Article

Friday, July 03, 2009

Top-Flight Sound Reinforcement For A Raucous Week Of Concerts At Arizona’s Annual CycleFest

Pro Production Services handles the wild ride by combining experience and top components to serve a week of live performances by a roster of A-list artists

Fifty weeks out of the year, WestWorld in Scottsdale, Arizona is an equestrian center and special events facility, but during Arizona Bike Week (ABW), held in early April of this year, the ABW Cyclefest turns WestWorld into a raucous concert venue.

A 27,000 square-foot tent is erected on the premises and dubbed the HandleBar Saloon, and it’s where the majority of the concerts take place during Bike Week.

Arizona Bile Week, which actually spans 10 days, is an annual event that has taken place for the past 13 years in the territory in and around the city of Scottsdale and is the fourth-largest bike festival in the country.

Pre-Rally Days, the five days leading up to Cyclefest, also features concerts, rides and events – including the coronation of Miss Arizona Bike Week – that take place at various towns in the area. This year Pre-Rally Days started off with a concert staged at Chester’s Harley-Davidson in Mesa by the Charlie Daniels Band.

Pro Production Services has been handing production for ABW Cyclefest since 2004, and the Tempe-based company maintains a pretty diverse client base covering everything from fairs and festivals to corporate shows and even events for the White House.

Donovan Mote, director of operations for Pro Production Services out of their Phoenix office, relied on EAW KF750 loudspeakers and BH760 subwoofers driven by Crown I-Tech Series amplifiers to provide the wide horizontal coverage needed for the tent, as well as satisfy rider requirements for the bands coming through. This year’s lineup featured performances by hometown group the Gin Blossoms, Blackfoot, Eddie Money, and Cheap Trick, as well as performances by local bands.

Faced with the physical challenges of staging a large scale production in a tent, Mote has tried various configurations opting this year to go with ground stacked mains positioned in front of a 40- by 40-foot stage, expanded up this year from a 40- by 32-foot stage to satisfy set requirements for Cheap Trick. Crown I-Tech amps provided the power, and Rational Acoustics Smaart was employed by Mote primarily for time alignment of the system.

At front of house and monitors, both analog and digital consoles were represented. “This year we had a Midas XL200 (analog) at front of house, which is kind of our festival workhouse,” Mote says. “It’s really easy to use. There’s not always a lot of time for sound checks and there are charity raffles and things like that that happen in the tent where the concerts occur as well, so something that is really familiar to everyone is pretty important.”

A Midas Heritage 3000 (analog) console was provided for monitors, while Yamaha M7CL digital consoles were situated at both house and monitors. Mote continues, “It’s really kind of an ideal scenario for us, some bands prefer to work on the M7CL because they already had the cards or were more familiar with the console, so we were able to provide the headlining acts what ever they wanted to use.”

Both 12- and 15-inch Radian MicroWedges (the original Dave Rat design that the new EAW monitors of the same name are based upon) were employed on stage, again powered by Crown I-Tech Series, though as Mote recalls, “Several of the acts had in-ears, and we provided some. We used the Shure PSM 700s and the PSM 600s for hard-wired, and Cheap Trick brought out their own set.

“We just picked up Crown I-Tech 4000s for the monitor amp rig and that was the first time we used them all networked. It’s really nice to be able to set up a custom panel and see the status of all the amplifiers and know how much headroom you have.”

“I’ve got our laptops set up so I can monitor all of the (Harman Pro Group HiQnet) System Architect devices, and in this case, it’s the I-Tech Series and a dbx 4800 DriveRack at house,” he continues. “I can also monitor all the wireless systems, and I also have it so I can run either of the M7CLs remotely from the laptop, walking through the crowd making mix adjustments and so forth. To be able to do all that from one laptop is pretty cool.”

The microphone selection is diverse, including AKG C414s and Neumann KM 184s, Shure KSM 9s and a standard selection of Shure SM58s and SM57s. Sennheiser e908s handled toms with Sennheiser 421s for other drum needs.” We just bring out basically two full mic packages and they’ll have pretty much anything that they could want to choose from,” says Mote.

In addition to sound reinforcement, Pro Production Services also provided all staging, lights, video, and backline, with Mote explaining, “More and more these days clients want to have company that they can make one phone call to and have most of their needs, if not all of them, met. So we’ve been trying to diversify our business model to accommodate that.”

He estimates that 50,000 people attended the five days of Cyclefest. “This year there were two other really large festivals that were the exact same time frame, and I hear reports from the other two as well as Bike Week that the numbers were as high as ever and in some cases higher. So while the economy is affecting a lot of people, we’ve been real lucky.”

Arizona Bike Week Equipment List

Front of House Consoles
Midas XL200
Yamaha M7CL-48

Front of House Processing
Yamaha SPX-2000
Yamaha SPX-990
TC Electronic D-2
BSS DPR-404
dbx 1066
dbx 1074

Front of House Drive
dbx DriveRack 4800
Klark Teknik DN360
Shure P4800

Front of House Loudspeakers
EAW KF750/755
EAW BH760
EAW JF80
Mackie SRM450 (vendor area public address)

Front of House Amplifiers
Crown I-Tech Series

Monitor Consoles
Midas H3000
Yamaha M7CL-48

Monitor Processing
BSS DPR-404
BSS DPR-504
Yamaha SPX-990

Monitor Drive
Klark Teknik DN360

Monitor Amplifiers
Crown I-Tech Series

Monitor Loudspeakers
Radian MicroWedge
EAW KF600i
EAW SB250
EAW SB1000
Buttkicker
Shure PSM 700 wireless IEM
Shure PSM 600 wired IEM

Microphones
Shure UHF-R wireless
Shure
Sennheiser
AKG
Audio-Technica
Neumann

Miscellaneous
Motion Labs Distro
Rational Acoustics Smaart
Harman HiQnet System Architect

Mark Johnson has been involved with audio and video for more than 35 years, including production, manufacturing and writing for various publications.

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Separate For Sanity: Mixing Monitors From Front Of House

Making the situation tenable for both ourselves and the band

If you’re a front of house mixer, here’s what you need to know about monitors. If asked to adjust monitor settings, look the person directly in the eye and say, “I handle front of house; I don’t do monitors.”

Speak it with authority and perhaps add a bit of disdain, then turn quickly and as you’re walking away add, “let me find the monitor mixer for you.”

At this point your job is done, unless of course, you actually happen to run across the monitor person. If so just say, “someone on stage is looking for you.”

Now let’s return to the real world.

Years ago, a band could be booked at a club three or four nights in a row; today, most clubs rotate bands on a nightly basis. This means the sound system has to go up and come down as quickly and easily as possible, including the monitors.

For those of us who do this system “up and down” work in a different club every night, a large monitor rig is neither time nor cost effective, especially without roadies.

While even smaller shows (think clubs) are now outfitted with a separate monitor world, a lot of mixers still must run monitors from front of house (FOH), usually due to necessity. And in my experience, it’s by far one the hardest things to do.

It doesn’t take very long to realize that regardless how good the house mix, your gig is insane if the monitors aren’t right. Between feedback and band member complaints, it can make for a long night.

A big drawback in handling monitors from FOH is often the limitations of the console – the majority used in club situations offer just two separate monitor sends, meaning that for a four-piece band, someone is sharing the same monitor mix.

For obvious reasons, a separate mix to each member of a band is preferred.

However, many of the consoles we use are also outfitted with several aux sends that can be used as monitor sends, thereby increasing the number of separate mixes on stage.

Keep in mind that when doing this, it’s important to use the pre-fade aux sends. Otherwise, any time house level is increased by using the fader that is on a post-fade aux, the monitor level will increase as well.

Most musicians want their monitors loud to begin with, so any further increases in level can result in feedback. So by running pre fader, any adjustments for the house will not affect the monitors.

This is not the case for the EQ section and gain control of every channel, where even running pre fader, everything else on the channel affects the monitors.

For example, let’s say you want a little more 3 kHz in a vocal out front - adding it via your channel EQ will also add it to the monitor as well. And this may not be favorable to whoever is getting that particular monitor mix; it may sound too thin and be on the verge of feedback.

By the way, one way to combat this is to put a separate EQ in line of each monitor send. 

In A Good Way?
Now, with all of this (and more) going on, the house still needs to be mixed, and keeping in mind that almost anything done to the mix out front will almost certainly affect the monitor mix. Will it affect it in a good way? Dream on!

Therefore, I threw in the towel years ago and came up with a separate monitor system that is cost effective, mobile, easy to set up/take down and keeps the musicians happy. And, it works well if you work with one particular band or several.

My gear list includes a 16-channel rack-mount mixer, two dual-15-band equalizers, a 16-channel rack-mount snake, a 1/4-inch patch bay and a rack on wheels.

Of course, depending upon need, the size of the console and snake can vary, and compressors, gates and effects units (and so on) can be added. Digital consoles also provide a lot of the comps, gates, effects and more that used to require separate outboard gear.

