Article

Wednesday, May 16, 2012

Church Sound Files: What You Need To Know About Wireless Systems

An in-depth yet easy-to-understand discussion of wireless systems, how they operate, issues that can plague performance, and solutions that do the trick in the vast majority of situations.

Editor’s Note: This article provides straightforward explanations of the primary issues that account for a full 80 to 90 percent of all wireless microphone system problems, while also presenting solutions that will do the trick in most cases.

However, keep in mind that the best solution is avoiding these problems from the outset. Certainly this won’t guarantee completely trouble-free operation, but the odds dramatically improve.

This compilation of wireless system knowledge is provided by several highly qualified professionals, with Gary Stanfill, who has worked with wireless and related technologies for more than 40 years, topping this list.

Our sincere thanks to Gary as well as others who have contributed this important information.

This primer is presented in three parts.

Part 1, Getting Started, begins directly below.

Or, go directly to the other parts:
Part 2: Avoiding Wireless System “Issues”
Part 3: Downsides Of Digital

Part 1: PSW Wireless Primer
 
Getting Started
Anyone who has used wireless microphone systems for even a short time doesn’t need to be sold on their advantages. “Going wireless” allows concentration on the message rather than on the mechanics of delivering the message. (No more pesky mic cables!)

Yet wireless systems can be slightly mysterious, prompting suspicion among some users - particularly if they’ve experienced problems for unclear reasons.

The easiest way to understand wireless systems is to think of them as small-scale radio and TV broadcast stations – a transmitter sends out a signal that is picked up by a receiver.

For a number of reasons, including size, weight, battery life and government regulations, wireless systems operate at quite low power and thus have limited range.

The wireless microphone (or bodypack) is the transmitter, complete with a mic capsule, some audio circuitry, and an antenna (usually built into the case). It sends radio signals to its companion wireless receiver, which also has an antenna and some circuitry to select and process the signal, which is then sent via a cable to the sound system.

The transmitter and receiver of each wireless system must share the same frequency. Any other wireless systems in use in the same area must have their own frequencies as well. Ugly noise is produced if two wireless systems are using the same frequency in the same area. 

The same goes for other transmitters, especially those of TV stations.

And because these transmitters send out very powerful signals, they are a common cause of interference for wireless systems.

Even though a wireless system needs a clear frequency for the area where it’s going to be used, every frequency is used again and again across the nation.

Again, this is because the power of the output signal of wireless systems is very low.

Keep in mind, however, that there is no absolute guarantee that a clear frequency in one area will be clear elsewhere, even just across town.

This is an aspect about wireless systems that sometimes puzzles users; the government takes care of the problem for the high-power signals of commercial broadcasting, but wireless system users are responsible for avoiding this problem on their own.

Fortunately, most modern wireless systems (developed in the past 15 years or so) offer some degree of frequency agility (also called frequency synthesis). This means that the user is able to select an operating frequency from a number of possible choices, ranging from as few as four frequencies to 1,400 or more, depending upon the model.

The more frequencies offered by a wireless system, the better the chance of finding a clear frequency that is not being used by someone else in the area. Further, in larger cities, where there are more frequencies occupied by numerous users, the ability to choose from a larger number of frequencies is especially important.

Having plenty of open frequencies also helps wireless system users get around another potential problem: intermodulation (or intermod for short). This can occur where the frequencies of two transmitters (of any type) “combine” in a wireless system receiver, resulting in noise and interference.

Most often, intermod is caused by a combination of the frequencies from two TV transmitters, or by the frequency of a TV transmitter combined with the frequency of a wireless system transmitter.

Because the source of intermod is usually not under the control of the wireless user, there is usually little choice except to change the frequency of the wireless system. This is yet another reason for choosing a wireless system outfitted with a wide range of frequency selections.

By law in the U.S., wireless systems are supposed to operate only on TV channels not in local use. If a wireless system happens to cause interference to TV viewers in the area of its use (and this can happen even with their lower output level), the interference is likely to be reported, resulting in the user drawing unwanted attention from law enforcement. 

Thus it’s vital for the wireless system user to keep handy a list of local TV frequencies in use (available online at www.antennaweb.org/aw/Address.aspx), and to avoid those frequencies.

Although many wireless systems can “automatically” select frequencies or scan to see local RF activity, it is still possible to select the frequency of a local TV channel and get the innocent user into trouble.

Wireless systems are available for “VHF” and “UHF” frequency ranges (also called bands), roughly corresponding to VHF TV channels 7 though 13 and the UHF TV channels 14 through 69.

The question as to which range is “best” has pretty much been settled by the wireless manufacturers, who generally only offer systems with numerous frequency choices in the UHF band.

Additional bands used by wireless microphones include the “944 MHz” band between 944 - 952 Mhz. This is a band reserved for use exclusively for broadcasters.

Also, the “ISM” band between 902 - 928 MHz is an unlicensed band used by several wireless microphone products. Finally, the 2.4GHz band is another unlicensed area used by wireless manufacturers.

Although the UHF TV band classically extended up to channel 69, channels 52 to 69 (698 MHz to 806 MHz) has been converted to non-TV use - divided up by the U.S. government/FCC and auctioned to various companies for wireless devices available on the consumer market.

Accordingly, it is now against the law to use wireless microphone systems in this band. Even though a system has operated in this range without problems for years, it is illegal.

With all these competing signals in the air throughout the VHF and UHF bands, even high-quality wireless systems can run into problems when operating at distances of 100 feet or less between the transmitters and receivers.

Range problems usually appear as “fizzing” or “swishing” noises, perhaps followed by the complete loss of the audio signal. (This is called dropout.)

In addition to the low transmitter power, two other problems can limit the range of wireless systems. The first is signal absorption due to building construction and internal equipment, or shielding by metallic objects such as electrical wiring, air conditioning ducts, storage cabinets and the like between the transmitter and the receiver.

The term “line of sight” is often used to express the idea that the signal path from the transmitter to the receiver should be open and clear of obstructions.

This simply means that if the wireless user can physically observe the receiver antenna, RF signal absorption is likely to be low.

The second problem is called multipath. It’s a phenomenon that results in numerous small areas where little or no wireless signal is present because of reflections and the resulting phase cancellations, and it often tends to occur within a fairly short distance between transmitter and receiver.

To overcome the problem, a majority of modern wireless receivers now use a technique called diversity. With diversity, two slightly separated receiver antennas are used, making it very unlikely that both will simultaneously be in one of the low signal (multipath) areas.

The receiver automatically selects the antenna with the strongest signal, not only solving multipath, but also increasing the reliable range of a wireless system.

A final note: most users are surprised to learn - despite urban myths to the contrary – that the U.S. government requires wireless systems to be properly licensed prior to use. 

Unfortunately, the agency in change of issuing these licenses (Federal Communications Commission, or FCC) makes it very difficult for conscientious users to actually comply.

As a result, the vast majority of users don’t go to the trouble. But keep in mind that unlicensed wireless systems are in technical violation of FCC rules, and therefore are theoretically subject to fines.

As a practical matter, the FCC has neither the resources nor the inclination to go after the “average” wireless user, so the risk is low. But not zero. Due to the recent changes in spectrum allocation, this issue is being re-visited.

It appears that the FCC may make it easier for typical wireless microphone users such as churches, theaters, musicians, etc. to register their products.

This would also be beneficial in the event that additional types of consumer devices appear and complete for the same spectrum we are currently using.

Click here to continue to Part 2 (Avoiding Wireless System “Issues”) of this series, or click here to go directly to Part 3 (Downsides Of Digital).

Part 2: PSW Wireless Primer
 
Avoiding Wireless System “Issues”
Although the popularity of wireless microphones continues to grow, there’s no denying that they present more opportunities for problems than their wired counterparts.

In addition to the normal acoustic concerns that come with any mic are the complications of RF (radio frequency) transmission, interference, frequency selection, batteries and several other issues.

And technical improvements in wireless systems have not entirely kept pace with increasing frequency congestion, digital television and other recent complications.

Still, the hundreds of thousands of wireless systems employed in the U.S. is compelling evidence that the majority of users will live with the added challenges. Besides, many of the problems encountered by wireless users are largely avoidable, and happen primarily due to oversights, mistakes and misunderstandings.

Addressing the following common issues greatly improves the reliability of wireless systems and goes a long way toward ensuring trouble-free operation.

Issue: Frequency planning and coordination. Wireless systems share the RF spectrum with TV stations and several other types of authorized users. As a result, interference is very likely unless appropriate precautions are taken.

Solution: The first step is to determine the TV channels that broadcast over the air in your area.

When the local TV channels are known, they can be compared to the frequencies of the wireless systems. If there’s a conflict, the wireless frequencies must be changed. This is relatively simple for synthesized systems as well as ones that search for vacant frequencies, but the solution is more difficult for fixed-frequency wireless.

Despite the inconvenience, wireless systems should not be used on occupied TV channels. Not only is interference almost certain, the practice is illegal. 

Issue: Intermodulation. Wireless systems can also experience severe interference even when operating on “vacant” frequencies. This is created by intermodulation distortion - basically two strong signals on other frequencies combining in the wireless receiver to create an interfering signal.

Called “intermod” for short, generally this type of interference is more common than direct on-frequency interference from other transmitters.

Intermod is typically caused by other wireless systems, or by other wireless in conjunction with local TV signals.

Even single systems can be affected, but the probability of problems grows roughly proportionally to the square of the number of systems in simultaneous use, plus the number of active analog TV channels present. 

By the time eight or more wireless systems and six or more TV channels are involved, it can become quite challenging to find usable frequencies.

Solution: One or more wireless frequencies will have to change. There is generally no other practical solution.

Again, synthesized systems and “auto-search” frequency finding can be very helpful.

However, any frequency can potentially interact with any other, so changing one frequency can solve one problem can create another - or several others.

When changing frequencies or searching, it’s absolutely critical that all RF systems of any type at the location be turned on and operating.

As one clear wireless frequency is found, that system must be left on, and the next system tested until all are operational.  Otherwise, the situation can quickly become a snarl of changes and more changes, “phantom” problems, confusion and frustration.

Some manufacturers offer assistance in selecting usable frequencies, and as always, don’t hesitate to get your sound contractor involved.

In addition, there are a number of readily available software packages that are designed to aid in calculating your frequencies so that intermod problems are avoided.

Several manufacturers of wireless microphones offer this kind of software, and there are third-party options as well. Often, the third-party solutions are the most flexible – offering coordination of many types of systems by most manufacturers.

Issue: Shielding or covering antennas. In order to properly launch a radio wave, a sizeable volume of free space is required around an antenna, and in general, they must be unobstructed.

Solution: For efficient operation, all wireless system antennas must be kept clear of metallic objects that can weaken and distort signals in addition to reducing range. With bodypack transmitters, the antenna must be kept away from the mic cable, the bodypack case and ideally, the wearer’s body.

Securing antennas to the transmitter case and tying antennas to cables, as is sometimes done, can be absolutely deadly to range. Skin and flesh can absorb RF energy, so it is best to have the transmitter case and antenna away from the body.

Further, receiver antennas must extend away for the receiver case, as well as away from other antennas, equipment racks, other equipment, cabling and, again, metallic objects.

It’s best to mount receivers at the top of the rack so that the antennas extend above and away from the rack and other equipment. Using rear-mounted antennas inside a metal rack will almost always result in very poor reception.

For multiple receiver installations, the common practice of positioning front-mounted antennas in a “V” configuration, with all the antennas parallel, will also reduce range. It causes them to function together somewhat like a TV antenna that’s pointed upwards.

Even worse is when antennas from two different receivers touch. Not only will range be seriously compromised, interference becomes much more likely. In such a situation, it is much better to incorporate a single pair of antennas and then an antenna splitter to distribute the signals to the receivers in the rack.

Issue: RF path. A clear path between the receiver and the transmitter is also required. This is sometimes called a “clear line-of-sight,” but remember, light will pass in a straight line through a small hole while radio waves will not.

Solution: Similar to the free space needed around an antenna, radio waves require a sizeable space in which to travel.

The amount of space necessary depends upon frequency - the lower the frequency, the more space needed.

Create an imaginary tunnel of open air between the transmitter and the receiver antennas.

For UHF systems, a tunnel diameter of 3 feet or so is usually adequate, but for VHF systems, it should be at least twice as large. There also should be no metallic objects - scaffolding, iron beams, cables, cabinets, pipes, etc. - within this space.

In particular, large flat metal objects such large ducts, rows of cabinets, truck bodies and the like that are parallel to the path should also be avoided.

Even though they might not be in the direct path, they can still act similar to a mirror, reflecting RF energy away from the direct path. Systems with diversity reception help avoid dropouts in these situations, but range still can be reduced considerably.

Issue: Long antenna cables. Sometimes it’s necessary or desirable to locate antennas at a farther distance from a receiver. RF coaxial cables can be used to connect the remote antennas to the receiver inputs.

However, they typically have considerable losses that will reduce operating range. The amount of loss depends upon the size, construction and quality of the cable, and upon the operating frequency.).

Even high-quality RG-58 cable will have a loss of about 8 dB per 100 feet at 200 MHz, and about 17 dB at 700 MHz. Since every 6 dB of loss cuts range by half, the working range with 100 feet of this cable will be only 40 percent of normal at 200 MHz, and a mere 14 percent of normal at 700 MHz. 

Premium RG-58 type cables, such as Belden 7806R, are better, offering about 4.7 dB loss at 200 MHz and 8.9 dB at 700 MHz.  Still, at 700 MHz, only 68 feet of this cable will cut range in half.

