When you get outside and into the real world, stray current, ground loops and other gremlins creep into the audio system.
The best way to ground equipment and avoid ground loops is by employing a star ground system.
With digital audio and surround sound systems becoming more commonplace, the public's awareness and expectations of audio quality is increasing. As more viewers install home theaters, high-end DVD players and even 96kHz-sampled monitoring systems, the need for TV stations to transmit the best in audio quality will even increase. Meeting these challenges is tough, even for an experienced engineer. However, lest we forget, not everyone has many years of audio technology experience. And, as many know, some video engineers still treat audio as “that noise that accompanies the picture.” To bring some sanity to the issue, this article is strictly targeted at those who may have something less than a Ph.D. in audio technology. Those of you with audio golden ears can go on to something else.
Check your wiring standards
Probably the single most common mistake for beginners is the wiring of an XLR plug. You'd be amazed at how often beginning technicians confuse the numbering and wiring of a simple XLR connector. For who knows why, it does not go from one to three or three to one. Rather, pin 2 is designated to be the hot or in-phase pin, pin 3 to be the cold or out-of-phase pin and pin 1 is ground. Years ago, there was no agreed-upon standard, and you may still find equipment with pin 3 wired hot. This is especially true with older, non-U.S. equipment and even some old Ampex VTRs. Check the operating manual for the pin configurations.
For 1/4-inch TRS (stereo plugs), military-style, bantam or tiny telephone connectors and jacks, the tip becomes pin 2 (hot), the ring pin 3 (cold), and the sleeve connects to the shield.
If you're dealing strictly with monaural signals, you can get away with a few phase reversals in your connectors and you probably won't notice it. However, connect two mics out of phase and try mixing them together and it's a completely different matter. The signals will cancel and create moving acoustic phase errors, resulting in havoc for the audience (and engineer). The fix is to swap the hot pin wire with the cold pin wire inside the XLR plug on one side of the offending XLR connecting cable. If you use this alternately wired cable to connect the two units, they will now be hooked up in-phase with each other. Be sure to label that cable as phase-reversed so that if it is unplugged and used again, it will be designated for use only when you need to reverse a phase. It's better to use commercially built phase-reverse adapters. They provide a quick and easy solution to phase problems. You can even paint them a special color so you'll know where they are and when they are in use.
Have a few phase-reversed cables and patch-bay jacks available in your studio, especially for use with rental or guest equipment that may have alternate pin configurations being brought in. This can save time and trouble if someone wants you to interface a special device.
If you need to make a cable with a 1/4-inch, unbalanced, mono connector on one end, and an XLR on the other, you must combine two methods of wiring. The hot connection goes to the 1/4-inch plug's tip on one end, and to the XLR connector's pin 2 on the other. The shield goes to the 1/4-inch plug's sleeve, and pin 1 on the XLR connector. If you are using single-conductor, shielded cable, you are done on the 1/4-inch side. For the XLR side, make the wire bridge between pin 1, which currently has the shield connected to it, and pin 3.
When wiring a 1/4-inch, unbalanced plug using balanced cable you have to combine methods. Connect the hot wire to the plug's tip and both the shield and old wire to the plug's sleeve.
Now that you know all you need to know about wiring XLR connectors, stop. All this becomes even more critical if you're dealing with powered microphones. When using phantom power for condenser microphones or direct boxes requiring power, you cannot connect the low side of the cable to ground. You cannot use unbalanced cables in these applications. If you do, you're going to ground or short the power supply. Not good.
Equipment levels and impedances
So what should the VU meter read? Low-impedance, balanced devices are often calibrated to output +4dBm with a VU meter reading of zero. Broadcast applications sometimes use +8dBm. That's a historical thing and it has to do with an antique technology called equalized phone lines. If you're old enough to remember this, you're probably not a beginner. Unbalanced equipment often is calibrated to an output level of — 10dBm with 0VU.
Historically, equipment designed to drive phone lines or long cables required a balanced transformer as the output device. Today, op amps can easily drive phone lines and long loops without difficulty. Even so, road gear usually uses transformers because of their reliability and ease of interface.