The main thing is to try to keep it all in one reasonably sized rack and pre-wired as much as possible.

I roll this on to the club stage, and then run three sub snakes: stage right, stage left and drums. These sub snakes plug into the splitter snake, which is mounted in the front of the rack.

The patch bay is also mounted at the rack front, and it provides sends to the monitor wedges, which are powered, by the way. (There are some amazing powered monitors these days.) When using non-powered wedges, I simply route from the patch bay to the monitor amplifier rack.

The only thing left to do is plug the splitter snake into the FOH snake, and then plug in the power. Having all connections at the front of the rack makes for fast and hassle-free interconnect, one of the big objects of this exercise.

When the band arrives, I dial in this monitor system for them, and then leave the rest to them. Each person in my band is familiar with this rig, and makes adjustments themselves throughout the night as needed.

If working with a different band, I still dial it in at the beginning of the night, with any adjustments made quickly during a break.

Driving Them Crazy
This, of course, brings me to the point that the monitor system needs to be as simple to operate as possible, and thus, it’s the reason I don’t include extra effects units.

Our drummer, however, does want some effects in his monitor, so I set him up with a small rack containing a rack-mount mixer and the effects unit he prefers. His monitor send routes to this rack and then to his wedge.

As you can imagine, the drummer was very happy about this. But the addition of effects to the vocals, guitar and keyboard in his mix drove the rest of the band crazy, because they could now hear reverb on things they didn’t want to hear it on. (Sigh…)

To fix this problem, I send two mixes from the monitor mixer to the drum mixer. One mix has the vocals and instruments, the other just the drums. Effects are applied to the drum mix only, with the other mix remaining “dry.”

Hurray! There’s nothing like a relatively content band, and we’ve been doing it this way long enough for the drummer to get quite good at changing his settings with his drumsticks.

I’ve yet to meet a FOH mixer that has expressed any sort of strong desire to mix monitors, and truth be told, most of us originally got into this line of work with the goal of working FOH.

So to all of you who take on the vital role of mixing monitors, simply, “thank you.”

And for the rest of us who are required to try to come to grips with doing both within the same show, remember, a little separation – as in separate systems – can go a long way to making the situation tenable for ourselves and the band.

With more than two decades of experience working with sound, Tim Andras is the mix engineer for the Tampa-based band Stormbringer and at Harborside Christian Church.

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Magical Mystery Spec: Why Is Distortion Overlooked In Loudspeakers?

The sooner we can begin to understand typical distortion in these devices and its effect on our work, the sooner improvements in performance can be made that can result in better sound

Almost everybody in the pro audio business is a specification or “spec” junkie. From folks on the recording side to the sound reinforcement practitioners, we all want every detail of the technical performance of every piece of equipment in a system.

Specs seem extremely important, particularly those that represent a numerical improvement from past products, which are heralded as an advantage. 

But man cannot live by specs alone. Quite often, those who turn to specifications to answer questions should instead be using their ears for the answer. You see, specs are all too often taken at face value, whether they’re actually relevant or not.

Manufacturers sometimes take advantage of this propensity to over-emphasize specs (and who can really blame them?) but sometimes it comes at the expense of real performance and value. Why is this phenomenon so prevalent in our industry?

The most straightforward answer is that it’s a by-product of a craft that has a highly subjective final product. It would be nice if we could judge the performance of the work via strictly objective measurement. But it’s not that easy. Therefore, the next best thing for sound professionals to do is point at the specs of as a validation of choices, system designs and configurations, and mixes.

No audio equipment spec is more revered than distortion. (O.K., a bit dramatic, but work with me.) Distortion is measured to two and sometimes three decimal points on almost every piece of equipment in the signal chain. From the input of the console to the output of the amplifier, every fraction of a percent of distortion is stressed.

Sure, this makes some sense from the standpoint that the sum of distortion throughout a signal chain can add up to something that could be easily heard and perhaps detrimental to audio quality.

But here’s the key question: What audio products do we rarely see distortion specs for? Transducers! Yes, somehow microphones and loudspeakers slip right under the distortion radar.

In many cases, I suspect that those who design and produce these products never even measure distortion. Leading to our next question: why? My best guess is that distortion measurements are omitted because they would be shockingly “substandard,” to put it kindly.

Mics are the first “acquisition point” for what is fed to a sound system. These electromechanical devices convert acoustic energy into electrical energy. We’re starting to see some digital mic concepts emerge, and some of these offer designs with the potential to lower distortion and therefore improve accuracy.

That said, I believe that distortion in mics is fundamentally different than distortion in loudspeakers. Mic distortion is part of the acoustic signature (or “sound”) of the device, which can be an important part of the creative process.

For example, a mic can “color” a source sound in a highly pleasing way, and very importantly, this applies only to that particular source, leaving the rest of the mix unaffected. Therefore, I tend to think that any distortion spec for a given mic would not be all that critical in telling us much that is meaningful about its performance. Further, this “data” likely would be very hard to qualify.

Not Talking Fractions
Loudspeakers, however, are a whole different ballgame. The measured distortion produced by many loudspeakers is quite high indeed - we’re not talking fractions of a percent, rather, figures in the high single digits and even double digits.

Simply, this is a huge number compared to the total sum of electronic distortion that is created through the signal chain in a typical system, and perhaps most significantly, it applies to all output, not a single source.

In other words, loudspeaker distortion colors the entire mix. And it is even more detrimental because the distortion can wildly vary at different points throughout the coverage area of the loudspeaker. Further, it can vary throughout the bandwidth of the device.

As an industry - collectively - we’ve never demanded that manufacturers provide distortion specs on loudspeakers for comparative purposes. In addition, no manufacturer has ever made distortion the cornerstone of a marketing campaign.

To do this would require pointing out the fact that loudspeakers are, by their very nature, relatively high distortion devices, in turn prompting the need for real education on why a distortion spec matters, and also, why lower is better. The only realistic way that distortion in loudspeakers can be addressed: publish the specs and then work to improve on them.

Therefore it might be time, as an industry, for us to start to paying much closer attention to the concept of distortion in transducers, and specifically, loudspeakers. The sooner we can begin to understand typical distortion in these devices and its effect on our work, the sooner improvements in performance can be made that can result in better sound!

Michael MacDonald has been involved in the professional audio industry for more than 30 years. Beginning as a freelance mixer/engineer in the 1970s, he transitioned to working for manufacturers and has been employed by, developed products for, and consulted with major companies, and currently is serving as a VP for Harman Professional.

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Thursday, July 02, 2009

The “Vibe” Is The Thing: Monitor Wedges Or In-Ear Monitors Or Both?

Insights, clearing up misconceptions and approaches on monitor wedges and in-ear monitoring systems

The first time I saw a show where stage wedges weren’t used as the primary monitoring system was a Steely Dan tour in 1993.

The rumor at the time was that founding players Donald Fagen and Walter Becker only agreed to tour if their long-time studio engineer and “audio wizard” Roger Nichols agreed to mix monitors, including in-ear monitoring systems (IEM).

I attended a rehearsal with a couple of colleagues (all audio engineers), and during a break, I asked Nichols about the rumor, which he confirmed to be true.

As a result of that experience, the importance of the role of the monitor engineer dramatically increased in my eyes.

Also during that rehearsal, I noticed that there were some microphones at the front edge of the stage pointed toward the audience. Nichols told me they were there so the musicians - again, using IEM - could hear the audience and get a sense of how they were being received.

In other words, without that additional input, they’d think the audience wasn’t digging the show. Interesting! Of course this is old hat now, but 12 or so years ago it was news to me.

Over the intervening years, these and other related concepts have me thinking about how the role of the monitor engineer has changed.

To get up to date on the subject, I contacted a number of monitor engineers who have worked with both wedges and IEM, including Ian Beveridge with Foo Fighters, William Miller with Josh Groban and Chris Sharp with Rob Thomas. Each provided insight on the topic while dispelling some of my misconceptions.

Although so many artists have moved to IEM during the past decade, there are still plenty that still prefer wedges and sidefills.

And many acts use a combo of both. Beveridge, in fact, notes that only one band he’s worked with over the years has been completely on IEM.

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The only common element I could find is that for many “more experienced” performers, particularly in the rock ‘n’ roll genre, wedges are the norm.

Obviously, artists in this genre want to interact with the audience, and they also have performed for decades using wedges, so it’s their “comfort zone.”

Further, they want to “feel the vibe” of the show, and frankly, this is easier with wedges because the audience, backline, mains, wedges and fills all forge a cohesive overall soundscape for artist and audience alike.

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Miller puts it this way: “In a traditional wedge situation, your job as an engineer is to augment the world on stage. There’s already some sound happening up there, some acoustic energy. In other words, there’s sound present before you ever switch on your console.”