Solution: If long cable runs are s necessary for your wireless systems to work properly, skimping on the cost of the highest quality cables available is a bad decision. For the best results, a premium foam-dielectric cable such as Belden 9913 should be used. This cable has only 1.8 dB of loss per 100 feet at 200 MHz, and 3.6 dB at 700 MHz.

Generally, it’s preferable to run audio cables out to remote receivers, keeping RF cables short. This is particularly true with runs longer than 75 feet or so. If remote location of the receivers is not feasible, go with the high-quality, low-loss cable noted above.

In-line RF amplifiers can also be used to boost the signal before the long cable run. These devices require power, and add cost. So before thinking that RF amps are the way to go, consider how the system can be configured to avoid using them and still keep your cable loss to a minimum.

Issue: Batteries. Simple but true and most certainly the number-one cause of wireless problems the world over!

Fortunately, it’s the one that’s easiest to fix.

The most common cause of short battery life is poor quality or old age, along with mixing used batteries with new ones and simply losing track of how long a battery has been in use.

Some sound operators also fail to understand that, when turned on, wireless transmitters draw power even if not being used, and that the “mute” switch does not affect the current drain.

Solution: Check transmitter batteries prior to every use. Get a battery tester to help you determine a good battery from a bad one. And when in doubt, change to a new battery!

Name-brand alkaline batteries such as Duracell and Eveready are the best bet. While private label batteries are often nearly as good, their useful life can vary considerably from purchase to purchase.

Make sure that to buy batteries that are date coded, and don’t accept any whose expiration date is less than three years away. And never use zinc carbon or toy batteries; most can’t even properly power up a modern wireless transmitter.

Classically, many techs recommend against use of rechargeable batteries, and for good reason. Rechargeable batteries used to have much lower capacity than alkalines, and the useful life was usually short. This was particularly true of 9-volt units, whose operating life was a fraction of that of an alkaline.

In the past five years, the technology for rechargeable batteries has improved dramatically. Now, NimH and LiPoly batteries are every bit as good as alkalines, and in some cases even better.

Still, it is important to recognize the added complexity of using rechargeable batteries – a clear strategy will be needed for keeping them charged, tested, and removed from the pool when the time comes. By doing this, you can save considerable costs and it’s also better for the environment.

Even more issues that are relatively simple to address can impact wireless performance.

Part 3: PSW Wireless Primer
 
Downsides Of Digital
Issue: Digital interference. Modern digital audio equipment, including processors, equalizers, controllers and other gear, operate at high clock frequencies that generate considerable radio frequency (RF) noise. (By the way, this noise is often termed RFI.)

As a result, it’s not at all unusual for such equipment to interfere with wireless systems.

Symptoms include low-level spurious tones, buzzing sounds, hissing and a varying noise floor.

Digital interference can also cause an unexplained loss of range and other problems.

Although FCC rules require that such equipment be tested to meet spurious emission standards, it’s a fact that not all units are indeed tested.

In addition, loose covers and casings, warped metalwork, lax grounding and other mechanical shortcomings can greatly increase spurious RF emissions.

Even properly approved digital equipment, in good working order, may generate enough RFI to affect wireless receivers located nearby.

When wireless interference occurs, one of the first things to do is to temporarily turn off digital devices to see if they are the source of the problem.

Solution: As a general precaution wireless receivers should be located as far as possible from digital gear. Often just moving the equipment a few rack spaces apart is enough to solve a problem.

More severe cases may require separating the wireless power, signal and RF cables from those going to the digital equipment.

Using remote antennas with the wireless systems may also be helpful.

And finally, try tightening up the covers on any offending digital gear and also adding a ground strap to the cabinet or other local ground point.

Issue: Lapel (or lavalier) (microphone sound quality. Lapel mics can cause a number of different problems. A common complaint is thin sound quality, which often occurs when the user has previously used only mics intended primarily for vocal applications.

These mics generally boost low frequencies to make the voice sound warmer and fuller, but the omnidirectional mics normally used with wireless bodypack transmitter systems don’t have this boost and thus can sound noticeably different.

Another cause of “thin audio” from lapel mics is interference. RF energy can “couple” into the mic cable and affect the preamplifier circuitry in the mic capsule. A high percentage of all lapel mics exhibit this problem under at least some circumstances.

If the voice quality and level varies when the mic and cable are moved around in close proximity to the wireless transmitter antenna and body, it is almost certain that RF interference is present.

Solution: In all cases, the manufacturer of the wireless system exhibiting this problem should be first contacted for specific recommendations. However, the problem is often solved with the addition of small RF bypass capacitors to the mic connector. Note that this should only be done by a qualified service professional only.

Issue: Lapel mic feedback. Users new to wireless often complain that a system is defective because feedback occurs where none was present before. Part of the problem is that the lapel mics typically used with wireless are not directional and thus provide little feedback protection.

However, the larger problem is usually that the mobility of wireless allows users to walk into zones more likely to cause feedback.

Solution: Use lapel mics with a unidirectional pattern, or use headset mics. Moving the mic closer to the mouth and lowering gain is also helpful. Many users think headset mics are unsightly, but unidirectional mics can suffer from sudden drops in level when wearers turn their heads.

The better solutions are acoustic, either by training users to avoid feedback zones, or by modifying the loudspeaker configuration to put feedback zones out of reach.

Issue: Lapel mic mechanical problems. This is common to lapel mics, in particular because their cables are small, often delicate and typically get considerable abuse.

Even if not damaged outright (i.e., the cable pulled out of the mic connector), lapel mic cables eventually wear out.

Most often this wear occurs first at the connector end, but keep in mind that it can also happen at the capsule end. Usually the cable shield fails first due to constant bending in the area where a cable leaves the connector’s strain relief.

When this happens, clicks, pops, other noise and “lost audio” are experienced. Even before there’s a complete break in the shield, pops and clicks due to RF disturbances can happen.

Therefore, it’s always prudent to check the cables when experiencing lapel mic noise of any type. Breaks at the connector end can usually be repaired (and don’t forget the bypass capacitors), but a break at the capsule end may not be fixable.

Mechanical noise due to lapel mic capsules rubbing on clothing is relatively common and can usually be eliminated by using the right type of mic clip, one that holds the capsule away from the fabric.

It may also be necessary to carefully secure the cable near the mic capsule. Static electricity sometimes creates audio noise, especially with certain types of fabric. Clothing anti-static spray usually solves this problem.

Issue: System quality. It may seem strange to list “system quality” as a wireless problem, but a great many wireless difficulties start with inferior equipment. Inexpensive systems can often work well in rural areas and/or in relatively undemanding applications.

But in larger cities and their surrounding suburbs plagued by typical frequency congestion and myriad interference sources, something better may be required.

The same is usually true when more than a few systems must be operated at the same site. And, this situation is going to worsen, with more and more digital signal sources going on the air almost daily.

The adoption of digital technology has greatly lowered the price of many audio products, but the impact of these advantages on wireless systems has been relatively small to this point. Wireless systems are still largely analog-based, and their manufacture is more labor intensive due to the requirement of considerable tuning, testing and tweaking.

Quality components also tend to be expensive in comparison to digital components and are less adaptable to low-cost automated assembly.

Unfortunately, there is yet no new magic technology that can cut the cost of a quality wireless system significantly - say 30 to 40 percent. Right now, if cost goes down, so do quality and performance. And it’s easier and cheaper for manufacturers to promote their mic capsules and “features” rather than build in better performance.

Consequently there is a growing tendency to regard the RF portion of a wireless system as being relatively unimportant. This is a serious mistake.

Solution: If a wireless system doesn’t have the selectivity and interference rejection to cut through all of the “junk” in the air, it doesn’t matter which mic elements it has, how neat the feature set, or how much money was “saved”. You’re simply left with something that doesn’t work like it should.

The recommendation is to pay a little more and go for performance over features. High-quality wireless systems cost less than half of what they did 10 years ago, and they work better in virtually all cases.

Final Thoughts: All in all, wireless microphone and in-ear monitoring systems can significantly enhance the experience for audiences and performers alike. Freedom of movement for actors, musicians, minsters, orators and politicians is a major benefit.

However, the complexity, cost and potential problems are the risks of using microphones. By following the guidelines presented in this series of articles, you should be well on the way to flawless operation from wireless systems.

Don’t forget that this is a changing world with respect to the RF spectrum and thus the operation of wireless mic systems. What works today may not work tomorrow.

Your best bet is to stay informed and educated. Watch for announcements about RF issues related to the FCC and potential other users of the spectrum. Keep up with the technology as manufacturers introduce new systems.

And most of all, stay up on troubleshooting skills so you can identify where the problems originate. Sometimes the wireless will be at fault, and sometimes not. It’s best to know the difference.

Click here to go to to Part 1 (“PSW Wireless Primer”) of this series, or click here to go directly to Part 2 (“Avoiding Wireless System “Issues”).

{extended}

In The Studio: Revelations In Recording & Mixing

Some quick "knowledge nuggets" shared by engineers

 

The ability to record and mix music is no doubt a skill and an art form.

Developing it takes time, and requires many “failures,” experiments, and learning experiences.

In my opinion, to truly succeed and excel in the recording industry – or any industry for that matter – you must have a craving and a passion to absorb the infinite amount of knowledge that’s out there.

I wanted to give you some quick knowledge nuggets that I received in response to a question that I posed to engineers on a recording forum.

I asked them: “What’s your most memorable ‘a-ha’ moment in learning to record & mix music?”

It’s difficult to pinpoint one “a-ha!” moment, as many of us have them all the time, but that’s a beautiful thing!

Regardless, I think some of us have a couple moments that really stand out – instances where we learned or observed something that stuck with us, opened new doors, and forever improved our abilities.

Below is a collection of what I thought were some intriguing responses to the question. I hope you learn something new from what others consider to be their memorable “a-ha” moments..

—“The moment i realized that there’s a big difference between knowing something and actually getting it.”
     
—“When, after years of recording rubbish, somebody of near genius level steps up in front of the microphone. Suddenly, I’m a recording genius too.”
     
—“Learning the most about the gear and tools I already have at my disposal.”
     
—“Compression: This is such a vast subject but getting the attack and release time is so vitally important. Compression can kill a performance or completely elevate it.”
     
—“Realizing that knowing techniques and tricks doesn’t make mixing that much easier, but knowing how you want things to sound is more important. The techniques just help you get there, perhaps quicker”
     
—“When I first realized that I could automate pretty much everything.”
     
—“The day I turned off my computer screen. Simple, but highly effective.”
     
—“The first time I recorded and sat down to mix in my newly acoustically treated room. Wow… first time I heard true seperation on my mixes.”
     
—“How simply pulling out a little 300-350Hz on most close miked tracks can really make things sound more real and less muddy and boxy.”
   
—“Subtractive EQ: get rid of freq’s instead of adding. How many times did I try to bring out a certain range, when it would have made much more sense in the big picture to take out what I didn’t want.”
     
—“Discovering Figure-8, and also hearing a ribbon mic on a distorted guitar amp.”
   
—“Avoiding ‘default’ presets – whilst presets and templates can be a time-saver some of the time, they can also be a crutch and a bad habit to get into..  better to listen then decide.”
     
—“When I removed all plugins from the session and found out that digital can sound decent after all.”
     
—“The first time I threw up a quick mix of the raw tracks instead of attempting to dial in “the perfect kick sound” etc on each track in an a la carte fashion. It cut my average mix time in half and increased overall quality.”
     
—“Getting stuck in a mix, throwing down all the faders, trashing all inserts and sends, and bringing up the faders again. Wow.”
     
—“Compression. Specifically dynamic EQ’ing via sidechain compression.”
     
—“You know, no matter how dynamic I get my performances to be, I find that I can always get them to another level by riding those faders (automation).”
     
—“When I stopped using presets on plug-ins to mix and realized I actually knew what sound I wanted and how to get it.”
     
—“Room mics & distance mics can add depth and dimension to tracks, when blended in – especially with drums.”
     
—“Making a quick level mix before EQ/Compression.
     
—“Buying my first high-end preamps. Took those same old microphones, and mic’ed my drum kit. I fell of my chair.”
     
—“When I first mixed in a well-treated room on good monitors and realized that all I needed to do was make the mix sound good in this room and on these monitors.”
     
—“High-Pass filters are your friend!”
     
—“Parallel compression allows more “natural” dynamics – and/or – use more compressors less aggressively.”
     
—“Musical Arrangement is VERY important to the outcome of a mix.”
     
—“You can’t always fix it in the mix. Don’t be afraid to re-track it if it’s not right”
     
—“Getting feedback on my mixes, even from non-musicians/engineers”
     
—“There are no rules – adjust as needed.”

Dan Comerchero is the founder and editor of the ProAudioFiles.com, a community blog where audio professionals from around the world share pro audio related articles, techniques, and advice on recording, mixing, production and more.

Be sure to visit the Pro Audio Files for more great recording content. To comment or ask questions about this article go here.

{extended}

Care & Feeding: Keeping Gear In Top-Flight Shape

Reduce problems, enhance system performance, and save significant time and money in the long run

To get the most mileage out of gear, regular equipment inspections and Preventative Maintenance (a.k.a., PM) are a must.

All equipment in your inventory should have PM scheduled at least once a year, and more frequently if it goes out the shop door a lot and/or is exposed to harsh environments.

PM comes down to inspecting, testing, cleaning, lubricating and repairing to keep systems in top operating condition.