Transformers are ubiquitous and an old-fashioned technology that allow engineers to convert between impedances and levels, balanced and unbalanced as well as prevent ground loops. We still use them because, well, because they work. They don't require power or create heat. However, they do add weight, which is a consideration in portable equipment.
The second method of creating balanced circuits is with op amps. A pair of op amps can be used, configured 180 degrees out of phase with each other, to provide a high-quality, balanced output signal. These are cheaper and easier to use than transformers when building balanced circuits. The disadvantage is that they do not help cure ground loops and they don't like being connected to lines with voltage on them. A transformer can handle a certain amount of current, especially DC without affecting the audio quality. Try that with some op amps and you'll fry the output stages.
In the studio, you can get away with murder; almost anything works. It's when you get outside and into the real world that stray current, ground loops and other gremlins creep into the audio system. That's when a transformer can save the day.
Many pieces of newer equipment have been designed to easily interface with each other, so they will be more attractive to both the professional and pro-sumer markets. These usually have wider latitude in the levels they can receive on input and are able to output. They usually have adjustable I/O circuits to accommodate a wider range of equipment interfacing. They have also been designed to tolerate impedance differences, and many provide inputs for both balanced and unbalanced signals, on both XLR and 1/4-inch TRS connectors. Many of today's small mixing and touring consoles provide both XLR and ¼-inch I/O connectors. This makes interfacing with a variety of external devices much simpler.
Let's return to the issue of levels. A VU meter usually measures audio levels. A mechanical, analog VU meter relies on a needle to show the average level. The VU meters used on professional equipment display the average or root mean square (RMS) level of the signal. This level is defined as .775 percent of a signal's peak voltage. The ballistics of RMS VU meters are specifically designed to react not the peak level, but to the average power of the signal.
Don't assume that just because a device has a mechanical VU meter that it is calibrated to anything, let alone a standard. This is especially the case with consumer gear. You have to measure the device's output level with the needle set at 0VU to be sure what level you're dealing with.
Many devices today rely on LED or plasma metering. These electronic meters can indicate peak levels, VU levels or anything the manufacturer wants. Many times the meters are adjustable to several types of metering indication. Just keep in mind for now that there's a tremendous difference between peak and VU. (But, that's a subject for a different article.) Read the manual or measure the output level or you'll be guessing what 0VU means on your device.
If you're dealing with digital recording, typically DAT machines, watch your levels. DAT machines usually display peak levels, with a reference for 0VU somewhere between -14 and -18 on their meters. This allows for adequate headroom for the encoders. Analog circuits handle excessive peak signals smoothly and with less distortion than do digital circuits. Digital circuits go from low distortion at normal signal levels to massive distortion once you've exceeded the peak signal capability of the circuitry. That's another reason the meters read on the low side, to help prevent clipping the signals.
Plasma displays and LED VU meters come in many different types and appearances, and their features vary greatly. Some may display the average level, some the peak level, and some can display a combination of both. Others are not only capable of switching between metering various levels, but also include features like peak-hold display, which leaves an indicator lighted at the highest level the meter has thus far measured until a higher one comes along.
Types of signals
Not all audio signals are identical. Some signals have what's called slow attack times, others fast attack times. Technically, that just means that the rise time or duration of the signal is quick or slow. Each type of meter responds to signals in different ways. A VU meter may be bouncing around the — 15dBm level where a peak meter shows red — the overload range.
For example, a kick drum will produce a signal with an extremely fast rise time. The VU meter will barely begin to move by the time the sound is finished. This means the VU meter will tend to underindicate the actual peak signal level. A peak meter will be able to immediately indicate the exact peak level of short duration signals like this. All this means is that you may want to employ both types of metering and the choice is highly user dependent.
Eliminating ground loops
The most common gremlin in any audio installation is the ground loop. Even when you have done all your homework, there's still a good chance that when you turn on the system, you're going to hear a hum. That's the ground loop. It's likely to be your biggest nightmare. However, there are a few tricks you can try.
The first choice is to “lift” the ground on the input connector. The best way to do that is, again, with a special adapter device. This will disconnect the ground (shield wire) from pin 3 of the output plug from pin 3 of the input plug. This usually works.
