom between nominal line level and the amplifier clipping point should be greater than 16dB. A line amplifier for a +4dBu line should, therefore, clip at +20dBu or above, and an amplifier for a +8dBu line should clip at +24dBu or above. IC-based equipment (which almost always clips at +20dBu or so unless transformer-coupled) is not suitable for use with +8dBu lines. +4dBu lines have become standard in the recording industry, and are preferred for all new studio construction (recording or broadcast) because of their compatibility with IC opamp operating levels. The same headroom considerations that apply to analog also apply to many digital systems. The only digital systems that are essentially immune to such problems are those that use floating point numbers to compute and distribute the digital data. While floating point arithmetic is relatively common within digital signal processors and mixers, it is very uncommon in external distribution systems.Even systems using floating-point representation are vulnerable to overload at the A/D converter. If digital recording is used in the plant, bear in mind that the overload point of digital audio recorders (unlike that of their analog counterparts) is abrupt and unforgiving. Never let a digital recording go “into the red”—this will almost assuredly add audible clipping distortion to the recording. Similarly, digital distribution using the usual AES3 connections has a very well defined clipping point—digital full-scale—and attempting to exceed this level will result in distortion that is even worse-sounding than analog clipping, because the clipping harmonics above one-half the sampling frequency will fold around this frequency, appearing as aliasing products. Many systems use digital audio sound cards to provide a means of getting audio signal in and out of computers used to store, process, and play audio. However, not all sound cards have equal performance, even when using digital input and output. For example, a sound card may unexpectedly change the level applied to it. Not only can this destroy system level calibration, but gain can introduce clipping and loss can introduce truncation distortion unless the gain-scaled signal is correctly dithered. If theanalog input is used, gain can also introduce clipping, and, in this case, loss can compromise the signal-to-noise ratio. Further, the A/D conversion can introduce nonlinear distortion and frequency response errors. Level metering in sound cards is highly variable, with average, quasi-peak, and peak responses all common and often inadequately or incorrectly documented. This bears upon the question of line-up level. EBU R68 specifies reference level as –18dBfs, while SMPTE RP 155 specifies it as –20dBfs. If the sound card’s metering is accurate, it will be impossible to ensure compliance with the standards maintained within your facility. Many professional sound cards have adequate metering, while this is far less common on consumer sound cards. Further, consumer sound cards often cannot accommodate professional analog levels. Many audio editing programs permit a sound file to be “normalized,” which amplifies or attenuates the level of the file to force the highest peak to reach 0 dBfs. This is unwise for several reasons. Peak levels have nothing to do with loudness, so normalized files are likely to have widely varying loudness levels depending on the typical peak-to-average ratio of the audio in the file. Also, if any processing occurs after the normalization process (such as equalization), one needs to ensure such processing does not clip the signal path. If the processing adds level, one must compensate by applying attenuation before the processing to avoid exceeding 0 dBfs, or by using floating point arithmetic. If attenuation is applied, one must use care to ensurethat the attenuated signal remains adequately dithered.  

Voice/Music Balance
The VU meter is very deceptive when indicating the balance between voice and music.The most artistically pleasing balance between voice and music usually results from peaking voice 4–6dB lower than music on the console VU meter. If heavy processing is used, the difference between the voice and music levels may have to be increased. Following this practice will also help reduce the possibility of clipping voice, which is much more sensitive to clipping distortion than is most music. If a PPM is used, voice and music should be peaked at roughly the same level. However, please note that what constitutes a correct “artistic balance” is highly subjective, and different listeners may disagree strongly. Each broadcasting organization has its own guidelines for operational practice in this area. So the suggestions above are exactly that: just suggestions. It is sometimes difficult to train operators to maintain such a voice/music balance. However, this balance can easily be automated if the console has (or can be modified to have) separate summing amplifiers for live voice and music. Simply build a separate summing amplifier (using a single IC opamp) to drive the VU meter, and then sum the output of the voice-summing amplifier into the VU amplifier with greater gain than the output of the music-summing amplifier.