In addition to annual PM, all gear should be given a quick inspection during setup and tear down at every gig. This includes a visual inspection, placing a hand on equipment to feel operating temperature, tugging on cable ends to see if strain relief is in good shape, etc.

If irregularities are noted, further inspection should be performed and problems addressed. Not paying attention to small problems allows them to build up to big problems that are much more expensive to correct, and they can also result in a failed gig. Here I’ll share some of the PM approaches I regularly utilize with my own gear.

Electrical

PM for electrical gear like processors, amplifiers, and snake boxes always starts with a complete visual examination. Each unit’s case is opened up for visual inspection of the interior. I’m looking for loose or broken wires, unseated connectors, blown fuses, discolored circuit boards, and so on.

While the case is open, it’s a great opportunity to run a vacuum and clean out all dust and road gunk that has accumulated inside. Sometimes an air compressor, or at least some “canned air,” is used to blow out the dirt. I also remove filters and clean or replace them per the manufacturer’s instructions.

Next up is checking and cleaning signal connections. If the equipment has faders and knobs, it’s time for cleaning and lubrication (again, per the manufacturer’s recommendations).

All electrical pins and connection surfaces are evaluated for corrosion and misalignment, and input and output connectors are given a thorough cleaning with an electronic cleaner such as Deoxit from Caig Labs. If connectors need to be repaired or replaced, this is the time to do it.

With the case still open, it’s a good ideal to double check all power cable connections, and if the unit has a fixed power cord, to make sure the strain relief is in good shape and the cord has no cuts or tears in the outer jacket. I also run my hand down the cable to feel for internal cable damage. If the unit takes batteries, they get a check, and the battery terminals are cleaned.

Before plugging in and powering anything, I make sure all cleaning fluids or solvents have dried. After a quick check to make sure the equipment is operating correctly, each component is sealed back within its case.

Rack-mount gear is a little harder to access without removing from the rack, but I strongly believe that doing maintenance is so important it’s worth the trouble. Note, however, that opening up some gear may void the factory warranty, so please read and follow all manufacturer instructions on maintenance.

Microphones

Modern microphones are pretty robust and usually don’t require a lot of attention, but they should be inspected after each use because they’re regularly dropped, exposed to liquids, etc.

Because the majority of my jobs are corporate gigs, which are usually relatively tame, I only do serious mic maintenance once a year.

But for those doing outdoor festivals and/or working with much more “raucous” forms of entertainment, there could be need to do maintenance as often as every month.

Many models allow you to remove a damaged grille/head to simply screw on a new one.

Factory replacement heads are usually available, and a few companies also make generic heads that fit popular microphone models.

And sometimes they can be fixed. For round ball-shaped grills, the handle of a large screwdriver can be used to gently pressure out dents. If a dent is a little stubborn, I place the ball on top of a folded towel and tap the screwdriver with a small wooden mallet.

Before grills are re-attached, they should be cleaned with a mix of dish soap and warm water, with a soft bristled toothbrush to help scrub out the dirt. Some folks use Listerine for cleaning, and there’s a foam-based cleaner called Microphome available as well.

Inner foam windscreens can be replaced or washed in a mix of dish soap and warm water. These should be wrung out and air dried completely before being reinstalled. For mics that don’t have removable grills, I use a dry soft bristled brush on the exterior of the grille to remove dirt and then hold the mic upside down to help loose dirt and debris fall away.

Don’t leave batteries inside mics between shows because they can leak and corrode the contacts and generally ruin the electronics. To keep these terminals (as well as mic connectors) clean, I use Deoxit, then wipe them dry with a clean cloth.

Don’t forget the clips! Mic clips should be checked for signs of cracks and missing pieces. Also evaluate the threads and the tightness of the swivel. I normally place a drop of light lubricating oil or WD40 on the threads so they’ll screw easier on to mic stands.

Loudspeakers

Safety is more important than looks or sound, so the first thing I check on loudspeaker cabinets is the rigging, making sure nothing is cracked, bent or distorted. All moving parts should be cleaned and lubricated per the manufacturer’s recommendations.

Also don’t forget to keep an eye on external hardware like handles, corners and grills, fixing anything that requires attention.

Connectors (and their panels) should always get attention as well, to make sure they’re intact and secure. For powered loudspeakers, give the power cord and amplifier a check before testing out the box.

During down times, I power up boxes and run a sweep tone through them to insure that drivers and crossover (if applicable) are O.K. For subwoofers, I usually run a kick drum sound from a drum machine as a general test, in addition to evaluating frequency tones.

Cables

Without cable and interconnect, a PA system is just a bunch of unconnected gear.

Yet cables seem to get the least attention – until they don’t work. After every usage, cables should be checked.

At the start of the wrap process, give the connector at that end the once over, to see if any pins or contacts are corroded or bent, and to confirm that the connector body is in good condition.

Make sure strain relief is tight and that the cable jacket has not pulled out of the connector body.

Then during the wrap, slide a hand along the cable, feeling for flat spots, twists or other irregularities inside the jacket. Check the outer jacket for cuts or tears.

At the end of the process, check out the other connector, then secure the cable and lay it in the proper storage case. (Don’t forget to also do this with AC extension cords.)

Cables that are obviously damaged or that need another look should be set aside. A common practice to mark a suspect cable is to put a half-hitch knot on each end, warning others not to use until it gets checked out. Another tactic is to place pieces of gaff tape over the connector ends.

It really only takes an extra second or two per cable to check them as they’re wrapped, but the extra seconds spent can save minutes (or hours) of chasing down problems at the next gig.

All cables should also get a more extensive yearly check and some PM, including signal check with a cable tester and a thorough cleaning. When using a cable tester, check for intermittent signals by wiggling the cable where it joins the connector, and also flex the cable at any suspect spots to see if there is a break.

Many times a cable may have a break in one or more of the conductors, but the problem won’t rear its head until it’s flexed or wiggled.

With analog snakes, check the strain reliefs, and also open up the stage boxes to check the internal connections. Clean and lubricate snake reels per the maker’s instructions, and ditto for both fiber optic cabling and reels.

For general cleaning of outer cable jackets, I use a cleaner/degreaser called Simple Green. For removing sticky taperesidue (and this applies to other gear as well), I turn toGoo Gone, a Citrus-based cleaner.

When that won’t cut it, I switch to a stronger solvent called Goof Off, which contains acetone, so caution is strongly advised. It will eat through many materials, so just use enough to get rid of the gunk in the affected area, and then thoroughly wash the area clean of any remaining solvent.

Stands

Ubiquitous and ever supporting, stands are often forgotten about until something breaks. Mechanical stands need maintenance just as much as sound reinforcement equipment.

On mic stands, check the clutch regularly to make sure it operates smoothly. Replacement parts are available from manufacturers to rebuild a loose clutch mechanism. I also remove gaff tape residue (Goo Gone or Goof Off ), dry the tubes with a rag, then work a few drops of oil into the end threads so they screw into the bases and clips easily.

If I spot any damaged threads, I “chase” them (running a thread cutting die over a section to try to repair the threads) or simply cut off the end with a pipe cutting tool and make new threads on the fresh section of pipe.

Evaluate the stand’s base and replace any rubber isolation feet as needed. For tripod stands, check the legs and lubricate the hinge joint with a silicone- or Teflon-based lubricant. For loudspeaker stands, the process is similar, and also address rivets that hold the leg hinges as well as any safety stops on the stands.

Last month I focused on racks and cases (here), and these items also require scheduled maintenance. Check all hardware in general, and make sure rack rails are firmly bolted to the rack shell. Clean and lubricate the handles and hasps with a silicone or teflon lubricant, and clean and grease the castors as per the caster manufacturer’s recommendations.

While it may seem like a large outlay of effort, keeping up with regular PM can reduce problems, enhance system performance, and save significant time and money in the long run.

Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb, and is the owner of Tech Works, a production company based in Las Vegas.

{extended}

Church At Former Harley Davidson Factory Outfitted With NEXO Loudspeakers

When LifePointe Christian Church in Elk Grove, CA was looking for a sound system to compliment a new location—a converted Harley Davidson showroom—it turned to the expertise of CCI Solutions of Olympia, WA, with project manager David McLain recommending NEXO PS15 loudspeakers for the 400-seat church.

Specifically, the new system includes three PS15s and a flown NEXO CD18 subwoofer, all driven and controlled via a 4x4 NXAMP.

“Pastor Chris Delfs sought out a new location where they would not only own their space but could create a unique, new culture,” states McLain. “They weren’t looking for a NEXO system per se, but were looking for good advice. At only 60 feet deep, with less than 20 feet of trim, a line array would not have been the best tool, whereas, the asymmetrical horn pattern of a PS15 reaches the back rows with ease.”

“David actually discouraged a line array approach in this case, and if you look at how much the HVAC and lighting impacts the trim height, the PS solution was much better,” notes Steve Armstrong of PROS Inc.,an iindependent rep firm for NEXO.

“I’d been hearing a lot of talk about NEXO speakers over the past few years,” and have listened to various models and I was impressed enough to recommend them for a couple of rooms, particularly given the outstanding support I’ve been receiving from Yamaha Commercial Audio Systems,” says McLain. “I didn’t really know the extent of the NEXO lineup until I had opportunity to listen critically and extensively to their whole selection of speakers at a demo at the Cerritos Performing Arts Center in California.

“The Cerritos Center is an awesome building. We used their beautiful main room to try out the NEXO speakers. I spent the first demo day designing speaker systems for rooms using various software, and I liked the way the NEXO speakers worked in the planning, and also liked what I saw in the computer models. The NS-1 software was easy to work with, so I imported some real-world rooms, like LifePointe, that I’d been working on. It appeared the NEXO PS Series would provide excellent coverage.

“I have to admit, there’s a fair bit of skeptic in me,” he continues. “A box that promises a rectangular coverage pattern had better do more than just advertise well! It needs to offer an actual rectangular coverage pattern. And more importantly, it needs to sound good! In the next day’s listening tests, I measured 112 dB at the back of the listening room, and I have to say, it sure didn’t feel like 112 dB. In fact, it didn’t sound like a PA playing. It sounded like a woman was right there singing to me.

“Yamaha brought in a live jazz drummer and they just sounded louder, like there was no PA in between. Even the little NEXO PS8 two-way sounded way bigger than its small size. And, all of the subwoofers for the line arrays – which were shaking my pant legs at 100 feet – are cardioid subs. Even during the ‘fairly loud’ cuts (think aggressive Sunday morning volume), we could easily hold a conversation on the stage behind the subwoofers.”

CCI Solutions
NEXO/Yamaha Commercial Audio

{extended}

Tuesday, May 15, 2012

Accuracy vs. Realism: Simulating The “Human” Side Of Audio Measurement

Do you want to know what is actually happening, or what is perceived to be happening?

The human auditory system is equipped with two inputs - left and right ears.

This “binaural” processing system provides us with the ability to localize where sound is coming from, something that a one-eared listener would have difficulty in doing.

Playback systems may utilize any number of channels to surround the listener with sound, but two channels is always enough to simulate the human listener.

Recording enthusiasts have long discovered the benefits of stereo microphones. While not necessarily “human-like,” they can produce recordings that add spaciousness and realism to the recorded material.

Two-channel acoustic measurements are important for the same reason - they add a human characteristic to the data.

For our discussion here, I’ll use the term “binaural” to describe recording processes that provide data for two ears - there is no need to distinguish between making a recording and making a measurement, as either or both could be of interest.

Let’s look at some of the ways to get binaural data. Many modern measurement platforms support two-channel recording. We will assume that one of them is being used, allowing our discussion to be confined to microphone techniques.

One of the first decisions that must be made by the data gatherer is whether accuracy or realism is more important.

After a little consideration, it becomes apparent that one cannot have both. Setup parameters that provide a more accurate view of the loudspeaker’s response will require that the effects of the environment be minimized.

On the other hand, if the effect of the room is to be considered, then accuracy will need to be sacrificed to include it.

The question becomes, “Do I want to know what is actually happening, or do I want to know what is perceived to be happening?”

The answer to this question will fundamentally affect the method used to collect the data.

It’s important to note that at least three responses are being gathered in the recording - the loudspeaker, the listener and the room.

The listener’s response is a constant. The ear/brain system is assumed to be processing sound the same way at every seat. The loudspeaker’s response can be dramatically position dependent, but it does not have to be.

Loudspeakers that are designed for covering an audience evenly can have a similar response over a large area.

The room also has a response, but it is unique for each listening position. This is one of the reasons why we can’t correct room acoustic problems with electronics.

Is the goal of the measurement accuracy or realism? If the purpose of the measurement is to calibrate an equalizer or crossover network, then accuracy should be considered first.

It is desireable to know the true acoustic response of a transducer at a point in space, usually for the purpose of improving this response through signal processing.

In Figure 1 you wil see a stereo microphone on a stand at ear height might convey what a listener will hear, but this response will include seat-dependent artifacts, such as a strong reflection from the floor or other nearby objects.

The resultant comb filters will make it impossible to observe the response that is due to the loudspeaker alone.

If one were to attempt to compensate for the effect of the floor reflection, the compensation would not be correct for a closer or more distant listener seat. As such, it is best to ignore the floor reflection altogether when “tuning” the system.

Also, such a “seat dependent” response would average out if a large number of measurements were averaged across an auditorium.

This is why near-field and ground plane measurement techniques play an important role in sound system tuning. (This article is about neither – we’ll table this discussion for the future.)

Case For Realism
If the measurer wants to know what a sound system/room sounds like, then accuracy must give way to realism. Realism requires a binaural recording technique, and it must include the same effects from the room that might affect a live listener.