Unfortunately, this won't work for microphone connections. You need the shield to keep hum from being introduced to the mic's signals. Lifting the ground may not work when using TRS connectors. Unbalanced inputs require that both pins have a signal to work. If you have hum there, we need to move to a whole new, and higher, plane of solutions.
In the U.K. they call it “earthing.” If you're faced with solving grounding problems, you'll probably find a few of your own names for the problems. Often, grounding equipment according to electrical codes guarantees that the system will hum. While you're not being encouraged to violate any EIA rules, you may have to wink a bit at theory and go with practicality on the audio connections to get this to work properly.
Usually, just disconnecting the shield at the input to the device will eliminate the hum. The hum is a result of two devices being grounded at slightly different electrical potentials. Current then flows between the devices on the ground (shield) wire and hum is induced into the signal. Disconnect one end of the shield (usually at the input device) and you'll probably eliminate the ground loop and, consequently, the hum.
The best way to ground equipment and avoid ground loops is by employing a star ground system. This technique requires the connections of the AC (chassis ground) of every piece of equipment in the studio to the systems' main ground point. A facility-wide ground system should be a part of every audio and video installation. Typically this involves a ground bus of heavy copper wire or plating located in the bottom or side of each equipment rack. Each device in the rack is then carefully bonded to that bus. Then all those buses can be connected with heavy copper wire to the central grounding star point.
Occasionally it is necessary to reverse or lift the ground on a piece of gear in a system to eliminate a ground loop. Many portable devices provide an AC power switch that has three positions. Two are on, one (center) is off. The difference is that the two on positions reverse the chassis ground polarity. If the console you're using has such a power switch, simply moving it to the other position may cure the ground loop. Do not operate any equipment above ground (without an adequate ground). This is unsafe. Do this and we could lose a reader.
Digital audio connections
The AES standard provides for proper wiring of digital audio connections, using an XLR connector. The interface provides for two tracks of digital audio to and from a digital device. These tracks are accompanied by several other digital signals, such as various sampling and bit rates, timecode, digital clock, and word sync, all on one, two-conductor, shielded cable.
Sony/Phillips Digital Interface (SPDIF), was originally a consumer audio format that later crossed over into professional territory. This method uses an RCA/phono-type plug along with a 75Ω cable to carry its signal. It's the same interface technology as that used for typical video connections on home consumer devices.
Many consumer devices now use optical interfaces, which are simple and reliable, but expensive. They're good for eliminating ground loops.
Alesis' ADAT Modular Digital Multi-track systems can perform two-way transfers of eight channels of digital audio via a proprietary, laser-carrying, optical cable. The cable used with this system also simultaneously carries SMPTE timecode and other digital control signals.
Tascam's Digital Interface (T/DIF) is used by its DA series MDMs and is familiar to many broadcast and post-production houses. It is capable of performing audio tasks similar to those of the Alesis ADAT system, while using TASCAM'S own version of optical interface.
Several companies have made interface boxes (such as Otari's Conversion System) that will convert from any one of several various digital audio systems to any of the other types. Some digital audio equipment manufacturers also include either Tascam or ADAT connectors on their products (sometimes both) and may include AES and/or S/PDIF capabilities as well.
Finally, if you do use a digital method of signal transmission to connect your system, don't forget to use a common timing source to sync them all together. In broadcasting, SMPTE timecode is the most common method of synchronization. Remember, however, that audio typically uses a frame rate of 30fps non-drop, while NTSC video uses a 29.97fps, or 30-drop frame rate as its standard.
If you do use SMPTE timecode for your sync, make sure all the audio gear locking to the timecode has its software set to the correct frame rate.
House sync or black burst generated by the video synchronizer in a broadcast or video setup is the best source of a master clock for the timecode used to sync the audio equipment to the video. Word clock, if available, is the best synchronization system for pure audio quality, because all the clocks in all the digital devices will be synced right down to the sample rate. If you happen to have an all-digital broadcast system and can generate word clock at the correct rates for the entire audio and video system, you'll really be looking and sounding good.
Take a few minutes to look over your studio setup, and see if you might better interconnect the equipment, while optimizing its operation. When that happens, you should be able to get your product to start sounding as good as it already looks.
Bob Buontempo is an independent audio consultant based in Elizabeth, NJ.