There are several reasons for the trend toward IEM and away from wedges. The most obvious is leakage, i.e. loud volume on stage makes it more difficult to mix because of bleed into the stage mics, reducing isolation and therefore control. And, feedback (or at least the potential for it) is always a cause for concern.

That said, Miller and Beveridge both feel that the job can still be done just as well with the traditional tools. According to Miller, “Feedback is most often caused by improper gain structure, misplaced monitors and poor equalization.”

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Beveridge adds: “Modern mics and loudspeaker technology allow you to make a very good sounding conventional monitor system with whatever coverage you like - sound quality does not have to be sacrificed for volume.”

But on the issue of stage coverage, Miller doesn’t completely agree. “Assuming no RF (radio frequency, with wireless systems) problems, an IEM mix is going to sound relatively the same no matter where you are on stage,” he notes.

“Obviously this isn’t the case with wedges. However, on a recent tour I found that by using line array loudspeakers as sidefills we could really provide the ‘sonic glue’ to the stage sound. The vertical consistency certainly helped to widen the ‘sweet spots’ on stage and provided a fuller sound when performers weren’t right in front of their wedges.”

Performers are also interested in protecting their hearing and feel that this can be better accomplished with IEM. Sharp adds: “It’s always nice to start off with your lead singer on ‘ears.’ It makes life easier at FOH when the engineer doesn’t have to fight stage wedges blowing into mics.”

Finally, the use of IEM can result in less stage “clutter,” which is desired by some artists and production designers in addition, to providing added space for things like more elaborate dance sequences.

To all of this, Miller lends a bit of perspective: “Now, performers don’t have to make a choice between good sound and a massive set. Artists have always used the entire stage for performances - think of Michael Jackson and The Rolling Stones in the 1980s - but with IEM, artists can take their mixes with them wherever they go on that stage.”

More Responsibility
Because IEM systems for touring are generally wireless, monitor engineers are sometimes faced with the need for more of a working knowledge of RF issues, frequency coordination and wireless system maintenance.( In addition to often being responsible for vocal wireless mic systems.)

Due to the added challenges of more wireless, one misconception can easily be dispelled, which is that using IEM saves time in comparison to setting up and “ringing out” a wedge/sidefill system.

“When you have as many wireless channels as I was dealing with on a recent tour, tuning for clear frequencies took as much or more time than tuning wedges,” Miller explains.

Sharp agrees, adding: “You must do your homework and know what frequencies that you as well as other bands are using at a particular gig site in order to avoid train wrecks.”

Most wireless manufacturers offer resources to help in this regard, and fortunately, current systems provide both RF and sound quality that is better than ever.

Longing For Better
During my touring days, I was well aware of how much difference that mics could make in terms of vocal quality, stage bleed and resistance to feedback.

For example, it’s not wise to use a single wedge directly behind a hypercardioid mic; rather, far better to use two wedges, each placed at the “nulls” of the hypercardioid pattern.

At the time, my choice for vocals was usually a Sennheiser MD431, an “old-school” high-end dynamic mic that sounded great and was very good at resisting feedback.

But still, I longed for something better, and the condenser mics I tried at that time sounded a bit harsh for my tastes, plus they tended to pick up too much stage wash.

Later, when I worked with Neumann, I was part of the team that developed the KMS105 handheld condenser. And although it picks up more stage wash than most dynamic mics, the sound is quite good with certain singers, particularly jazz vocalists where excessive stage volume is not the norm.

Additionally, at the time of its creation, the idea that vocalists were moving toward IEM was already in play.

Over the past decade or so, there’s have seen a sort of renaissance with respect to mics, a renewed passion for new and improved designs.

“In the past, the subtleties of mics were much harder to detect, but with higher fidelity concert systems, everyone is looking for the very best sound possible. And the best sound starts with the right microphones,” Miller notes.

In addition, these newer mics are more rugged and less expensive than they used to be, and condensers designed for both live applications and the rigors of the road are in abundance.

Beveridge also considers mics to be part of a larger picture of what’s happening on the stage: “Spill must be given a lot of consideration. What may make a snare-top sound amazing may absolutely destroy the rest of the drum sound.”

For Dave Grohl’s vocals, Beveridge uses a MD431 mkII, a new version of the old classic, for its flat response, lack of distortion regardless of level, good rejection and stability against moisture and humidity.

“It’s a good example of modern materials technology applied to an older design,” he says of the MD431 mkll. “The Neumann 105 is an amazing sounding mic but it would be useless for the Foo Fighters because of the amount of ambient sound it collects. We used a Crown CM310 for Kurt Cobain because of its amazing rejection.

“Both mics are condensers but at the opposite ends of the spectrum as far as picking up stage bleed.”

Most Important Thing
In talking to these highly qualified monitor engineers, I’ve come away with the impression that the “vibe” is the most important thing with monitor mixing, whether it’s done with IEM or wedges or both. But with IEM, there’s more of a mix challenge.

“In an IEM situation, there is no vibe - the mix engineer is responsible for building the artist’s entire world,” says Miller. “In my view, this is what separates a mix where you can simply hear everything from a mix that is exciting, dynamic and inspiring to the performer.”

Of course, there’s an art to creating this world, and everyone approaches it differently. What I saw with Steely Dan back in ’93 was a pair of AKG C3000 mics, one on each side of the stage, pointed out towards the audience.

Several years later, while touring and recording what would be the Mirror Ball album, Sarah McLachlan and band were outfitted with a combination of different mics at the edge of the stage, in combination with shotgun mics hanging from a front stage truss.

I found this interesting, because it reminded me a lot of how large audiences are mic’d for events like award shows. Generally, it’s done to pick up the overall “wash” of audience response, but is also key in getting some of the individual clapping and vocalization of audience members to provide the sound some added immediacy.

“Don’t just set and forget audience mics, place them with the same care you would for any other mic,” Miller cautions. “In fact, I’m more obsessive with audience mics than most others I use on stage because they can make or break your performance.

“With Josh Groban, I’ve used up to six and am considering taking more next time. Covering a crowd 17,000 is difficult, but you have to make a huge audience sound huge.”

Miller also employs some additional tricks, sharing one in particular. “While I don’t time-align stage mics, I find aligning audience mics to be absolutely critical, especially when some of them are placed at FOH, which I tend to do,” he explains.

This parallels nicely with something that Sharp points out, which is the importance of time alignment for wedges and fills. “I time align my center wedge mix to my sidefills, and am careful about the quality of my mixes without having to over-EQ,” he says. “’Ears’ take away the time-alignment issue, at least for the most part.”

Not As Expected
The world of monitors has definitely changed since IEM came along, but not always in the ways I had expected. With the addition of audience mics and wireless systems, running monitors on IEM is a more sophisticated operation than the standard wedge/side fill method – of course, with some notable exceptions.

At the same time, with proper gain structure, mic placement and loudspeaker placement, there’s no reason to abandon wedges.

However, I do think that with the proliferation of better quality wireless, mics and PA systems, as well as more demands from artists, there will be more using the IEM approach. By the time we get it perfected, naturally, some new technology will come along.

Digital wireless cochlear implants, anyone?

Karl Winkler is Director of Business Development for Lectrosonics and has worked in professional audio for more than 15 years.

More articles by Karl Winkler on PSW:
Back To Basics: Seven Habits Of Highly Effective Sound People
Steps You Can Take To Improve Your Mix Right Now
Things I’ve Noticed About Working With Sound, And What They Might Mean
Top 10 Reasons For Bad Sound (And What You Can Do About It…)

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Product Review: DiGiCo SD8 Digital Mixing Console

What can this new digital console offer your church sound system?

DiGiCo has come out with a steady stream of digital consoles utilized in a wide range of sound reinforcement applications, including larger church sound reinforcement systems.

The company’s most recent development, the SD8 digital console, has 60 stereo or mono channels available (the equivalent of 120 channels of DSP)—plenty for most churches—and certain key functions are available on all channels all the time, in contrast with some digital consoles that don’t offer consistent processing on all channels.

How does it work, and how well does it work? Is it worth a look?

Click here to check out John F. McJunkin’s in-depth audio product review of the DiGiCo SD8 digital console featured in the June 2009 issue of Church Production, and also featured on http://www.churchproduction.com

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Wednesday, July 01, 2009

The Accidental Electrician: A Tale Of Eliminating Dreaded Sound System Hum & Buzz

The primary point in dealing with ground loops: treat the causes, not the symptoms, all as part of the quest for ultimate system quality and performance

There’s no better feeling than when you’ve set up your system and powered it up to find it lacking any noise. 

This is no major feat when you have your own electrical distribution, but when you’re relying on the house or festival distribution, things can be a bit more dicey.

The classic scenario when the system powers up with a hum is for the operator to declare “ground loop!” followed by muttering and cursing while digging in the accessories box to locate the power cord cheaters.

I find it funny how a device that’s supposed to be used to provide an electrical ground connection on older two-prong outlets is most often employed to lift the ground instead.

But I often see that the whole ethereal concept of a “ground loop” is a distraction from one big fact: not every hum is from a ground loop.