Mic placement is actually much easier than when considering accuracy, as the measurer simply listens to the system wherever he/she like and then replaces his/her head with the microphone. See Figure 2 for mic choices, which include:

Stereo.

A simple stereo mic can yield left/right information. Two cardioid mics in an X/Y configuration can yield convincing stereo.

Spaced omnidirectional mics are another popular method. This is art, not science so there really aren’t any rules to break. If you like what you hear, then it’s O.K.

Head Simulation. An added element of realism can be achieved by simulating the presence of a human head. The “head effect” is called the Head-Related-Transfer-Function (HRTF.) The Crown SASS uses omni mics spaced at human dimensions with an absorptive mass in between.

Frequency-dependent directivity is achieved by boundary-loading the mics on small, flat panels.

Head/Torso/Pinnae Simulation. Perhaps the best binaural mic is the dummy head. This includes the effect of the head, torso, and even the ear structure. The major benefits of this technique are customization and repeatability.

The response can be modified electronically and physically to whatever is desired, and setups can be recalled in the future if needed.

Digital signal processing provides a low-cost, powerful way to modify the response.

Dummy heads can cost many thousands of dollars, but the cost is easily justified for researchers that need the benefits.

Human Mics. One way to make a “poor man’s” dummy head is to utilize your own (no offense intended). Everything is already in place except the microphones. I’ve seen numerous mic placement mechanisms over the years, including eyeglass mounts, wires, and even earrings.

Possibly the most clever and realistic approach to date is the In-The-Ear (ITE) recording technique pioneered by Don and Carolyn Davis in the late 1980s.

This involved placing probe mics at the surface of the ear drum. This technique captured the outer ear response, including the ear canal resonance. The resonance was removed with an inverse filter during playback.

A variation on this technique that sacrifices some accuracy for practicality is to place small mics at the entrance to the ear canal. I will call this “At-The-Ear” to distinguish it from the previous technique.

The mics are held in place by some foam inserts (Figure 3).

The two mics have XL male connectors that can connect directly to my data recorder.

I normally survey an auditorium without wearing the mics to determine the measurement positions, and then return to the seats with mics in place to gather data.

Figure 4 shows a comparison between a free-field measurement and the “At-The- Ear” placement in both the time and frequency domains.

The responses have been overlaid for comparison.

The methods used to gather data are determined by the intended use of the data.

This often requires more than one technique, each preserving or enhancing the information in a way that yields more insight into the particular problem being solved.

When making measurements, arrive equipped to acquire both accurate data and realistic data, and then let the question being pondered determine the preferred perspective.

Pat Brown teaches the Syn-Aud-Con seminars and workshops. Synergetic Audio Concepts (Syn-Aud-Con) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, Syn-Aud-Con is dedicated to teaching the basics of audio and acoustics. For more information visit their website.

{extended}

Basic Principles For Suspending Loudspeaker Systems

A few terms and techniques for hanging loudspeakers

Excerpted from JBL Professional Technical Note Volume 1, Number 14: “Basic Principles for Suspending Loudspeaker Systems.”

Design Factor
Design factor is a term used by the rigging industry to denote theoretical reserve capability.

The rated capacity / of all lifting and hanging equipment b based upon the nominal strength of the equipment reduced by the design factor.

Design factor is a number representing the fraction of equipment nominal strength chosen to be appropriate for the particular application.

Rated Capacity = Nominal Strength / Design Factor

Example: Design factor = 5. The rated capacity of equipment is only l/5 of its nominal strength.

Minimum design factors vary according to the application, and may be regulated from location-to-location.

No design factor discussed herein should be assumed to represent a recommendation on the part of JBL.

Users must assume all responsibility for the determination of design factors suitable for local conditions.

Shock Loading
When a load is suddenly moved or stopped, its weight may be magnified many times the original value. This is known as shock loading. Shock loading of lifting equipment should be avoided at all times.

Shock loads will usually be instantaneous and may go undetected unless equipment is visibly damaged. No equipment is designed to compensate for poor rigging practices or foolish planning, however.

Every tool and piece of equipment has limitations. Safe working practices demand that these limitations he known and fully understood, and that they never be intentionally exceeded.

A 900 pound loudspeaker cluster dropped four inches cod cause a shock load of 4500 pounds if the rigging were attached to rigid structures and of a material that would not stretch.

However, because all rigging will stretch under shock loading, the exact shock load on a piece of equipment isn’t easily predicted.

To protect people and property, all tools and equipment should be limited to stresses that are several times smaller than their minimum breaking strengths.

Although shock loading of equipment and structure is usually confined to lifting and installation, it should also be recognized that other forces (such as earthquakes) can impose shock loads upon structures many times that of the static load.

It is therefore imperative that hardware and structures be capable of supporting several times the weight of the equipment being hung.

Center of Gravity
The center of gravity of an object is the point at which the weight of the object acts as though it were concentrated. It is the point at which the object may be completely supported or balanced by a single force.

The center of gravity of a regularly shaped object may be estimated fairly accurately by determining its approximate center.

Finding the center of gravity of irregularly-shaped objects can be more difficult, but it is necessary, nevertheless. A load will always hang from its attachment point through the center of gravity. It is important to visualize this before making a lift.

All loads to be lifted should be rigged above the center of gravity in order to prevent tipping and possible hazards to equipment and workers. The lifting force should always be located above the center of gravity and exert a straight vertical pull to prevent swinging of the load.

Ropes
Before discussing actual rigging hardware and systems, it is appropriate to examine ropes and their proper use. Ropes are used for many rigging functions.

Although synthetic ropes of great strength are available, most codes prohibit their permanent use in rigging for a variety of good reasons. Nevertheless, ropes are necessary to lift approved cables, fixtures, tools and equipment into position.

In the interest of safety it is important that ground workers be familiar with the proper use of rope and a few basic knots used in rigging.

Rope Terminology:
1. The Standing Part is the end of the rope which is inactive.

2. The End is the part of the rope that is free—typically the part in which knots are tied.

3. A Bight is the central part of the rope between the standing part and the working end.

4. An Overhand Loop is formed by crossing the end over the standing part.

5. An Underhand Loop is made by crossing the end under the standing part.

6. Tightening. Once formed, a knot must be tightened slowly and with care. Failure to do so could result in a tangle, or an untrustworthy knot.

Knot Efficiency
Knot efficiency is the approximate strength of a rope with a knot as compared to the full strength of the rope.

It is expressed at a percentage of the ropes rated capacity, and refers to the stresses that the knot imposes upon the rope.

When a knot is tied in a good rope, failure under stress is certain to occur at the knot. This is because bends result in uneven stresses upon the fibers, with the outsides of the bends taking a greater share of the load.

lt follows that the tighter the knot, the greater the percentage of the total load that is carried on fewer fibers.

Bends
Bends are used to join two pieces of rope, usually temporarily. Typical knot efficiency is 56%. Bends offer some advantage over binding knots, as they resist untying when slackened or jerked.

The Sheet Bend is a simple knot to tie, consisting of an overhand loop on one piece, with the second rope end fed up through the loop from behind, around the standing part of the first rope and back down through the loop from the front.

Binding Knots
Binding knots are also used to join two pieces of rope. In general, binding knots have a knot efficiency of 50%, but can untie easily when a free end is jerked.

In the square knot, the end and the standing part of each line tie together through the bight of the other. In the untrustworthy granny knot, the end and the standing part are separated by the bight.

The granny knot is particularly treacherous in that it will appear to be secure-only to slip under load. The thief knot is deceptively similar to the square knot, but has the two loose ends coming out of the opposite sides, instead of from the same side as in the square knot.

This knot is almost certain to fail under load.
Loop Knots
Loop knots are used to hold objects where security is of paramount importance. The bowline, widely used in rigging, won’t slip, yet is easily tied and untied.

It may be tied in the hand or used as a hitch and tied around an object, usually for lifting purposes (Figure 2 in the PDF below).

Hitches
Hitches are used for temporary fastenings that untie readily. They are generally tied directly around the object instead of first being tied in the hand and then placed over the object.

Hitches must be drawn up tight, as they have a tendency to slip if loose.

The clove hitch (Figure 3 in the PDF below) consists of two underhand loops, which may be tied in the hand and slid over an object at any point along the length of a rope. Knot efficiency is 60%.

Wire Rope
Vast wire ropes are constructed from plow steel, improved plow steel, or extra improved plow steel wire. The wires are woven into strands, which are woven to form the wire rope.

Typical wire rope may consist of six strands wound around a central core. The central core supports the outer strands and helps to prevent the rope from crushing under stress.

Wire rope core materials may be fiber (abbreviated FC), independent wire rope (abbreviated IWRC), or wire strand (abbreviated WSC).

Wire rope is classified by diameter, number of strands, number of wires making up each strand and core material construction.

Rope diameter is measured at its widest dimension. Wire rope is also classified according to the direction the strands and wires are twisted. The distance along the rope required for a strand to make one full revolution is one Lay.

In Right Regular Lay construction, strands twist to the right, wires twist to the left.

Right Lang Lay construction finds both strands and wires twisting to the right.

Left Regular Lay ropes are constructed with strands twisted left and wires twisted right.

The Left Lang Lay configuration twists both strands and wires left.

Regular lay ropes are less susceptible to crushing and deformation because the wires lie nearly parallel to the rope. Lang lay ropes twist the wires across the direction of the rope, and are therefore more flexible and resistant to abrasion damage. If both ends of a lang lay rope are not fixed, however, it will rotate severely when under load.

Most sound and stage rigging requirements are easily handled by two wire ropes: 3/V and l/2” 6 X 19 IWRC classification.

These ropes in improved plow steel have a nominal strength of 13120 pounds and 23000 pounds, respectively. If we assume a design factor of 5, rated capacities become 2600 and 4600 pounds.

Just as knotting a fiber rope reduces the nominal strength of the rope, bending of a wire rope also results in a reduction in its nominal strength.

The tighter the radius of the bend in the rope, the greater percentage of the load is concentrated on fewer wires and strands. This results in a reduction in the rope’s nominal strength and rated capacity.

Experienced riggers always pad beam edges with softeners before wrapping the beam with a sling, and avoid sharp or jagged edges that could possibly injure the wire rope or sling. Heavy burlap or thick polyester is usually used for this purpose.

Excerpted from JBL Professional Technical Note Volume 1, Number 14: “Basic Principles for Suspending Loudspeaker Systems.” Copyright and courtesy of JBL Professional. To continue reading and to see he referenced diagrams, feel free to download the PDF.

{extended}

Monday, May 14, 2012

Fill-osophy 101: Using Fill Loudspeakers To Optimize Coverage, Localization & More

The majority of successful sound system designs rely on at least one type of fill loudspeaker.

A well-designed main/primary loudspeaker system is expected to provide clear and intelligible sound to the entire audience.

Over the years our industry has benefitted from a steadily growing selection of types and configurations of primary loudspeakers to do this; they possess the specific characteristics (performance and physical) that we need in our base inventory, or for a given project, or to solve a specific problem.

Many of these varied types of products are also available over a wide range of size, quality and price options.

Yet more often than not when we deploy primary loudspeaker devices - singly or in all manner of arrays - there still may be holes in the coverage. This occurs for a number of valid reasons:

• Not enough primary loudspeakers for the specific venue;
• The need to avoid aiming primary loudspeakers to cover all of the front seats (at high frequencies) due to the low and mid frequency energy that ends up on stage or in the orchestra pit;
• Box seats that are located close to the stage (proscenium) that can’t be covered by the primary loudspeakers due to their close proximity;
• Architectural elements, including balconies and columns, that may shadow the sound projected from the primary system;
• The depth of seating (indoors and outdoors) can require additional devices, located further into the seating area, to “kick it up” a bit; 
• The need to provide sound to contained areas to the sides of the stage.

And the list goes on.

Coverage is an absolutely critical performance characteristic, at the very heart of what we must provide. The challenge to be met is supplying the correct number of fill loudspeakers that individually, or collectively, insure the coverage needed to fill the existing hole(s) and to ensure minimal overlap into the coverage that is already provided.

Few low-profile fill loudspeakers exhibit narrow coverage patterns across a wide frequency range and are also inherently restricted in acoustic output. Therefore their throw is restricted, although this actually may be of benefit in some cases.

Further, in many cases, clear and intelligible coverage is not our only concern. There often is an equally important need to provide or maintain localization to the source, and for as many seats as possible.

Localization is defined as the aural perception that the sound is coming from the apparent point of origin (where we locate the source visually, usually on the stage or platform) or, in larger venues and specifically for those seated further away, from the stage itself. 

The need for localization is no more prevalent than in sound reinforcement for musical theater, but in many other venues and types of productions there may be just as much desire to provide realistic imaging. This can be accomplished (or improved) using fill loudspeakers plus various processing functions, measurement and our ears.

We see these challenges often, usually when someone has misinformed us, or forgotten to mention key information, or has “changed their mind” on seating location/configuration.

So we grab what we can (assuming that we can) and set it/them up to fill any holes. It might be sloppy but it usually solves or reduces the immediate problem.

Similar Behavior
When faced with advanced knowledge of the need to fill holes in coverage or improve imaging, etc,. the type of loudspeaker we use requires just as much thought as we (hopefully) apply to the primary system devices.

Ideally, we’re able to utilize loudspeaker(s) made by the same manufacturer as the primary loudspeakers so that there is sonic consistency throughout the system.

Most reputable loudspeaker manufacturers go to great length to ensure that the various devices within a model line, if not their entire catalog, sound alike and exhibit similar behavior across their pass bands.