Electronic Quality
Assuming that the transmission does not use excessive lossy compression, DAR has the potential for transmitting the highest subjective quality to the consumer and requires the most care in maintaining audio quality in the transmission plant. This is because DAR does not use pre-emphasis and has a high signal-to-noise ratio that is essentially unaffected by reception conditions. The benefits of an all-digital plant using minimal (or no) lossy compression prior to transmission will be most appreciated in DAR service. FM has four fundamental limitations that prevent it from ever becoming a transmission medium that is unconditionally satisfying to “golden-eared” audiophiles. These limitations must be considered when discussing the quality requirements for FM electronics. The problems in disk and tape reproduction discussed above are much more severe by comparison, and the subtle masking of basic FM transmission limitations is irrelevant to those discussions. AM quality at the typical receiver is far worse, and “golden ear” considerations are completely irrelevant because they will be masked by the limitations of the receivers and by atmospheric and man-made noise. The four FM quality limitations are these:
A) Multipath distortion. In most locations, a certain amount of multipath is unavoidable, and this is exacerbated by the inability of many apartment-dwellers to use rotor-mounted directional antennas.
B) The FM stereo multiplex system has a “sample rate” of 38 kHz, so its bandwidth is theoretically limited to 19 kHz, and practically limited by the characteristics of “real-world” filters to between 15 and 17 kHz.
C) Limited IF bandwidth is necessary in receivers to eliminate adjacent and alternate channel interference. Its effect can be clearly heard by using a tuner with switch-selectable IF bandwidth. Most stations cannot be received in “wide” mode because of interference. But if the station is reasonably clean (well within the practical limitations of current broadcast practice) and free from multipath, then a clearly audible reduction in high-frequency “grit” is heard when switching from “normal” to “wide” mode.
D) Depending on the Region, FM uses either 50µs or 75µs pre-emphasis. This severely limits the power-handling capability and headroom at high frequencies and requires very artful transmission processing to achieve a bright sound typical of modern CDs. Even the best audio processors compromise the quality of the high frequencies by comparison to the quality of “flat” media like DAR. These limitations have considerable significance in determining the cost-effectiveness of current broadcast design practice. Most older broadcast electronic equipment (whether tube or transistor) is measurably and audibly inferior to modern equipment. This is primarily due to a design philosophy that stressed ruggedness and RFI immunity over distortion and noise, and to the excessive use of poor transformers. Frequency response was purposely rolled off at the extremes of the audio range to make the equipment more resistant to RFI. Cascading such equipment tends to increase both distortion and audible frequency response rolloffs to unacceptable levels. Modern design practice emphasizes the use of high slew rate, low-noise, low-cost IC operational amplifiers such as the Signetics NE5534 family, the National LF351 family and the Texas Instruments TL070 family. When the highest quality is required, designers will choose premium-priced opamps from Analog Devices, Linear Technology and Burr Brown, or will use discrete class-A amplifiers. However, the 5532 and 5534 can provide excellent performance when used properly, and it is hard to justify the use of more expensive amplifiers except in specialized applications like microphone preamps, active filters, and composite line drivers. While some designers insist that only discrete designs can provide ultimate quality, the performance of the best of current ICs is so good that discrete designs are just not cost effective for broadcast applications—especially when the basic FM and DAB quality limitations are considered. Capacitors have a subtle, but discernible effect upon sonic quality. Polar capacitors such as tantalums and aluminum electrolytics behave very differently from ideal capacitors. In particular, their very high dissipation factor and dielectric absorption can cause significant deterioration of complex musical waveforms. Ceramic capacitors have problems of similar severity. Polyester film capacitors can cause a similar, al though less severe, effect when audio is passed through them. Accordingly, DC-coupling between stages is best (and easy with opamps operated from dual-positive and negative power supplies). Coupling capacitors should be used only when necessary (for example, to keep DC offsets out of faders to prevent “scratchiness”). If capacitors must be used, polystyrene, polypropylene, or polycarbonate film capacitors are preferred. However, if it is impractical to eliminate capacitors or to change capacitor types, do not be too concerned: it is probable that other quality- limiting factors will mask the capacitor-induced degradations. Of course, the number of transformers in the audio path should be kept to an absolute minimum. However, transformers are sometimes the only practical way to break ground loops and/or eliminate RFI. If a transformer is necessary, use a high-quality device like those manufactured by Jensen4 or Lundahl5. In summary, the path to highest analog quality is that which is closest to a straight wire. More is not better; every device removed from the audio path will yield an improvement in clarity, transparency, and fidelity. Use only the minimum number of amplifiers, capacitors, and transformers. For example, never leave a line amplifier or compressor on-line in “test” mode because it seems too much trouble to remove it. Small stations often sound dramatically superior to their “big time” rivals because the small station has a simple audio path, while the big-budget station has put everything but the kitchen sink on-line. The more equipment the station has (or can afford), the more restraint and self-discipline it needs. Keep the audio path simple and clean! Every amplifier, resistor, capacitor, transformer, switch contact, patch-bay contact, etc., is a potential source of audio degradation. Corrosion of patch-bay contacts and switches can be especially troublesome, and the distortion caused by these problems is by no means subtle.

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