What seems to be nearly universal is that a lot of folks don’t understand that the existence of a ground loop is not actually the cause of a ground loop hum.  A ground loop is an existing condition that is exploited by the true problem, which is an electrical current flowing through the loop.

Just so we’re all on the same page, a ground loop occurs when there are multiple ground paths between two pieces of gear. Typically, these paths are through the electrical ground, the audio wiring shield, and through shared chassis contact when together in the same rack.

Carefully designing a system to not have ground loops is certainly a valid engineering endeavor, which goes to my primary point: treat the cause, not the symptom.

An Interesting Case...
I had one system upgrade project at a mid-sized church here in Washington state where the sound system had a buzz problem that historically had not been solved.

When the original system contractor and I first walked into the sanctuary, the buzz was immediately apparent. The contractor turned to me and said, “Oh by the way, we’re going to solve their noise problem too.” Which was to say, “You need to fix this because I can’t.”

I hemmed and hawed about how the problem could be from the transformer on the pole, but was actually just making stuff up out of thin air, aggravated about this additional time-consuming task being added to my already conservatively budgeted schedule.

But I got on with acquainting myself with the system. Much of it had been obtained as surplus from the 1986 World’s Fair in Vancouver. I was surprised to find a rack filled with Bryston amps, a few years of warranty left and working just fine!

What disturbed me was that everything at the amp rack was plugged into “cheaters” and Radio Shack power-line RF filters. Someone wasted a pocketful of cash to buy those, and it was simply very bad AC practice to boot. 

This led to fix number one: I removed all cheaters and filters, and the molesto mucho buzz was transformed into a simple hum. A step in the right direction.

No other obvious conditions that might be causing the hum could be detected at that point after eliminating the lighting dimmers as a source.

The Saga Continues...
Two days later, we were back, doing the first thing these projects often require: taking everything apart. As the week progressed and we slowly re-assembled the system, I got to a place where I could examine the power distribution. 

I strung an extension cord from one of the sockets at the amp rack to the front of house position, and then deployed a Wiggins (solenoid voltage tester) to check between the hot on the extension cord and the hot at the FOH outlets. 

As feared, it measured 208 volts, meaning that the two different power circuits were on differing phases. Time to root around in the breaker box…

The building had a modern electrical system, which is fortunate. I’ve run across some problems in systems that were on legacy electrical distribution, and short of violating code, there’s sometimes not a whole lot that can be done to fix a problem. 

Suffice to say that a dedicated electrical ground and a modern distribution system is imperative for safe, noise-free sound.

Four circuits in the breaker box were dedicated to the sound system, and sure enough, they were grouped together all in a column. This was a typical commercial three-phase breaker box with the phase alternating for every row. 

The shame was that I could tell by reading the written-over labels that originally, the sound system circuits had been on every third row. My guess is that a well-meaning electrician had thought it would be smart to group the circuits together during one of the church expansion projects.

So, fix number two: I returned the circuits to their original spacing.

It’s a good idea to have all the power circuits on an identical phase because power supplies can leak.  Float a piece of gear and you can usually measure a small voltage from the chassis to power ground. 

When the chassis is electrically grounded, this potential doesn’t magically evaporate, and under the right circumstances, it can actually help create a ground loop current. 

If there happens to be leakage from gear elsewhere in the ground loop and it’s of a different phase, an inter-chassis current can flow through the loop due to differing resistance in the multiple ground paths.

Having all the sound system gear on a common electrical phase minimizes the chance of having a leakage induced ground loop current.

In other words, less chance of current = less chance of hum.

Moving Right Along...
The original installation also included a remote power switch for the amp rack, which consisted of a key-switch on a panel at front of house that fed low-voltage AC from a doorbell transformer to the coil on a relay at the amp rack. Turn the switch, the relay closes and the amps have power.

The problem was that the electrical installer appropriated one of the shielded balanced lines running through the front of house to stage/amp rack conduit for the remote power switch.

I don’t care what kind of coupling was going on in that 80 feet of conduit because no matter how you look at it, what was happening in there was a sin.

Thus, fix number three: eliminating the remote power switch. We could have run an alternate line, but ultimately decided to skip the hassle and just do away with it.

Almost There...
When we got the system to the point where the console was re-patched and turned on, there was still a hum. 

What the heck?

Examining the console, I noticed some channels and sends marked “Bad, Don’t Use”. Perhaps it was just age, or maybe a surge during a lightning storm - the details weren’t really important, but the console had been in this condition for some time without repair.

And so, we implemented fix number four: replacing this faulty gear (combined with a lecture about surge protection).

We snagged the console from the church’s youth room and patched it in, and ahhhh - sweet silence! 

I thought we were home free.

One Final Twist...
The last day of the project, and nearing time for a sound check with the worship team, we patched in numerous microphones for vocals, piano, drums, guitar cab, as well as DI boxes for bass and keys. 

Opened the channels, pushed up the faders, and again there’s a hum. It wasn’t on just one channel, but almost every channel from the stage. Pull the faders down or mute the channels and the hum would go away, so the problem’s got to be before the console inputs. 

Phantom on or off made no difference. Dynamic mic, condenser mic, or DI also made no difference.

I hit the stage and started taking a closer look, and noticed the printing on the jacket of one of the mic cables.

This cable bore a respected name but had no business being there. And I had seen this before.

Whether the cable was the result of a manufacturing problem or possibly a counterfeit, it had no valid use anywhere in this sound system. It wasn’t even shielded twisted pair, but instead had dual shielded conductors that didn’t even appear to be marked for identifying + and - . And, the shields were tied together at both ends.

The first time I encountered this cable it caused a hum on any channel it was used. Now, looking across the stage, I saw upwards of two dozen of these demon-possessed cables from hell. This system never had a chance!

Finally, fix number five: A trip to the local pro audio store to procure suitable replacements, and before throwing the demon cables into the dumpster, I took the added precaution of cutting them into small pieces, lest anyone be tempted to salvage them.

Moral Of The Story...
For the first time in the history of this church, there was silence in the sanctuary with the sound system turned on.

And there wasn’t a single lifted ground or shield anywhere in it.

The church as so happy that they soon devoted funds to a new house console, as well as investing in proper surge protection.

I’d entered this quest reluctantly. It proved far more challenging that I had imagined.

And I’m really glad I pursued it all the way through to a successful outcome.

Solving hum and buzz problems is never easy, and it’s easily among the least glamorous facets of our work in sound reinforcement.

But the fact is that it can make all the difference in the world when it comes to the final result of any project.

Since his start 29 years ago on a Shure Vocalmaster system, James Cadwallader remains in love with live sound. Based in the western U.S., he’s held a wide range of professional audio positions, performing mixing, recording, and technician duties.

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More articles on PSW by James Cadwallader:
Coffee Beans & Microphone Techniques: The Desired Result Determines The Method
Feedback: A Big Necessity In Developing Quality Live Mixing Skills
How And Why Unity Mixing Can Make All The Difference In The World
Yes, Virginia, System Gain Structure Matters - Here’s Why
No Slave to Gear: Maximizing What You Get Out Of What You Have

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Tuesday, June 30, 2009

Church Sound Files: Eliminating Sound System & Operator Distractions That Take Away From Worship

As people who work with sound, we focus first on features and technology. On the other hand, pastors often consider aspects that can have a negative impact on a service, which ultimately detracts from worship...

When working with sound at church, we all know just how many things can go wrong.

The kicker is usually when they go wrong, which invariably seems to be at the worst possible moment.

The church I belong to, like most, doesn’t have a great sound system. We sure would like to have one, but like many, we’ve chosen to “make do” over the years.

One day, I asked the senior pastor what his primary goal would be if we could get a new system. His reply? “We need something that would cause no distractions.”

Hmmm…

Of course, I was expecting him to mention things like audio quality, ease of use, uniform volume levels at every seat, wireless features, and so on. So his answer surprised me at first.

But after thinking it over, I realized he was exactly right. As people who work with sound, we focus first on features and technology.

On the other hand, pastors often consider aspects that can have a negative impact on a service, which ultimately detracts from worship – and that’s why we’re all there in the first place! Everything else comes second.

I’ve always thought of the church sound operator as a referee at a sporting event.  Most of the time, when either does their job well, no one notices. That’s the way it should be - no distraction.

But when something goes wrong, everyone takes note – distraction.

My train of thought continued. Can a well-designed modern sound system, with simplified controls and intuitive applications, lead to fewer problems and therefore less distractions?

The majority of modern audio components perform far better than their predecessors, due to superior design and manufacture. Not to mention they’re newer and thus less susceptible to problems.

As to the issue of whether they’re “easier” to operate, I believe that’s a subjective opinion of each system operator.

But this did lead me to consider another potential source of distractions, and where they often originate when it comes to sound: the training (and lack thereof) of system operators.

Beyond training, how well are most churches equipped to schedule and manage volunteer (or even paid) system operators?