This can help to save time during measurement and optimization and may be more appealing and/or visually reassuring.

We must also provide appropriately sized fill loudspeakers. Too big may look bad (and may block sight lines) and too small may run out of gas (including potential damage to drivers) in addition to not providing the required pattern control.

The word “throw” relates to the distance a loudspeaker is able to project sound to a specific (targeted) seating area, and at the required sound pressure level (SPL).

Let’s look at a hypothetical example that’s quite common in the real world of permanently installed (and some touring) point-source array systems.

At the top of the array, there are “long throw” loudspeakers.

These 2-way, horn-loaded loudspeakers provide 40-degree (horizontal) by 20-degrees(vertical) dispersion, projecting sound to the most rear seating areas at sufficient SPL, without scattering sound on to the walls or ceiling.

Below that are “medium throw” loudspeakers supplying 65-degree (h) by 40-degree (v) coverage to the “heart” of the coverage area, and below that are 80-degree (h) by 50-degree (v) devices employed as “short throw” to cover seats in the nearer field.

When aimed and combined correctly, arrays of this type can provide complete, fairly seamless (including even levels) vertical and horizontal coverage.

A Necessary Part
Note that only a few of the multiband line array systems offer optional narrow horizontal coverage devices intended to sit at the top of the array for what we would consider to be long-throw duty.

But in the far more common line arrays with duplicate/like devices, the elements are aimed and gain adjusted for the desired characteristic of providing even SPL from front to back, and we would therefore categorize the collective elements at the top as “long throw.”

Properly designed (shaped) and configured J-arrays consist of elements or sections that we would categorize as long, medium and short throw.

But be it line arrays, point-source clusters or any other approach to mains, fill loudspeakers have remained a necessary part of what we do for decades.

We may need just one, a few, or many, but the majority of successful sound system designs rely on at least one type of fill loudspeaker to achieve complete coverage and/or for localization.

Over the decades we’ve assigned names to categorize fill applications: front, in, out, under balcony, over balcony, delay, box seat, lawn, down, overflow, stage, and so on.

Most of the time a device to meet a specific need is available, but once in a while, a new, specialized “custom” loudspeaker may need to be developed.

While there are numerous passing references to fill loudspeakers in articles and on manufacturer websites, and a book that discusses the alignment and equalization of them within larger systems, my research indicates that there’s nothing readily available that thoroughly details applications faced by live sound practitioners on a regular basis.

So next time out, I’ll begin a two-part series looking at specific fill design approaches and practices, as well as the basics of fill system optimization. 

Tom Young is a senior engineer at Altel Systems, Inc., a metro-NYC contracting firm. Recent projects include The Juilliard School and WaMu Theater at Madison Square Garden. He is also principal consultant at Electroacoustic Design Services in Connecticut and recently measured/optimized loudspeaker systems at a recital hall, a swimming pool complex and several churches. In addition, he is the moderator of the ProSoundWeb Church Sound Forum.

{extended}

Church Sound: Helping Vocalists Master Their Monitors

Putting too much stuff in the mix for a vocalist only causes confusion

 
Let me start off by saying that Tim Corder has done a great job of writing about monitor mixing. He has several posts tailored for various musicians and even went so far as to include audio examples.

If you haven’t already read them, you should. Tim wrote about building IEM mixes for Electric, Bass, Drums, Keys, and Lead Vocal. Check it out when you have a few minutes.

With that said, I was thinking about helping vocalists set up a useful monitor mix, particularly on a wedge. Though we hope to be putting the band on IEMs soon, the vocalists will stay on wedges for some time.

I was talking with our worship leader about helping the vocalists—all of whom are volunteers—set up better mixes. In my experience, setting up a good monitor sound for a volunteer vocalist is one of the hardest jobs in church audio (perhaps all of audio for that matter), mainly because they often don’t really know what they need.

Most vocalists tend to think they need the entire band in their wedges, all mixed to sound like a CD. While a professional singer (who recorded the CD in the first place) might be able to get away with that, I’m not convinced that technique serves the volunteer vocalist. So here’s what I recommend.

I’ve been using this technique for a good many years, and it was confirmed a few years ago by non other than Robert Scovill during a session I attended (with Tim, incidentally) at the Willow Arts Conference.

I’ve found that putting too much stuff in the mix for a vocalist only causes confusion. The more we put in their wedge, the harder it is for them to sing on pitch—because it’s a lot more for the brain to process. So I suggest, and Robert concurs (hmmm, probably should be the other way around) that vocalists need three, maybe four things in their wedge:

—Tempo reference
—Pitch reference
—Their voice and (optionally)
—Harmony reference

Tempo Reference

This could be snare, kick, hi-hat or perhaps overheads. It doesn’t need to be (and arguably should not be) a complete drum mix. Which part of the drums they need for time will depend on how the drummer plays and what songs they are doing.

In the end, all they need to know is the tempo. Personally, I like hi-hat for this because it’s up and out of the vocal range and thus easy to pick out. It’s also subtle enough that it doesn’t overpower the mix. Your mileage may vary.

Pitch Reference

Again, depending on the band and orchestration, this could be piano, keys or perhaps a guitar. Also again, it should not be all three. Whatever the source, it should provide pitch and key reference. And that’s about it.

Their Voice

One might think this is the most important element—and it is. Singers need to hear themselves. In fact, what is most typically asked for in a monitor mix? More me. The challenge of course, is that if the whole band is in their wedge, we have to put a whole lot “more me” in there so they can hear themselves.

Then they sound unnaturally loud (at least to their ears), so they ask for more of everything else. Which leads to another round of “more me.” Stripping the mix down to Tempo, Pitch and themselves makes it easier to keep levels under control and gives the vocalist what they need.

Harmony Reference (optional)

If you have a group of singers who are singing harmony together, it is sometimes helpful for them to hear the other parts of the harmony. Use this sparingly, however, as it’s really easy to get into an out of control level situation again.

Vocalists need to spend some time learning to hear their voice. It’s a matter of training on their part. By the same token, we can help by not giving them more than they need (which ultimately confuses them).

Now we know what they really need; how do we convince them that this is actually good for them?

Talk to Them

Sounds crazy, I know. However, too often sound engineers will stand behind the board at front of house (or monitor world) and try to convince someone on stage they need this or that. I suggest this is a poor way to go.

Get out from behind the board, walk up to the stage and have a conversation. Don’t take the tack, “I’m the sound guy, I know best,” or “I read this on some really smart guy’s blog,” (let me know which one it was, by the way…).

Rather, come at it as a suggestion. Note that you’ve noticed that they often are struggling with the monitors. Offer an option to try something new that you think might help. Ask them to try it for a few weeks to see how it feels.

Give them some rationale for your technique, then work really hard to make it work for them. Seriously folks, if our bands know we’re working really hard to serve them well, it won’t even matter if it sounds better or not. They will come around.

Show Them

I once had a really hard time convincing vocalists to hold their mics closer to their mouths than their navel. I tried all kinds of things: Explaining the inverse square law (met with blank stares); turning down their feed in the monitors to make them sing louder (didn’t work); motioning with an imaginary microphone (resulted in very confused looks).

Finally, during a break, I picked up the talkback mic and demonstrated it. Near…far. Near…far. I talked close to the mic, and far from the mic. They instantly heard the difference and they all held the mic properly after that.

Same concept here. After you’ve talked with them, set up a mix the way you think it should be. Then have them try it. Point out to them how much easier it is to hear the pitch, tempo and their own voice with a simpler mix.

Often, it takes just one song for the lights to go on in their heads.

Get Worship Leader Buy In

Sometimes, you really need the worship leader to have your back. Again, a lot of this comes from relationship. If you have a good relationship with your worship leader, you can talk with him or her and come up with a plan. If they get the concept, if you have to you can push it through.

When we changed from wireless IEM to Aviom at Upper Room, I spent a lot of time talking that change through with our worship leader. He agreed with the switch, and when I asked how he thought the band would react, he said, “They’ll use whatever we give them. It’s not up to them.”

He didn’t say that with arrogance, but with the understanding that the band is not in charge. As the worship leader, it’s incumbent on him to make the good calls; not in a vacuum, but ultimately, it’s his call.

So give that a try. And remember, if they continue to resist, you are the one who turns the knobs…

Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.

{extended}

In The Studio: The Latest On “Mastered For iTunes”

The most up-to-date details directly from the source

 
On Friday I had the pleasure of sitting in on a meeting with three representatives from Apple at Oasis Mastering, as they briefed us on the latest info on the new “Mastered for iTunes” program.

While there’s already some info on the program currently available, it was good to finally get the most up-to-date details directly from the source. Here’s the gist of the program.

“Mastered for iTunes” at its most basic is iTunes finally opening up to hi-res masters. This means a number of things:

1) iTunes now prefers that you supply the master audio files at 96kHz/24 bit, but any sample rate that’s a 24 bit file will still be considered “Mastered for iTunes.” Music files that are supplied this way will have a “Mastered for iTunes” icon (like on the left) placed beside them to identify them as such.

The reason why they’re asking for 96/24 is so they can both start with the highest resolution source material for a better encode, but also for a bit of future proofing in the event that iTunes later converts to a better format or a higher encode resolution (it’s now 256kbs, but more on this in a second).

2) “Mastered for iTunes” doesn’t mean that the mastering facility does anything special to the master except to check what it will sound like before they (or the record label) submit it to iTunes, and then check it later once again. All encoding for iTunes is still done by Apple, not by the mastering houses, record labels, or artists.

The reason for this is to keep the encodes consistent and to prevent anyone from gaming the system by hacking the encoder, but also to avoid any potential legal problems that might occur when a mastering house sends the files directly to iTunes instead of the label without their permission, or uses different specs, etc.

3) As stated above, the mastering house doesn’t do any encoding directly, but Apple has provided a number of tools that they can use to hear what the final product will sound like when it’s encoded. That way they can make any adjustments to the master to ensure a good encode.

One unique aspect of “Mastered for iTunes” is something that’s not been publicized called a “test pressing.”

When Apple finally encodes the file, they’ll send a copy back to the label/engineer/artist to check. If they sign off on it, the song then goes on sale in the iTunes store.

Of the few mastering houses that are currently participating in the program (all of the major ones), it was surprising that most of the time a test pressing was rejected not because of the audio quality, but because it was the wrong master.

Yes, as record companies seem to do, someone would actually send the un-mastered file or a completely different song or version. Luckily, the problem is now able to be caught in the test pressing stage.

4) Speaking of the sound quality, iTunes is now using a completely new AAC encoder with a brand new algorithm and the sound quality it produces is stunning. It provides an excellent encode if you use a few common sense guidelines (more on this in a bit), and if you do, the result is almost impossible to hear (at least on the music we listened to).

I mean, there we were, mastering engineers Eddy Schreyer, Gene Grimaldi plus myself, listening in this fantastic listening environment, and we literally couldn’t tell between the source and the encode most of the time.

Now there were some where we could hear the difference too, but it wasn’t that big a difference and certainly didn’t sound anywhere near as bad as the typical MP3.

So what are the tricks to get the best sound quality from an iTunes encode?

It turns out that the considerations are about the same as with MP3 encoding:

  a) Turn it down a bit. A song that’s flat-lined at -.1 dBFS isn’t going to encode as well as something with some headroom. This is because the iTunes AAC encoder outputs a tad hotter than the source, there’s some intersample overs that happen at that level that aren’t detected on a typical peak meter, and all DACs respond differently. Something that won’t be an over on your DAC may be an over on another playback unit. If you back it down to -.5 or even -1 dB, the encode will sound a lot better and your listener probably won’t be able to tell much of a difference anyway.

  b) Don’t squash the master too hard. Masters with some dynamic range encode better. Masters that are squeezed to within an inch of their life don’t. Simple as that. Listeners like it better too.

  c) Although the new encoder has a fantastic frequency response, sometimes rolling off a little of the extreme top end (16k and above) can help the encode as well.

5) “Mastered for iTunes” is only an indication that a hi-res master was supplied; it’s not a separate product. There will always be only one version of the song on iTunes at the same price as before. “Mastered for iTunes” doesn’t mean you get to charge more, or that iTunes charges you more. Everything is like it was before, you just supply a hi-res master so it sounds better.

6) So how do you supply that hi-res master? This is where it gets a bit tricky. If you’re signed to a major label, they’ve been contacted my their Apple reps and everything is in place, so no problem there. If you’re with an indie label, insist that they contact their Apple rep for instructions.

If you use CD Baby or Tunecore, at the moment they’ll tell you they don’t take 24 bit or high sample rate masters. Insist that they contact their Apple rep and don’t take no for an answer (this is what the Apple iTunes guy told us).

Apple is greatly encouraging everyone to get with the program, so the more pressure you put on them, the quicker it will become a standard. Of course, if you can find out who your local Apple rep is (ask the local label), that could expedite things too.

The bottom line is that “Mastered for iTunes” is a great thing for digital music. As far as I can see, there’s no downside to it (except maybe for the initial hassle you may go through as an indie), and you’ll be giving your fans a much better sounding product as a result.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

 

{extended}

The Backbeat Goes On: Microphones & Techniques For Snare Drum

A few easy rules are helpful, along with some simple tools

In the beginning, the list of microphones on drums was minimal, even in recording studios.

Before the arrival of rock and roll in the late 1950s, with its steady emphasis on the 2-and-4 backbeat, putting microphone near the snare was out of the question, and there weren’t many drum microphones on The Ed Sullivan Show.

The difference between country and western was the drums required to push western swing music, but it wasn’t until the Grand Ole Opry moved to Opryland in 1973 that entire drum sets were even allowed on stage.

By that time, mic’ing the snare was as normal as nailing the kick drum to the riser.

Early models commonly used on snare include the Shure 545 “Unidyne III” and Electro-Voice RE-15. Also introduced in the 1960s, the Shure SM57 dynamic remains the hardest working snare microphone and first choice of many of engineers.

Besides the obvious “We’ve always done that,” or “It’s what’s in the microphone box,” there’s a logical reason for this preference.

Many still employ the age-old method of checking the PA by repeating the mantra of “check, one-two” into a vocal microphone, and since there’s no discernable difference between the sound of an SM58 and a 57, it’s no surprise that the same engineer who tunes the system with a 58, finds the 57 sounds natural on snare (or any other close-mic’ed instrument).

If you deny an old sound guy a 57 to put on his snare, he’s as likely to use a 58 as anything else.

TUNING THE SNARE

Like the rest of the kit, if the snare sounds bad to begin with, you’re likely to end up with a louder, bad sounding drum. Though drastic EQ can sometimes make up for shortcomings and help a bad drum sound better, there’s no substitute for simply doing the deed of tuning the drum.

While entire books can be written on tuning drums, especially snares, a few easy rules are helpful, along with some simple tools.

Snare top heads (“batter”) are usually coated with a rough surface to give brushes more sizzle, while bottom heads (“resonant”) are uncoated and thinner.

Before mic’ing up a drum kit, it’s good to know how old the heads are and when they were last tuned. A drum can go from bad to good with a few minutes and a drum key, but this annoying chore should be tackled early, before others need to be in the room.

Both heads should be loosened and then tightened just enough to take the wrinkles out. By tapping the head lightly near each lug, the tension can be evened out so the tones all match.

Most methods for tuning the snare involve the bottom head being a few notes higher than the top head, A looser bottom head produces a fatter or “wetter” sound, while tighter produces a drier sound with more “pop.” The fundamental tone of each head affects the other, so small changes in one head can produce dramatic results, creating a sound that’s more muted, or more open, depending upon where you started out.

The timeworn method of tightening opposite pairs of lugs keeps the head centered and evenly tensioned. The best drum tool is a drum key, but the second-best is a second drum key, and those who subscribe to this method - as many eventually do often tie them together with a lanyard. A second key not only speeds up the process, it more evenly tensions the drum, making it easier to tune.

ARE YOU JELLIN’?

A snare drum with lots of resonance or ringing is often not what a live engineer wants, especially in a reverberant venue, Traditionally drummers (or their engineers) have resorted to adding a gaffe-tape damper, using a towel, or simply laying their wallet near the rim. Permanently installed cloth or felt often steals too much tone.

Another drum-roadie trick is the A-shaped ring. Either purchased as a drum shop product, or made from old heads, it creates a similar effect by dampening overtones. O-Rings and gaffe tape have been replaced by MoonGel, a stamp-sized sticky pad of blue gel plastic sold in 4-packs. Old drum techs will tell you they were using Blu-Tack adhesive art putty long before MoonGel arrived a decade ago.

MoonGel is drum magic and every microphone kit should have some in the same slot with the drum key.

Microphone angle and distance make a difference.

Too close and the snare sounds more like a tom as proximity effect emphasizes the drum’s tone.

Too far and isolation is lost as other drums and cymbals are heard clearly in the snare channel and sent to any effects employed.

The rule of thumb is two fingers between microphone and head.

An angle helps the microphone hear the buzz of the snare strainer from below when protruding slightly over the rim.

Old-timers swear if you position the microphone correctly on a welltuned head, catching the strainer from below with a second microphone isn’t necessary.

WHAT’S ON SECOND?

That said, secondary snare input channels are those used with—as well as those used instead of—the main snare microphone. The time-honored “snare under” microphone is frequently chosen as a condenser or a bright dynamic to better catch the snap and crackle of the strainer, though it’s equally common to simply use the same model as the main microphone.

Reversing the polarity of the “under” microphone combines it in-phase with the primary microphone, which hears the drum from the opposite direction,

Another alternate snare input is the “side-stick,” commonly used for ballads, jazz or Latin music, where the drummer lays the stick across the head and lifts it to strike the rim on the right side, The sound is different enough that a second input is often needed to correctly EQ and add the right reverb effect.

The position for this alternate input is at one o’clock instead of ten o’clock, and is dictated by the practicality of getting a microphone in close where the stick strikes the rim. It’s not necessary to get the
microphone over the head and successful positions include some beside the shell.

Side-address capsules allow the microphone to sneak up beside the snare from a number of angles because its vertical body more easily fits between the snare and the rest of the kit.

Miniature microphones or lavaliers are often used for this application so they don’t interfere with drumming, but they must be properly shock-mounted from the drum. Some mic the snare shell if it’s wooden or the top if it’s metal, while others prefer the sound of mic’ing the air hole.

Brushes also require a different microphone and a condenser helps to bring out their subtleties without resorting to drastic EQ.

The use of an alternate microphone and channel allows entirely different EQ, dynamics and effects to be used by simply changing channels.

There are countless schemes for using multiple snare microphones with different effects, depending on the music, and it’s not unusual for there to be several dedicated snare effects, some driven from a different microphones.

I’ve also seen engineers simply Y-split the same microphone to two channels to achieve this. In smaller venues, where the drum kit and especially the snare - seems too loud, the best use of a snare microphone is mostly as an effect send, due to the amount of snare getting in other microphones.

A DOZEN WAYS

Here are a dozen favorites for engineers looking for a new snare microphone.

Man-made magnets begat the Beta 57, along with the Beta 58, and the tighter super-cardioid pickup pattern provides greater gain-before-feedback and better isolation on louder stages.

The Shure Beta 56A is a Beta 57A with an integral 180-degree swiveling shock-mount stand adapter that takes the place of a microphone clip; a design borrowed from the swiveling Unidynes of yesteryear, and carried over to the Beta 52.

The cable connects beside the stand, making it easier to get into position. like the 546, it also makes a great drum vocal microphone on a boom or gooseneck.

When AKG reinvented the C451 a few years ago, they recreated an old standby for hi-hats, brush-work and the underside of the snare drum, retaining the characteristics of the CK-l capsule, while improving the electronics.

The new one-piece transformer-less design helps clarity and provides higher output than the older modular versions.

Any setlist with an emphasis on brushes benefits from a 451 on snare plus it also stands up to sticks with a 10 or 20 dB pad. Its slim body and cardioid pattern allow it to capture a snare at several angles and leakage from the hi-hats matches evenly when another 451 is used.

Take 2: One singer’s preference for the C535 has led me to try it on various instruments over the years, including snare, and it’s probably the best all-around utility microphone ‘for acoustic music. It has a less pronounced presence peak than the 451, and a simpler integrated 4-way pad-and-HPF switch.

The Audix i5, like the D6 kick-drum microphone, has a loyal following. Its tapered, cast zinc alloy body houses a transformer-less low-mass cardioid dynamic capsule that provides a full sound that’s not overly bright. Its frequency response is sufficiently contoured to stand up in the mix without covering up the vocals.

Due to a generous double row of side vents, its off-axis response is smooth and its polar response is surprisingly tight for a cardioid. This microphone is designed to rock, and it also works well on conga.

Audio-Technica has been producing innovative microphones for years, with the ATM450 being a unique side-address pencil-condenser that is more easily positioned above or below the snare drum, especially on tightly-spaced drum kits. It has an 80 Hz high-pass, a 10 dB pad and comes with a rubbery clamp that provides additional shock-mounting.

One ‘80s studio snare trick was to tape a pencil condenser and a dynamic microphone together to time-align them. The process was to first put them out of polarity, position them for maximum cancellation, and then return the polarity to normal and tape them—now easily adjusted on digital consoles using channel delay. The two microphones combine to make a fuller sound, with the condenser providing more attack and the dynamic helping the fundamental tone.

Another notable microphone in the Audio-Technica Artist Series is the ATM250DE, a more affordable version of the Artist Elite AE2500 dual-element design. It has both a hypercardioid dynamic and a cardioid condenser in the same chassis, employing as-pin XLR. The condenser element has an 80 Hz HPF and a 10 dB pad.

Though designed for kick, it provides the benefit of a time-aligned set for the price of the individual microphones.

beyerdynamic introduced the M201 instrument microphone over three decades ago, yet engineers are still discovering them for live sound as well as recording. This hypercardioid dynamic workhorse performs well in a wide range of applications, and its smooth, accurate response is typical of a condenser, which it resembles.

CAD Audio introduced the Equitek e60 cardioid condenser instrument microphone a few years ago, and it went somewhat unnoticed until recently. It has a gentle, 6 dB-per-octave high-pass filter, selectable as 40, 85 or 122 Hz, plus a 10 dB pad. A unique body design has an integral mic-stand mount with its XLR connector at an angle, allowing for easier placement.

DPA introduced the 4091 (and the 6 dB more sensitive 4090) for users who wanted their omnidirectional lavalier microphones - first choice for Broadway body microphones, as well as close-mic’ing symphonic strings - but needed the preamp and capsule in the same housing to provide a rugged and familiar form-factor that mounts on microphone stands.

In applications where omni polar response is acceptable, it provides DPA-quality in an affordable package that can also be used as a measurement microphone.

Electro-Voice introduced the original N/DYM 408 “egg microphone” in the late ‘80s, and the ND468 is an update. The first microphones to use neodymium increased frequency response while producing higher output using man-made magnets, and they’re known for a clear, natural sound on loud sources.

The unique swiveling ball makes placement and tweaking easy. They remain popular for toms and they’re equally effective on snare top and bottom.

Oktava is perhaps the oldest transducer manufacturer in Russia, but didn’t emerge until the fall of the iron curtain. The MK219 is a large-diaphragm fixed-pattern cardioid condenser, with reed switches for its 10 dB pad and high-pass filter. Its rugged cast body makes it a great crossover product for live sound. Though cosmetically challenged, its side-address design makes it a great snare microphone.

The Sennheiser e905 dynamic cardioid snare microphone provides a smooth frequency response that tilts towards the highs with a slight presence peak. And, possibly the best dynamic instrument microphone ever, the MD441 has a supercardioid pattern, a bass roll-off, a ‘brilliance’ switch and long, slender form-factor.

 

{extended}

Friday, May 11, 2012

Wireless Primer: The Key Issues Of Digital Audio Transmission

Issues of sound quality, data rates and more

Digital is a buzzword that many presume solves all the technical issues we face today.

More and more digital equipment, such as mixing consoles, audio signal processors, and the like, are used for several applications, as a digital audio signal chain offers many advantages.

A digital signal on a wire (i.e., fiber optic cable) is easier to handle than on a copper wire because 48, 64, or more audio channels can be transported on one thin fiber optic cable. If an audio signal is already in the digital domain, it makes sense to keep it in this domain as long as possible.

As for digital wireless transmission, a digital wireless system is beneficial when the sound, occupied RF spectrum, and battery lifetime is as good or even better than an analog system. On top of this, latency (time delay between input and output) is always a very important topic to keep in mind.

Let’s start with sound and the related data rate.

The best sound can be expected if there is no audio data compression used in the wireless system. This will lead to a very high data rate.

• Minimum for 20 kHz audio and approximately 110 dB dynamic range: 18-bit 48 kHz = 0.864 Mbit/s

• Necessary overhead (framing, channel coding) leads to even higher data rate (factor approx. 1.5 to 1.296 Mbit/s)

When transmitting this high amount of data, it is no longer possible to use a simple and robust digital modulation scheme like FSK (Frequency Shift Keying) ASK (Amplitude Shift Keying) or PSK (Phase Shift Keying), because these concepts will be not able to fulfill the spectrum mask, 200 kHz of occupied RF spectrum, defined by the FCC. Even if this constraint didn’t exist, greater occupied RF spectrum could inhibit large multichannel systems.

To improve this, it is necessary to use a more complex modulation scheme with narrow filtering. The amplitude and the phase of the transmitted signal must be very precise when usmg this approach.

Behind every point of the constellation diagram, a digital word is deposited, which the receiver has to pick up and transfer back into an audio signal. This requires a very linear RF amplifier. This is a current-hungry device. The unwanted effect is reduced battery life of transmitters and portable receivers. By driving the RF amplifier with a better efficiency, the occupied RF spectrum will increase in an unwanted way.

If the data rate described above can be reduced, the modulation scheme can be simplified and the amplified RF can be used in a more efficient way to conserve battery power and increase operational time.

To reduce the amount of digital data, a compression algorithm has to be defined. This algorithm will add some latency to the whole data= transmission process. low latency is especially important during a live performance on stage.

If the total latency in a PA system, including contributions from digital mixing consoles, effects, etc., is greater than 10 ms, the timing of the band will be thrown off.

Further, if streaming video is projected to accommodate a large audience the picture and sound will be out of sync.

New audio data compression algorithms show good performance with a very low latency’, However, audio compression would introduce the possibility of audible artifacts (at least with awkward signals).

As technology improves, there will be solutions to the obstacles described above and digital will become available for wireless transmission.

The key questions for a digital system at this time are:

—Is data compression used?

—What RF spectrum is necessary and how will this impact multichannel systems?

—What is the latency of the system?

—What is the battery lifetime?

Volker Schmitt is a senior engineer for Sennheiser US, and Joe Ciaudelli also works with Sennheiser US and has a history of providing frequency coordination for large multi-channel wireless microphone systems used on Broadway and by broadcast networks.