Now, let’s backtrack for a moment.

As noted, the church I belong to doesn’t have a “whiz-bang” sound system, but it does get the job done, and we work very hard to make sure it causes as few distractions as possible. This is because we invested in quality components, which were installed by a qualified A/V systems contractor.

Not only did we choose to go this direction with the system when it was new, but we also rely on this professional to handle any upgrades of components, to fix problems that come up, and to assist with “check-ups” on a regular basis. A little preventative maintenance goes a long way.

I understand the temptation to try to purchase new systems and products in the least expensive manner possible, and to “self-install” them. This is natural – we all, churches included, want the best bang for our buck.

But if there’s one absolute fact I’ve learned after working in audio for nearly 30 years, it’s this: one of the best ways to eliminate potential distractions is to have a system designed and installed by trained professionals.

Installation mistakes such as poor grounding, sloppy wiring and terminations, improper cable selection and a host of other little things, can all add up to one gigantic mess. And these types of mistakes tend to be fruitful and multiply!

Worse yet, I’ve walked into churches and have seen loudspeakers that are not designed to be suspended being hung by eyebolts screwed into the side of their particleboard cabinets. I just hope that these “accident waiting to happen” distractions don’t occur during a service.

Here’s a checklist for evaluating distraction potential:

1) Was your sound system designed by a reputable audio consultant who understands the needs of the church, the acoustical properties of the sanctuary, and the capabilities of those who operate the system?
2) Was your sound system installed by a certified individual employed by a reputable systems contracting firm?
3) Is the company that installed the system still in business, and involved in your additions and changes?
4) Has your system been installed in phases or added to over time?
5) Are system operators well trained and knowledgeable?
6) Does your sanctuary’s physical layout require a lot of audio equipment to be moved around and re-connected between services? 
7) Does your church struggle to find trained, motivated people to run the system?
8) Does your system produce random hums and buzzes, level changes, dropouts, crackles, distortions, pops, feedback or other noises that seem to go unexplained?
9) Do you own and consult instruction manuals and documentation on your equipment and system?
10) Is your system subject to regular maintenance inspections?

If you answered “yes” to checklist items 1, 2, 3, 5, 9 and 10 – and no to the rest – then your system is probably in good shape.

If not, it’s time to consider taking the proper steps in making sure your church is a distraction-free place to worship.

Chuck Wilson works with sound at his church, was a sound contractor for more than 20 years, and is now the director of the National Systems Contractors Association (NSCA).

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Monday, June 29, 2009

Simulating A Live Drum Solo In The Studio

Hear and see a step-by-step process of adding a studio recorded drum solo to a live recorded track

Suppose you just recorded a band in a club to create a live album.

A few days after the gig, the drummer asks, “Can I play a drum solo in your studio, and have you add it to the album? I want it to sound “live”, as if I played it at the gig.”

This happened to me. We recorded a drum solo in the studio, then edited it onto the beginning of one of the live-recorded songs. People listening to the final CD thought that the solo was part of the set.

I’ll offer some suggestions on how to simulate a live, in-the-venue drum solo after the fact.

The techniques described here also apply to overdubbing other musical parts in the studio to replace flawed live performances.

Here’s the basic procedure:
1. In the studio, try to duplicate the miking setup that you used live. Match the microphone models and placement.
2. Record the instrument in a dry, neutral studio if possible.
3. Then add some artificial reverb that sounds like the live venue’s reverb. Set the reverb parameters to re-create your memory of the venue’s reverb time, degree of warmth, and so on.
4. Enhance the tracks with EQ, gating and compression as needed.
5. Add crowd noise and applause.

I put some mp3 samples of the recorded drum solo in this article so you can click on them and hear the results. If your playback stutters, right-click the samples to download them first, then play them.

The Finished Mix
Let’s listen to the studio-recorded drum solo after it has been enhanced to sound “live”: [complete mix.mp3]

As you can hear, we added some crowd reaction and reverb to the dry studio tracks in order to simulate a live recording.

Here is a screen shot of the drum-solo tracks. From top to bottom, the tracks are
• Kick
• Snare
• Overhead left
• Overhead right
• Rack tom
• Floor tom
• Audience reaction, left and right mic signals
• A different audience reaction, left and right mic signals

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There’s also a reverb plug-in inserted into a stereo bus. The snare and toms have sends to that reverb bus.

Studio Mix Without Processing
Now let’s break down the mix so you can hear how it was put together. First, here’s a mix of the studio drum solo without any EQ or effects: [complete mix-no reverb-no audience.mp3]

The sound is not too exciting, is it? Let’s work on one track at a time.

Kick
This is the raw sound of the kick-drum track—with no EQ and no gating: [kick-no gate-no EQ.mp3]

You can hear some leakage into the kick-drum mic, and the kick lacks punch. I added some EQ and gating as shown below. Here’s the result: [kick-with gate and EQ.mp3]

The EQ and gate settings are shown here (at left).

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It’s common to cut around 400 Hz on a kick-drum signal to remove the “papery” sound and to tighten the beat. I also added some beater click at 4 kHz, and brought up the low end slightly.

The gate threshold was set to pass the kick drum but remove most of the leakage.

Snare
Let’s move on to the snare drum. It lacked clarity and sounded kind of puffy: [snare-no eq no compression.mp3]

The EQ shown at right really helped to define the snare sound. [snare-with eq and compression.mp3]

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Some cut around 600 Hz, a shelving boost above 4 kHz, and a little peaking EQ boost at 195Hz made the snare drum sound more “expensive”.

Of course, that EQ might not work with a different snare drum.

Rack Tom
As you can hear in the sample below, the rack tom and floor tom mics picked up a lot of leakage, and the toms lacked punch: [rack-no gate-no eq.mp3]

Again, some gating and EQ helped those problems: [rack-with gate and eq.mp3]

Shown at left are the gate and EQ settings for the rack tom.

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A cut around 600 Hz is fairly typical for toms. A low-end boost brought out the tone of the drum. I set the gate release time long enough to hear the tom-tom ringing.

With the overhead mics, I wanted to pick up mostly just the cymbals, so I rolled off everything below 500 Hz.

The Mix With Reverb & Effects
Now that the sound of the drums was improved, we needed some reverb to simulate the original venue.

I set up a reverb plug-in with 0.4 second reverb time, and inserted a reverb send in the snare and toms tracks. Here is the result: [complete mix-with reverb-no audience.mp3]

Reprise: The Finished Mix
Those drums are really starting to sound live, but one vital ingredient is missing: the audience reaction.

I had recorded some applause and yelling at the end of each song.

As shown at right, I copied and pasted some of that under the drum solo in the bottom four tracks.

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And this was the final result: [complete mix.mp3]

Adding some crowd reaction works amazingly well to simulate a live event.

I hope you enjoyed hearing and seeing how this “simulated live” recording came together.

AES and Syn Aud Con member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.
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Related PSW Recording Articles
Remastering Jazz Classics: The Dave Brubeck Quartet, Art Pepper, and Sonny Rollins
Deconstructing Hip-Hop To Hear How The Mix Comes Together, By Bruce Bartlett
Recording Microphone Techniques To Produce Warm, Spacious Stereo, By Bruce Bartlett
What Is Greatness? (In Recorded Music, That Is…), By Fletcher
How To Compare & Hear Differences In Microphones, By Ty Ford
Watch Out For The Big Lie: “We’ll Fix That During Mastering”, By Jackson B. Jackson

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Tuning A System At The Ultra Music Festival With EASERA SysTune (And “Dr. Bassenstein")

A look at the tuning process - for subwoofers in particular - for an annual festival featuring performances by some of the top DJs in the world (Also be sure to check out "An Overview Of EASERA SysTune" by Charlie Hughes)

The Ultra Music Festival takes place annually in the spring in Miami, concluding a week-long Winter Music Conference attended by electronic music artists, producers, and fans from around the world.

Ultra Music features performances by some of the world’s top DJs, with the festival site crammed full of PA systems from various vendors.

This year I returned at the invitation of UMF Audio Chief Terry MacNeil (“Dr. Bassenstein”) to perform alignment work on systems as a couple of the stages.

In particular, a lot of attention gets paid to the subwoofers - there’s a lot of content below 50 Hz, and the subs need to be as “right” as possible. Unfortunately, festival scheduling issues restricted my efforts to a fairly tight window..

Advance Work
The PA vendor for both the main stage and Bayfront stage was Beach Sound (www.beachsound.com). The main stage would be equipped with 32 d&b audiotechnik J8/J12, 16 d&b J-Sub subwoofers flown along the J8s, and 24 d&b B2 subs in three high CSA stacks, four stacks per side.

In addition, BASSMAXX supplied 16 beta test subwoofers currently given the model designation SP218 or the “Dub-ill 18.” These are double 18-inch direct radiating vented subs. The challenge would be integrating the centered BASSMAXX subs with the B2s flanking them.