{extended}

Church Sound: How To Win Over The Three Typical Problem Musicians

Working behind the mixer is more than mixing and doing the technical work

 
One of the hardest parts of our job is mixing sub-optimal sounds. 

The frustrating part is when it’s the fault of a musician and you and I don’t have the authority to say anything. 

Don’t misunderstand me…I’m not saying I demand professional level musicianship from the band. 

I’m saying there are times when they need to be better.

Let’s back up…

Recently, some articles have been focused not on the technical aspect of your work but on the human aspect of what you do. The first was a guest post from my pastor on his view of church audio and the roles of sound techs. The second article focused on how you can enlist the worship team to help you after the service. 

Keeping with the theme this week of purpose and teamwork, let’s look at working with problem musicians.

This month, I’ve received quite a few emails from people asking how they can deal with problem musicians. Mind you, these usually aren’t musicians who are intentionally causing problems. They are musicians, singers included, who have adopted poor habits or aren’t able to perform at the level expected of them. 

The result is you and I have a harder time creating a solid mix and the congregation suffers.

But what can you do?

As the sound tech, you don’t have authority over the worship leader or the musicians. You can’t tell them what to do or how to do it. That is to say, not in the ways that are usually ascribed to those in leadership. There are ways you can help, however, as you’ll soon learn.

My experiences

I’ve seen a few problem musicians myself. Thankfully, it’s been a long time since I’ve seen any more. But looking back, all of the problems could easily be corrected when the right steps were taken while keeping in mind their emotional well-being.

The three typical problem musicians

There are three typical “problems” I have seen with musicians. Their hearts are in the right place but they don’t realize the impact of what they’re doing.

1. The amp lover. The amp lover is the easiest to correct of the three. The problem they present is insisting their amp be the source of their audio feed (mic’d or not) all the while having it pointed at their knees and cranked too loud. 

They simply love their tone but don’t know how to make it work with the mix.

2. The stylizing singer. Singers are on the stage to do one of two things; lead the congregation or support the lead singer in the case of background singers. When they sing outside of their expected melody, then they are no longer leading or they are no longer harmonizing.

They simply sing more freely than they should and it’s hard on the other singers/congregation to follow along.

3. The double-duty musician. There are some musicians who can sing and play an instrument at the same time while doing both tasks very well. And there are those who can’t. A musician might say “God has blessed me with a great voice and a love for the [insert instrument] so I feel I should use both gifts at the same time.” Just because God blesses someone with two similar talents doesn’t mean he’s telling them to use both at the same time. 

Their heart is in the right place but when their double-duty results in doing one or both tasks poorly, then their sound suffers, other musicians have problems, and the mix suffers.

How you can change all of this?

Let’s start with the amp lover.

Your first level of assistance is showing them how they can point the amp up at their heads for a better sound.

Considering I’ve listed them in this article, it’s not going to be that simple. 

Let’s say the musician does move it but still cranks the amp. Or, they don’t listen to your recommendations.  Then what?

Talk with the worship leader. Explain the effect of the amp on the overall sound and its negative effect on the mix. 

More than likely, the congregation is already unhappy with the sound. Give them a week or two. If nothing has changed, record the next service and give them a copy of the worship set after the service.  They will hear for themselves how the amp is negatively affecting the mix and they’ll take steps to get that problem resolved.

Next, the stylizer.

Working with the stylizer is a bit harder. I haven’t found anything that I could say that would help. It was only in going to the worship leader that changes started to take place. Much the same way as with the guitarist, give them a copy of the worship set so they can hear the impact of the stylizer.

I suggest you go one step further. Recommend bringing in a vocal coach who isn’t someone who attends your church. Have that vocal coach attend a few services and then have them work with all of the singers.

This way, not only will the stylizer have a professional point out their issue and show them how to overcome it but the coach can help the other singers… and can’t everyone benefit from some level of professional instruction?

Finally, the double-duty musician.

This is the hardest of the three. Their heart is in the right place. My recommendation is that if you are good friends, sit down and have a heart-to-heart talk with them. A copy of the worship set might help but I’d focus on being honest with them.

You might say “whenever you sing lead for a song, your rhythm playing becomes inconsistent and the band feels it and the congregation hears it.”  Or, “whenever you sing while playing, your singing volume is all over the place and sometimes, whether you realize it or not, you’ll not sing a couple of lines when you sing harmony.”

I’ve had to deal with this issue with a musician and it took a long time until they finally accepted it and committed to doing only one of the two for each song.

If you don’t think this will work, talk with the worship leader about it.

One other option

Sometimes problems don’t need to be solved directly. Sometimes it’s a matter of having the person in the right mindset to see what they need to change. Consider having a bible study with all the tech crew and the musicians together with a study on worship. 

The result of the study can be an evaluation by each person as to how they work on the worship team to promote worship by the congregation and what they can do to improve.

In conclusion

Each of us, whether on the tech crew or on the worship team, are working for the congregation. Our goal is for them to worship completely and without distraction. Working behind the mixer is more than mixing and doing the technical work. It’s also about helping everyone reach that goal.

And, if you bring up one of these issues with the worship leader and it’s not dealt with over time, let it go. It’s their job to shepherd and lead the worship team, not yours.

Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.

{extended}

Real World Gear: Options Abound With The Latest Digital Consoles

Choice is largely a matter of personal preference, specific features and capabilities, and budget

The tide of digital consoles continues to rise. In general, they’re getting smaller and lighter, but overall capabilities are increasing at a steady pace.

Already this year we’ve seen several companies introduce new models and series, and they all look to be smartly executed packages, providing users with more. More I/O. More networking. More recording capability. More effects. And, well, just more.

This is an ideal scenario for the marketplace. If you’re more focused on the “look and feel,” there are options. If you’re more concerned about getting a lot of signal in, out and around a gig, there are options. If you’re focused on the live recording realm, there are options. If you’re looking at the coming AVB networking standard, there are options.

Significant upgrades can also come from the software realm, where the simple upload of a new software version can bring loads of new capabilities to existing hardware.

For example, Soundcraft recently released V2 software for the Si Compact Series, and it offers 23 new features, updates and enhancements, including eight additional DSP channels that expand the Si Compact 16 and Si Compact 24 to 32 and 40 inputs to mix, respectively.

Another facet that’s come of age is the ability to operate consoles remotely via laptop computers. Of course, that almost seems quaint now with the advent of apps, most of them available for free download, that foster control and mixing from smart devices such as the Apple iPad. Even further, the new Mackie DL1608 more fully integrates a digital mixer with an iPad.

Where did all of this come from? Almost 15 years ago, Yamaha introduced the PM1D, which launched the modern era of digital mixing consoles in live sound. While production of the PM1D ceased in late 2009, its impact continues to be mighty.

The development and deployment of digital consoles stretches back far further than that, however. Yamaha had already had some earlier success with the O1V and the O2R, and around that same time came the Innovason Senory, a production large-format digital console built specifically for the live market.

Soundcraft also came out with the Broadway, a digital control surface that would control analog input and output racks, most notably utilized for a tour by Celine Dion. Texas-based sound company Showco, one of the giants of the day, developed the Show Console, which had a digital control surface. These weren’t sold, only available for rent.

Then came the PM1D, and the rest, as they say, is history, as we now enjoy a time where digital consoles are proliferate, perpetuating the stream of development that’s possible with digital technology.

Enjoy this Real World Gear tour of a wide range of digital consoles available today. All of these models meet and exceed needs in various sound reinforcement applications, and at this point, choice is largely a matter of personal preference, specific features and capabilities, and budget.

{extended}

In The Studio: Record Great-Sounding Drums Using Only Four Tracks

If you are looking for an earthy, realistic kit sound that can easily be manipulated, try this out

Here’s a simple, common-sense method to record a great-sounding drum kit on only four tracks. I’ve always been a follower of the less-is-more philosophy, and this kit technique goes all the way back to my analog 4-/8-/12-track days when track economy was a must.

There have, of course, been volumes dedicated to recording “trap kits”, from only two microphones to two mics on each drum! I think the concept of a drum kit as a group of separate instruments is off-base. A drum kit is an ensemble and has a “group sound”.

To me, using a mic on each drum is like multi-miking piano strings. And using so many mics can become a balance, EQ, and panning headache—as well as a phasing nightmare!

Now back to square one: What exactly are you looking for in a kit sound? Most pop, rock, R&B, jazz and country music requires a good separate kick and snare sound and maybe a separate hi-hat depending on the musical arrangement.

But how about all those tom-toms? They have essentially the same fundamental sound with different harmonics depending on shell size, diameter and tuning. If struck properly, they will have approximately the same volume level, and most pro drummers will do this subconsciously. So a simple mic technique can capture them well.

If you are looking for an earthy, realistic kit sound that can easily be manipulated, try this out. You’ll need one pair of stereo mics, one good snare mic, one good kick mic and last but not least, a good location for the drum set in the studio.

Setup
Location is very important and is an oft-neglected starting point. I prefer to set up the kit across a corner. A wall will do as well.

I use baffles from the floor up to about three feet, and approximately four feet wide behind the drummer in the corner. My baffles are low tech, made of 3-foot by 4-foot jalousie window frames with 6-inch wooden slats. There are four layers of wool moving blankets behind them. Thank you U-Haul! 

Other baffle materials that work well are cork faced bulletin boards with combos of styrosheet and blankets behind them. Portable office walls work well too.

The baffles reduce, but do not completely remove the resonances and reflections of the tom, snare and kick. We want some of these reflections for our kit sound.

Microphone Placement
Once the kit is set up with baffles in place, and tuned as you and the drummer want it, you can proceed with mic placement. It’s always a good idea to watch the drummer play for a while to observe where he places his hands and sticks while going around the kit. This will help you put the mics where he won’t hit them or have to move around them. We want him to be comfortable.

Walk up in front of the kit, put your head over the tom-toms, find a spot where the drums seem to focus, and listen for the toms and reflections off the corner. What you’re hearing is a larger percentage of top skins, some bottom skins and wall reflections.

I usually find this spot about two or three feet above the toms and two-thirds of the way over the toms. That’s fairly close but out of the drummer’s stick path. This will be the position for the stereo mics. I have been using a Crown SASS-P MKII stereo mic for this job for more than a decade. Any good pair of mics in a stereo configuration should work well.

Remember, tom-tom and snare spill is actually an important part of the overall sound. If you listen to the drum solo tracks on the Beatles Anthology CDs, you’ll hear a great example of this: Ringo’s Ludwigs drone along just beautifully in “Strawberry Fields”  Another example is Levon Helm’s kit on all The Band’s classics and—oh yeah, Atlantic R&B.

Once you’ve positioned the stereo mics, the rest is straightforward except for the optional hi-hat. For snare and hi-hat use a mic that has plenty of proximity effect. Position the mic at the snare-drum edge between the drum and the high-hat. This should keep the mic out of the drummer’s sticking path.

A Shure SM57 will work OK. I prefer a condenser, a Neumann KM84 or AKG 451 type. The Asian clone mics are recommended here.

You want the mic nice and close to exploit the cardioid proximity effect to get the snare drum “bulge” sound. I prefer positioning at a slight angle. The mic will pick up the high-hat thanks to leakage into the side of the mic.

Finally, the kick mic is whatever you’re comfortable with. Your criteria should be good low frequency response, excellent transients and—very important—ability to handle high sound pressure levels at low frequencies.

Find a sweet spot where you hear a definite increase in volume and tone. I use a Sony ECM 322 (an ancient cardioid condenser) inside the drum under a layer of blanket, about 4 inches away, parallel to the drum head. This is for a one-head kick drum. For two heads, I use a Neumann U47 FET in front. Here a large-diaphragm, Asian mic clone will also work fine, but be aware of room noise.

If you really need overheads for the cymbals, add them. If arrangement calls for a hi-hat played open and closed, use an extra mic.

Finally, record as hot as you can without clipping. This gives you the dynamic range needed for a good drum kit sound. I don’t recommend compression while recording. 

Mixing
That’s it for setup—now on to mixing where we will tailor—not create—our kit sound. As always, if you got it right in the recording, the mixing will be easy and breezy and not a time consuming chore.

If at all possible don’t mix immediately after tracking because of listening fatigue. It’s much better to come in fresh with the concept of simply getting the right sound.

I personally hate to mix after tracking. While recording I have a pure “techno Nazi” head: ears tuned for all the bad stuff, rattles, hum, clipping, pitch, meter, mistakes, etc.

While mixing, first listen with no effects, EQ, reverb, etc. Start with only the overhead pair. You should have a nice overall kit sound that’s almost usable by itself. Listen to the sound and use EQ to trim out any unneeded resonances.

Be careful not to cut into the floor toms’ fundamentals. If your board has minimal EQ, beg, borrow or steal a pair of graphics or parametrics. A single sweepable mid EQ will make life difficult here. You’ll need to EQ mids at more than one frequency.

Now’s a good time to add some reverb. You will use the drums to “trigger” the reverb and they will complement each other.

Don’t smother the drums with too much low-end EQ on either the track or the reverb. Always remember less is more! 

Listen for significant tom fills and cymbal crashes. You should be able to tweak low mids and mids and separate upper mids/highs for the crash cymbals. Also cut out high-end in the reverb. We don’t want any reverb on the cymbals.

Next listen to the snare/hi-hat mic. Again, start flat. Trim the bottom for unwanted room rumblings. Work for a big snare sound, usually found in the low mids and even the upper bottom. Add the reverb and work the two.

Next, on to our optional hi-hat. Many engineer/producers make the mistake of going too high in frequency looking for a hi-hat sound. Cymbals have a broadband signal and reach well down into the mids.