Issues:
1. Physical separation between sources, setting us up for interference problems.
2. Different models of subwoofers, setting us up for potential phase (frequency specific delay) issues.
3. Subjective sound quality difference between the two models of subs.

BASSMAXX chief David Lee supplied some phase data for the new sub, and it appeared the phase response wrapped smoothly enough (for example, no abrupt variations in the operating band) that there would be a good chance of acceptable integration.

I contacted Neil Rosenstock, Beach Sound System Engineer, about “the plan” and we began to coordinate a rational approach to getting as much of the work done in advance as possible. The initial plan was to be able to use incremental delay taps for the center cluster. However, the stacking arrangement proved to be advantageous, allowing us to fill the center without beaming as much as if it had been an eight wide/two high system.

Neil came up with a CSA stacking plan that would steer the B2s away from the center a bit, supplying an ArrayCalc solution that did just this.

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As we shall see later, a bit less steering got it within acceptable limits out in the overlap areas.

EASERA SysTune
I decided to give SysTune a test drive for the event, so three weeks prior I downloaded the evaluation and worked my way through the tutorial.

The tutorial is very good, and anyone who understands measurement issues and has used a dual channel FFT analyzer before will be in a good starting place after completion. Particularly, I wanted to use SysTune’s ASIO multichannel capacity.

My multichannel measurement rig currently consists of:
PreSonus Firestudio Project 8 channel FireWire mixer
Small custom 2 space rack, with power strip
4 SIA RTA-420 microphones
Josephson C-535 microphone
4 Manfrotto collapsible microphone stands
7 microphone cables, of 50- and 100-foot lengths
Assortment of cables and turnarounds
WiFi router and IBM X41 tablet PC for remote access into measurement computer

This all rides in a Pelican 1650 case. Because I was arriving on tuning day and didn’t want take a chance on an airline losing it, I used FedEx to deliver it to Beach Sound.

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On With It
After some travel delays, I finally arrived at the festival site around 2 p.m. the day prior to the kick-off of the festival. Due to the schedule restrictions noted above, we weren’t able to make “big noise” until 5 p.m.

I got my multichannel rig set up in less than 10 minutes, but the amp rack for the center BASSMAXX subs had not yet arrived. Rosenstock and I walked the field briefly to see what the steered B2’s were doing, and sure enough there was somewhat of a hole carved out in the center.

In the meantime, I took a small measurement rig to the Bayfront Stage, with this rig consisting of the Dell Inspiron 4150 measurement notebook, an M-Audio MobilePre USB, and the Josephson microphone.

The PA consisted of flown d&b J8/J12 (no J-Subs), and 16 BASSMAXX X2C “Deuces” lined up across the stage.

The sub line was long enough to get some serious pattern control outside the edges, and in fact this is what we wanted in order to avoid spill to the extent possible in other areas.

With this many systems, and this much sub content, any control available is gladly exercised.

The quick alignment job consisted of sub alignment via phase trace to the flown PA, some quick EQ on the PA (the d&b systems seem to never need much, particularly outdoors), and some sweetening by ear on the subs.

A 6 dB low shelf boost on the Deuces suggested by Dr. Bassenstein was applied, and after a little experimentation using Lake Contour controller, we liked what we heard.

The lack of a tech day meant that forklifts and lulls were constantly working everywhere. At one point I had my back turned to the stage, trying to figure out what was going with this crazy transfer function that could not possibly be right, noise running at a fairly high level.

I turned around and a lull had pulled up next to my measurement microphone, completely contaminating the measurement. It turned out this would be the rule, and not the exception, the rest of the day.

Back To The Main Stage
The amp rack for the center BASSMAXX array eventually arrived and we began. The first set of measurements was mostly on axis with the house right portion.

Using the multi-channel capacity of SysTune, it was quite easy to quickly switch between measurement microphones. Additionally, the easy management of overlays helped me move quickly between tasks.

Since we were short on time, our efforts were concentrated on integrating the center BASSMAXX stack with the spread CSA B2 stacks residing under each side of the PA.

The Figure 1 screen shot shows a measurement overlay taken on site, and reloaded back into SysTune after the fact. The measurements were done with a 64 kHz FFT size, yielding 1.46 Hz frequency resolution.

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This resolution is OK for low frequency work but is too fine for HF work.

Also, the delay offset and measurement levels do not affect the loaded overlays. Savvy users will notice the “zero” delay time – this is because we’re looking at reloaded overlays, not live measurements.

The previous measurement is of only the steered-out B2s, taken from the center of the audience area. The next step is to add the BASSMAXX cluster in the center as seen in Figure 2.

Keep in mind here we have steered the outside B2s away from the center a bit to allow the BASSMAXX boxes to have some of their own space. We have not adjusted the gain on the center sub array at this point.

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Listening tests validated our original notion that if they sound different enough, we should avoid overlap if possible, and these subs definitely do not sound “the same” - whatever that means.

So in our center position, the B2s began to drop off just below 40 Hz as seen in the screen shot. The BASSMAXX subs remained quite flat to 30 Hz.

The phase angles are mostly matched, if not perfectly timed at the measurement position.

However, after listening we decided to treat the BASSMAXX array separately and Rosenstock inserted an 18 dB octave low pass filter at 47 Hz (after some experimentation), and this yielded the above result.

Note the phase response in the areas of interest, not varying more than 90 degrees between the two systems. We played with some delay times but this yielded no appreciable difference, so we left it “as is.”

Walking the field revealed a few but mostly insignificant nulls, certainly far fewer than a traditional left and right sub arrangement. This is the measured response of the summed systems, in the center of the audience area, as shown in Figure 3.

A little subsequent tweaking sweetened the very bottom end, of which a few of the DJs (and particularly The Prodigy) tested to the limits.

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Overall, everyone was satisfied with the arrangement.

DJ Tiesto’s Production Manager walked the field and commented on a few minor nulls but said “it is powerful enough.” Rosenstock and I walked around a bit and decided to pull the CSA stacks back in toward the center and it did help.

The Prodigy’s Front of House Engineer Jon Burto, had a few comments regarding the implementation and offered some suggestions. He had an interesting night mixing, between artists directly in front of the PA, an artist with a somewhat weak voice that evening, and the unenviable position of not hearing the same bass response his audience was hearing.

Final notes and observations:
• The mix platform was located on an SL-100 mobile stage. In fact, the bass response “up there” was significantly different than what the audience experienced.
• The RTA-420 microphone delivered equivalent performance to an Earthworks M30 for subwoofer work. The sensitivity is different but I detected no real difference in either magnitude or phase.
• The BASSMAXX array was quite powerful. We inspected it a few times during the show and experienced blurred vision and difficulty communicating. Yes, I had my -25 dB earplugs in.
• Behind the barricade, there was complete (and I mean COMPLETE) cancellation for a small distance between the stacks.
• While mixing sub models is generally not recommended, with some careful planning and overall awareness of the issues, if you have the tools you can make it work (usually…).
• Yes, it was “powerful enough.”

Doug Fowler is Director of Audio Engineering Services for Logic Systems Sound and Lighting in St. Louis.

(Also be sure to check out “An Overview Of EASERA SysTune” by Charlie Hughes)

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Loudspeaker Measurement: An Overview Of EASERA SysTune

An inside look at some of the more common functions and uses for this recently released program (Be sure to check out related article "Tuning A System At The Ultra Music Festival With EASERA SysTune And Dr. Bassenstein")

EASERA SysTune is a relatively new, patent-pending audio measurement program from the Ahnert Feistel Media Group (AFMG), designed for real-time analysis of acoustic signals and aimed at those doing system alignment, tuning and live sound applications.

SysTune is not a replacement or upgrade for the EASERA measurement program, but rather, it has its own unique features that allow it to perform measurement tasks not currently available in other measurement programs. More in-depth acoustical analysis of the measurements made with SysTune can be done with EASERA.

SysTune runs under Windows 2000, XP and Vista operating systems. It’s a multi-threaded application that is capable of taking advantage of multiple processors within a computer.

SysTune supports up to eight input channels simultaneously at up to 192 kHz sample rate, and it can operate in a dual-channel FFT mode, meaning that one input channel receives a reference signal (mixing console output for example) to which the other input channels are compared.

SysTune can also output its own signal (sweeps, pink noise or user selected files) and use this as the reference. Switching between each of the input channels is accomplished easily by clicking on its channel button. These are located just above the mini-meter for each channel.

The mini-meters are very convenient to tell at a glance which channels are receiving good signal level. If an overload (clip) of the A/D converter occurs the meter turns red. After the overload passes the outline of the mini-meter stays red until it is reset.

There is a multi-channel mode which allows two or more inputs to be averaged in real-time and the result displayed. This feature can be very useful to spatially average the sound system response using multiple microphones in different locations.