Look for an effective stick and brass strike and then add highs to sweeten the sound. Not too much! Make them peek through and you’ll have the real deal.

Is the sound O.K. now? The kit should sound sort of like Levon Helm and The Band. But what if that’s not what you want? Do you need more balls, more commercial sound, more funk?

Let’s whip out the compressors. Some compression on the toms will even them out and make them cut through nicely. Again, not too much. Play with the ratio. It’s a good idea to let them build to a threshold for more dynamics. I prefer RMS compression for a more natural sound.

Snare is more critical with compression but more fun. If the drummer is consistent in volume, you can use a higher ratio, but watch the threshold. Let it limit only the top. Watch the lil’ red lights and make ‘em dance to the beat. This way you can control the ring tone of the drum and make it really funky.

This same snare technique will work fine on the kick drum. You can control the attack and tone. Watch the bottom end with the EQ. Don’t overdo it or it will get lost when heard with the bass player. Look for EQ frequencies that separate the kick from the bass.

I use little or no reverb on kick drums. You want to trigger the reverb with the compressed signal from the snare and toms to get a naturally reverberant sound. This is exactly what a properly tuned and played kit in a decent acoustic environment would sound like…….with a little help from our electro friends.

Now you should have all the control you’ll ever need. You can raise and lower kick and snare independently as needed in the mix.

You can also pan the tom-toms as you like. I usually use a medium pan on the overheads. This way when the drummer plays a fill across the kit, it will bloom across the sound field and then settle down the way a real kit would—unlike with the hard synthetic panning of individual toms that always stay separated from each other. This naturally occurring sound will also help the drummer, as his kit will sound the same in the cue ‘phones as it does live.

I usually leave the kick and snare near center, but not on top of each other. It’s best to slightly separate kick and bass. Bass that is too far to either side is bad news for the mastering engineer.

During tom fills, the effect of natural buildup is due to the toms’ resonance enhanced by the corner walls and your compressor.

Tweaks For Different Genres
Try this technique when you have time and you’re not under a deadline. It’s well worth the effort. If you can nail it, it will work with little variation on many types of music.

Some suggestions:
- R&B: toms medium spread, kick and snare tight-panned.
- Country: tight-panned snare and kick, medium tom spread (just like R&B).
- Jazz: close-mike the snare and kick, mike the toms not too close, and use very little “room program” reverb.
- Doo-wop: mike very close for mono sound, and use little or no reverb.
- Reggae: mike snare and kick very close, pan toms wide, use tight EQ and mucho reverb.

Neat huh?  Good luck!

Ward Lionel Kremer is a lifelong musician, producer, and recording engineer, who cut his first hit at age 17. In the 1960’s he recorded and performed in the New York pop/R&B music scene with The Four Seasons, The Chiffons, Joey Dee, The Temptations, and Ike & Tina Turner. In the ‘70s he worked in the Miami music scene with TK records, KC & The Sunshine Band, George McRae, and The Ritchie Family. Ward also recorded and produced soca, reggae, and jazz festivals in Italy, USA, and Mexico. He did live sound and recording for Randy Bernsen and Ken Basman. As Ward says, “There’s no music I can’t appreciate if it’s performed with soul, sincerity and love!”

{extended}

The Audio Expert: Lies, Damn Lies, and Audio Gear Specs—Part 2

Many specs are incomplete, misleading, and sometimes even fraudulent

Jonathan: “You lied first.”
Jack: “No, you lied to me first.”
Jonathan: “Yes, I lied to you first, but you had no knowledge I was lying. So as far as
you knew, you lied to me first.” — Bounty hunter Jack Walsh (Robert De Niro) arguing with white-collar criminal Jonathan Mardukas (Charles Grodin) in the movie Midnight Run

When it comes to audio fidelity, the four standard parameter categories can assess any type of audio gear.

Although published product specs could tell us everything needed to evaluate a device’s transparency, many specs are incomplete, misleading, and sometimes even fraudulent.

This doesn’t mean specs cannot tell us everything needed to determine transparency—we just need all of the data.

However, getting complete specs from audio manufacturers is another matter. Often you’ll see the frequency response given but without a plus/minus dB range. Or a power amp spec will state harmonic distortion at 1 kHz, but not at higher or lower frequencies where the distortion might be much worse. Or an amplifier’s maximum output power is given, but its distortion was spec’d at a much lower level such as 1 watt.

Lately I’ve seen a dumbing down of published gear reviews, even by contributors in pro audio magazines, who, in my opinion, have a responsibility to their readers to aim higher than they often do. For example, it’s common for a review to mention a loudspeaker’s woofer size but not state its low-frequency response, which is, of course, what really matters.

Audio magazine reviews often include impressive-looking graphs that imply science but are lacking when you know what the graphs actually mean. Much irrelevant data is presented, while important specs are omitted. For example, the phase response of a loudspeaker might be shown but not its distortion or off-axis frequency response, which are far more important.

I recall a hi-fi magazine review of a very expensive tube preamplifier so poorly designed that it verged on self-oscillation (a high-pitched squealing sound). The reviewer even acknowledged the defect, which was clearly visible in the accompanying frequency response graph.

Yet he summarized by saying, “Impressive, and very highly recommended.” The misguided loyalty of some audio magazines is a huge problem in my opinion.

Even when important data are included, they are sometimes graphed at low resolution to hide the true performance. For example, a common technique when displaying frequency response graphs is to apply smoothing, also called averaging. Smoothing reduces the frequency resolution of a graph, and it’s justified in some situations. But for loudspeakers you really do want to know the full extent of the peaks and nulls.

Another trick is to format a graph using large, vertical divisions. So a frequency response line may look reasonably straight, implying a uniform response, yet a closer examination shows that each vertical division represents a substantial dB deviation. The graphs in Figures 1—3 below were all derived from the same data but are presented with different display settings.

For this test I measured the response of a single loudspeaker in a fairly large room with a precision microphone about a foot away. Which version looks more like what loudspeaker makers publish?

Test Equipment

“Empirical evidence trumps theory every time.”

Noise measurements are fairly simple to perform using a sensitive voltmeter, though the voltmeter must have a flat frequency response over the entire audible range.

Many budget models are not accurate above 5 or 10 kHz.

To measure its inherent noise, an amplifier or other device is powered on but with no input signal present; then the residual voltage is measured at its output.

Usually a resistor or short circuit is connected to the device’s input to more closely resemble a typical audio source.

Otherwise, additional hiss or hum might get into the input and be amplified, unfairly biasing the result.

Most power amplifiers include a volume control, so you also need to know where that was set when the noise was measured. For example, if the volume control is typically halfway up when the amplifier is used but was turned way down during the noise test, that could make the amplifier seem quieter than it really is.

Although it’s simple to measure the amount of noise added by an audio device, what’s measured doesn’t necessarily correlate to its audibility. Our ears are less sensitive to very low and very high frequencies when compared to the midrange, and we’re especially sensitive to frequencies in the treble range around 2 to 3 kHz.

To compensate for this, many audio measurements employ a concept known as weighting. This intentionally reduces the contribution of frequencies where our ears are less sensitive. The most common curve is A-weighting, as shown in Figure 4.

In the old days before computers were common and affordable, harmonic distortion was measured with a dedicated analyzer. A distortion analyzer sends a high-quality sine wave, containing only the single desired frequency with minimal harmonics and noise, through the device being tested.

Then a notch filter is inserted between the device’s output and a voltmeter. Notch filters are designed to remove a very narrow band of frequencies, so what’s left are the distortion and noise generated by the device being tested. Figure 5 shows the basic method, and an old-school Hewlett-Packard distortion analyzer is shown in Figure 6.

Intermodulation distortion is measured using two test tones instead of only one, and there are two standard methods. One method sends 60 Hz and 7 kHz tones through the device being tested, with the 60 Hz sine wave being four times louder than the 7 kHz sine wave.

The analyzer then measures the level of the 7,060 Hz and 6,940 Hz sum and difference frequencies that were added by the device. Another method uses 19 kHz and 20 kHz at equal volume levels, measuring the amplitude of the 1 kHz difference tone that’s generated.

Modern audio analyzers like the Audio Precision APx525 shown in Figure 7 are very sophisticated and can measure more than just frequency response, noise, and distortion. They are also immune to human hearing foibles such as masking (1), and they can measure noise, distortion, and other artifacts reliably down to extremely low levels, far softer than anyone could possibly hear.

Professional audio analyzers are very expensive, but it’s possible to do many useful tests using only a Windows or Mac computer with a decent-quality sound card and suitable software. I use the FFT feature in Sony’s Sound Forge audio editing program to analyze frequency response, noise, and distortion.

For example, when I wanted to measure the distortion of an inexpensive sound card, I created a pure 1 kHz sine wave test signal in Sound Forge. I sent the tone out of the computer through a high-quality sound card having known low distortion, then back into the budget sound card, which recorded the 1 kHz tone. The result is shown in Figure 8. (Other test methods you can do yourself with a computer and sound card are described in Chapter 22.)

As you can see in Figure 8, a small amount of high-frequency distortion and noise above 2 kHz was added by the sound card’s input stage. But the added artifacts are all more than 100 dB softer than the sine wave and so are very unlikely to be audible.

Low distortion at 1 kHz is easy to achieve, but 30 Hz is a different story, especially with gear containing transformers. Harmonic distortion above 10 kHz matters less because the added harmonics are higher than the 20 kHz limit of most people’s hearing. However, if the distortion is high enough, audible IM difference frequencies below 20 kHz can result.

Sadly, many vendors publish only THD measured at 1 kHz, often at a level well below maximum output. This ignores that distortion in power amplifiers and gear containing transformers usually increases with rising output level and at lower frequencies.

The convention these days is to lump harmonic distortion, noise, and hum together into a single THD + Noise spec and express it as either a percentage or some number of dB below the device’s maximum output level.

For example, if an amplifier adds 1 percent distortion, that amount can be stated as 40 dB below the original signal. A-weighting is usually applied because it improves the measurement, and this is not unfair. There’s nothing wrong with combining noise and distortion into a single figure either when their sum is safely below the threshold of audibility.

But when distortion artifacts are loud enough to be audible, it can be useful to know their specific makeup. For example, artifacts at very low frequencies are less objectionable than those at higher frequencies, and harmonics added at frequencies around 2 to 3 kHz are especially noticeable compared to harmonics at other frequencies.

Again, this is why A-weighting is usually applied to noise and distortion measurements and why using weighting is not unreasonable.

———————————————————————————————————————

1) The masking effect refers to the ear’s inability to hear a soft sound in the presence of a louder sound. For example, you won’t hear your wristwatch ticking at a loud rock concert, even if you hold it right next to your ear. Masking is strongest when both the loud and soft sounds contain similar frequencies.

Audio Transparency

As we have seen, the main reason to measure audio gear is to learn if a device’s quality is high enough to sound transparent.

All transparent devices by definition sound the same because they don’t change the sound enough to be noticed even when listening carefully.

But devices that add an audible amount of distortion can sound different, even when the total measured amount is the same. A-weighting helps relate what’s measured to what we hear, but some types of distortion are inherently more objectionable (or pleasing) than others.

For example, harmonic distortion is “musical,” whereas IM distortion is not. But what if you prefer the sound of audio gear that is intentionally colored?

In the 1960s, when I became interested in recording, ads for most gear in audio magazines touted their flat response and low distortion. Back then, before the advent of multilayer printed circuit boards, high-performance op-amps, and other electronic components, quality equipment was mostly handmade and very expensive. In those days design engineers did their best to minimize the distortion from analog tape, vacuum tubes, and transformers.

Indeed, many recordings made in the 1960s and 1970s still sound excellent even by today’s standards. But most audio gear is now mass-produced in Asia using modern manufacturing methods, and very high quality is available at prices even nonprofessionals can easily afford.

Many aspiring recording engineers today appreciate some of the great recordings from the mid-twentieth century. But when they are unable to make their own amateur efforts sound as good, they wrongly assume they need the same gear that was used back then.

Of course, the real reason so many old recordings sound wonderful is because they were made by very good recording engineers in great (often very large) studios having excellent acoustics. That some of those old recordings still sound so clear today is in spite of the poorer-quality recording gear available back then, not because of it!

Somewhere along the way, production techniques for popular music began incorporating intentional distortion and often extreme EQ as creative tools. Whereas in the past, gear vendors bragged about the flat response and low distortion of their products, in later years we started to see ads for gear claiming to possess a unique character, or color.

Some audio hardware and software plug-ins claim to possess a color similar to specific models of vintage gear used on famous old recordings. Understand that “color” is simply a skewed frequency response and/or added distortion; these are easy to achieve with either software or hardware, and in my opinion need not demand a premium price.

For example, distortion similar to that of vacuum tubes can be created using a few resistors and a diode, or a simple software algorithm.

The key point is that adding color in the form of distortion and EQ is proper and valuable when recording and mixing. During the creative process, anything goes, and if it sounds good, then it is good. But in a playback system the goal must be for transparency—whether a recording studio’s monitors or a consumer playback system.

In a studio setting the recording and mixing engineers need accurate monitoring to know how the recording really sounds, including any coloration they added intentionally. With a consumer playback system you want to hear exactly what the producers and mix engineers heard; you’ll hear their artistic intent only if your own system adds no further coloration of its own.

“The Audio Expert” by Ethan Winer, published by Focal Press (ISBN: 9780240821009), is available here. To read part 1, Audio Fidelity, Measurements, And Myths, go here.

 

{extended}