There is a very important process at work under the hood of SysTune. This is the Real-Time Deconvolution (RTD) engine. This newly developed algorithm allows for the calculation and display of impulse response (IR) data of up to 10 seconds in real-time and with fast on-screen display refresh rates.

To accomplish this, SysTune performs an FFT on the signals from the input channel and the reference channel. The transfer function (TF) of the input channel is then calculated in the frequency domain by complex division of these spectra. Finally, an inverse FFT is performed on the transfer function to yield the IR.

Using Windows Functions
Another useful feature in SysTune is one of the windowing options. While some standard window functions are available (Rectangular, Tukey 50 percent, Hann and Blackman) a new Time-Frequency-Constant (TFC) window function has been developed by AFMG. This is not a fixed length window, as are the others.

The time length of the TFC window changes as a function of frequency. The window time is inversely proportional to frequency so that multiplying the window time at a given frequency by that frequency yields a constant value.

As an example, if the window time length is set to 10 ms this is the length for 1 kHz. At 8 kHz the window is 1.25 ms, while at 125 Hz it is 80 ms (see Figure 3 later in this article). This allows reflections to be excluded from the measurement at higher frequencies.

At lower frequencies the window is longer so frequency resolution is increased, on a fixed bandwidth basis. This can also be thought of as having a constant percentage octave frequency resolution across the entire frequency spectrum. A three-dimensional representation of the TCF window is shown in Figure 1.

One of the most useful aspects of SysTune is the way in which the measurement data can be displayed for the user. There is an upper and lower display graph, either of which can be maximized to fill the entire display if desired.

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Any of the possible display selections can be shown in either graph area to suit the user’s preference. The ability to view measurement data in both the time and frequency domains simultaneously can be of great benefit.

Figure 2 shows captured measurements made during the alignment of an under balcony loudspeaker to one of the main clusters in a large church. The upper graph shows the IR’s while the lower graph shows the transfer functions for a microphone position in the under balcony area.

The green and blue curves are the main house right cluster and the initial measurement of the under balcony loudspeaker, respectively. The signal from the under balcony loudspeaker precedes that of the main cluster by approximately 65 ms at this mic position.

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The cyan curve shows the result of applying 70 ms of delay and some filtering to the under balcony loudspeaker. SysTune allowed the delay and EQ to be set relatively quickly while listening to the results to confirm a natural sonic character with the focus of origin at the main house cluster and not at the under balcony loudspeaker.

Handy Tips
It is also possible to display the spectrum or spectrogram of the input signal. The spectrogram can be particularly useful during a live performance to help pinpoint feedback frequencies. Most of the programs I’ave seen display the spectrogram with time on the horizontal axis and frequency on the vertical axis. SysTune does this as well.

However, it also allows for the axes orientation to be switched. This places frequency on the horizontal axis to coincide with display of most other frequency vs. magnitude graphs (Figure 3).

SysTune also includes some handy ways to help determine if the displayed measurement data can be trusted or if it should be considered suspect. It does this in two different ways; Coherence and IR Stability.

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The coherence function is similar to other programs such as (Rational Acoustics) Smaart and (Meyer Sound) SIM. This is a measure of the correlation between the input signal and the reference signal.

When the coherence value falls below the user-defined threshold, the Coherence Mask can be used to make the measurement data in this same frequency region somewhat transparent.

This can be seen in Figure 4 below 100 Hz and from 2-5 kHz where the coherence value is below 35 percent. Hand claps were used to contaminate the measurement in the higher frequency region.

IR Stability is also a correlation of the input and output of the system under test.

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However, it does not directly relate the input to the output as coherence does. Instead it is a differential function that quantifies the similarity of the last two measurements. It is also given as a percentage from 0 to 100 percent as is coherence.

Because IR Stability only looks at the most recent two FFT measurement blocks it can give a fast indication of changes in the data which occur very quickly.

Real-Time Calculations
Since SysTune calculates an IR in real-time, it can also calculate a couple of metrics in real-time which are derived from the IR; Reverb Time (RT) and Speech Transmission Index (STI).

STI is a measure of how well a system under test transmits human speech so that it can be intelligible to a listener. SysTune will display either RT or MTI (Modulation Transfer Index) and STI as measured in real-time. It will display a curve and give the values at one octave data point intervals from 125 Hz to 8 kHz.

A broadband value for STI is also given. It should be noted that while SysTune implements the calculation of STI per the IEC 60268-16 standard, it does not include correction factors for signal masking and noise levels.

This can be done, however, by saving a measurement in SysTune and opening it in EASERA for more detailed analysis.

The capability of performing IR measurements in real-time can be used in concert with the dual-channel FFT mode to measure the IR, RT and STI of a venue with a full audience present, but without the use of sweeps or noise test stimuli. This can even be done for a purely acoustical performance without the use of a sound reinforcement system!

An actor or soloist may be fitted with a microphone to capture their performance at a relatively close distance. This will serve as the reference signal in SysTune. Several measurement microphones may be placed in the room.

SysTune can then perform all of the measurements of which it is capable. One potential deficiency with this method is that the source for the reference signal (actor/soloist) may not sufficiently excite the system under test at all frequencies of interest. This will result in low coherence values at these frequencies.

SysTune is also capable of measuring, displaying and saving SPL and LEQ histograms. Weighting functions (A, B and C) can be applied to the SPL measurements. These can have a Slow (1 s), Fast (125 ms) or Impulse (35 ms) time constant applied during the measurement. The averaging time for LEQ can be set by the user.

This capability can be useful for doing noise studies over extended periods of time. Noise Criteria (NC) can also be calculated and displayed from a measurement of the ambient noise present.

We have briefly touched on some of the more common functions and uses for EASERA SysTune. There are many more features to this piece of software that can make it a valuable tool to the electro-acoustical practitioner. A free demo version of the program can be downloaded from www.easerasystune.com.

Charlie Hughes heads up Excelsior Audio Design & Services; a consultation, design and measurement services company based near Charlotte, NC. Charlie is a member of the AES, ASA, CEA and NSCA. He is an active member of several AES and CEA standards committees.

(Be sure to check out related article “Tuning A System At The Ultra Music Festival With EASERA SysTune And Dr. Bassenstein”)

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Friday, June 26, 2009

Barry’s Toolkit: The Shure 55 Microphone Has Deep Roots, But How Does It Hold Up Today?

The 55SH Series II is essentially the same at the 70-year old Unidyne except for modern internal acoustic components are used to bring it up to present day requirements. But how does it sound?

How do you review an icon? Often seen in photographs along with iconic people from dictators to heads of state to rock n’ roll stars, the Shure 55 is the most recognized microphone in the world.

First produced in 1939, the original Model 55 Unidyne microphone was the first single-element unidirectional dynamic microphone.

From its beginnings at that time being a cardioid pattern microphone, its smaller size made it a true classic that was affordable and accessible to all.

Then as well as today, its impressive ‘art nouveau’ satin chrome-plated die-cast case was closely equaled by its break though performance.

The Model 55 (unlike competing omnidirectional microphones at that time) was able to operate in close proximity to loudspeakers without creating feedback. It’s characteristic Shure presence peak made it excellent for vocal pickup in any situation from radio broadcast to live shows.

What most people know and think as the 55 was called the Model 55S and was a smaller version of the original Unidyne 55.

First produced in 1951, Shure now makes an updated version called the 55SH Series II, which is essentially the same as the 70-year old Unidyne except for modern internal acoustic components are used to bring it up to present day requirements.

Both the cardioid polar pattern and the same presence peak of the original 55 are incorporated into the 55SH along with an on/off switch, a new cartridge shock mount to reduce stand noise, and a redesigned swivel stand mount that permits tilting from 45 degrees forward to 80 degrees backward.

And Now The Super 55
Most of the 55SH’s features and touchstones of quality come with the Super 55 Deluxe Vocal Microphone. Exactly the same sized chrome-plated die-casting is used and differences begin with an enclosed supercardioid element behind a vibrant blue foam material.

A blue “Circle S” logo is on the Super instead of the black logo seen on the 55SH and then there is that shiny new Super 55 nameplate. The 55SH has a wider low frequency response while the Super 55 has more high frequency extension but retains the 55SH’s “voicing” for natural-sounding vocal/speech recording.

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But say goodbye to the old on/off switch on the Super—these days it does not have a place on stage and especially not in the studio. (Good riddance!)

The Super 55 is also heavier and 5 dB more sensitive than the 55SH model.

In The Studio
In the studio there are plenty of good applications for the Super 55. I used it without an EQ or compressor--only my RTZ Professional Audio 9762 Dual-Combo Mic Preamp (http://www.barryrudolph.com/mix/rtz.html), a super high-end Neve 1272 copy.

I recorded a Martin D28 acoustic guitar with the mic positioned above the twelfth fret. The voice-tailored frequency response worked out well to produce a bright sound with less emphasis on the guitar’s boomy bottom end.

I found this brighter sound unusual for a dynamic mic—I expected a more nasally version of what I would obtain with a small diaphragm condenser mic.

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Compared to a Granelli Audio Labs G5790 Right-Angle SM57 (which is a brand new Shure SM57 modified with a 90-degree elbow in the middle), (http://www.barryrudolph.com/newtoys/toys6/granellig5790.html) the Super 55 was clearer, less compressed and a little warmer in the low midrange.

Just to make sure, I confirmed this with another brand new, stock SM57.

I have to squawk about one thing. The mic’s XLR connector is positioned so close to the threaded hole in the base of the mic, that there cannot be any threaded washers usually found on stands and booms.

You’ll have to remove them so the mic cable’s XLR connector will plug in. I didn’t try them but I’m betting that Atlas quick disconnect mic hardware will not work with the Super 55—and that’s OK with me as I never trust them with heavier mics.

On electric guitars, I played through my ZT Amps LunchBox 2 guitar amp (http://www.barryrudolph.com/newtoys/toys6/ztampslunchbox.html) and the venerable SM57 won only by a nose with its overall thicker midrange sound.

As with most guitar amps, the SM57 has a way increasing the midrange “guts” of the sound.

On the same amp the Super 55 was brighter with very slighter more low end. It worked better when I moved it further off center and towards the edge of the amp’s speaker. Both the 55 and 57 were pushed right into the amp’s grill cloth.

On voiceovers, the Super 55 delivers a radio-ready vocal sound. I preferred having the voice talent aim right into the front of the mic, as the mic’s supercardioid pattern does not allow for any free ranging around the mic. If you talk into the side of the mic, you’re gone—not heard!

I got nearly no room tone with this mic even when recording in a medium live space. I liked the amount of proximity effect of this mic too: it’s not exaggerated allowing some forward/back head movement without a huge increase/decrease in bass.

The great proximity sound might explain the many pictures taken of Elvis Presley on stage leaning over and erotically caressing a 55 in front of his band.

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He holds the 55 so close to his mouth that it looks like he was about to take a bite of a forbidden fruit. When that close, I’m betting he could hear and liked the bass build up while singing in the lower registers.

In truth he probably was leaning over straining to hear himself over the anemic sound systems of those days.

Around drum kits, the Super 55 is physically bigger than most dynamics normally used on kits and will preclude using it in tight spots such as close to snare drums.

Boomy bass drums benefit from its bright attack sound and if you can get it close enough to a snare drum (to get enough proximity) you’ll get a sound that won’t require any EQ.

I found the Super 55 a refreshing new/old tool in the studio that just works so well in many applications. Its look is an immediate conversation starter that’s backed up by its impressive sonic qualities.

The Super 55 Deluxe Vocal Microphone is part of Shure’s Classic Collection along with the 55SH Series II, the 520DX Green Bullet, 545SD Classic Unidyne Instrument mic, and the 565SD Classic Unisphere Vocal mic.

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There is also the R115S Cartridge kit that includes the cartridge, mounting plate, and detailed instructions.

When properly installed in the Super 55 Microphone, the R115S will provide performance equal to that of the original cartridge.

The Super 55 sells for $249 MSRP. For more info go to http://www.shure.com/ProAudio/Products/WiredMicrophones/us_pro_super_55_content

Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Visit his website at www.barryrudolph.com

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More Reviews & Articles By Barry Rudolph On PSW:
Thumbs Up Or Down For The Marshall MXL V89 Studio Condenser Microphone?
Inside The Peluso P12 Tube Condenser Microphone
Barry’s DAW Toolkit: Review Of The Novation Nocturn With Automap 3 Pro
Barry’s Recording Tips: Figure Of Eight Royer For Electric Guitars
Review Of The X-Tempo Pok DAW Wireless Footswitch Controller
Barry’s Toolkit Of Handy DAW Products
Recording Gear Hits At The 2009 Winter NAMM Show
Working At Recording Success: Taking Elemental Steps Can Make All The Difference
Recording Tip: Successfully Dealing With A Dead Room

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Church Sound Basics: Proper Console Gain Structure, Maximizing Signal-To-Noise Ratio

Getting to the benefits of unity gain while eliminating system "hiss" and lessening the possibility of distortion

On the typical mixing console, each channel strip includes a knob at the top that behaves like a volume control.

Meanwhile, the fader at the bottom of the channel strip also controls volume.

Why are there two controls that appear to do the same thing?

You’ve probably heard sound systems that issue a fairly audible hiss in an otherwise quiet room, as well as distortion when someone speaks loudly or when a singer gets aggressive. Both of these problems are usually caused by improper gain structure at the console.

In typical applications, several different types of microphones (and direct inputs) are used.

For example, at a church a pastor wears a wireless lavalier mic, the pulpit has a condenser mic, the praise team has four dynamic vocal mics, and there is one acoustic guitar pickup with no preamp and one electronic keyboard.

All of these devices send a different signal level to the console. The guitar pickup and the dynamic mics send a relatively weak signal.

The keyboard sends a strong signal because it’s a powered device. And the wireless and the pulpit mics are somewhere between the two.

A Related Idea
All electronic devices have a “noise floor.” Whether it’s a $50 component or a $50,000 component, all produce a certain amount of noise.

Most manufacturers of audio equipment attempt to maximize audio signal while holding noise floor to a minimum. The difference in level is the “signal-to-noise ratio” seen on manufacturer data sheets. (Typically, a 90 dB ratio is considered to be “studio quality.”)

In order to maintain a correct signal-to-noise ratio for each channel, thereby eliminating hiss and lessening the possibility of distortion in the channel, try this:
1) Set the top control on the channel strip (usually called gain or trim) to the fully counter-clockwise (off) position;
2) Set the channel fader at the “0” position, indicated with “+” numbers above it and “=” numbers below it;
3) Set the master output fader at the “0” position. However, be aware that on some consoles, the master output fader has the “0” position at the top of its travel;
4) Have the person talking or performer address the mic the same way as during a performance or worship service. Slowly increase the gain clockwise until it is loud enough for the typical application of the system in the venue;
5) Shut off that channel with the channel fader, and move to the next (and the next and so on) until all channels to be used are optimized.

Depending upon the level each channel has at the input, you’ll notice that the gain controls are all over the place. Some are high, some are low and some are in between.

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But despite the fact that many different signal levels are now coming into the console, as soon as they all pass through their respective gain stages, they’re all at the same level.

Things that once were different are now all the same, hence the term “unity gain.”

Input To Output
A wonderful side benefit of doing this procedure is that now all channels and master faders are set to “0”. The bottom line is that the signal-to-noise ratio is now maintained all the way through the console, from input to output.

Further, the noise floor is low, the audio signal is high, and gain settings are likely lower than they were previously. This decreases the possibility of overdriving the input.

If your input gain settings tend to be near or past the 3 o’clock position on the rotary dial, it’s an indicator to turn up the system’s power amplifiers.

However, be careful not to overpower your loudspeakers, and lower the gain setting a bit.

If left at the higher position, a strong momentary vocal signal may push the input into distortion.

Finally, if power amplifiers are already turned all the way up, use equalization to make additional adjustments.

Just keep in mind that all of the pieces of equipment in the signal path have a relationship to each other. It’s important to maximize signal and headroom while minimizing noise and distortion.

Starting with the proper input gain is a vital first step. 

Jon Baumgartner is a veteran system designer for Sound Solutions in Eastern Iowa, a pro audio engineering/contracting division of West Music Company.

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Thursday, June 25, 2009

Don’t Bet Your Life: Get System AC Grounding Right

Most sound systems consist of at least two devices that operate on utility AC power. Although hum and other problems are often blamed on “improper grounding,” in most cases there is actually nothing improper at all. Here's why.

After teaching seminars for more than 15 years now, it still amazes me how many otherwise competent professionals don’t understand the importance of proper equipment safety grounding.

Even more shocking (pun intended), many routinely and casually disconnect safety grounds to solve noise problems!

Generally speaking, the purpose of grounding is to electrically interconnect conductive objects, such as equipment, in order to minimize voltage differences between them.

National Electric Code (NEC) requires that 120-volt AC power distribution in homes and other buildings must be a three-wire system.

Figure 1 shows how AC power is typically delivered from the utility company to the load at an outlet. For simplicity, only two of the three main utility connections are shown in the drawing.

One of these incoming utility wires, which is often un-insulated, is the grounded or “neutral” conductor.

Note that both neutral (white) and line (black) wires are part of the normal load current circuit shown by the arrows.

Code requires that the neutral (white) and safety ground (green) wires of each branch circuit be tied or “bonded” to each other and to an earth ground rod at the service entrance.

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Any AC line powered device with exposed conductive parts (that includes signal connectors) can become a shock or electrocution hazard if it develops certain internal defects. Insulation is used in power transformers, switches, motors and other internal parts to keep electricity where it belongs.

However, for various reasons, the insulation can fail - effectively connecting “live” power to exposed metal as shown in Figure 2. Such a defect is called a fault.

For example, if the motor in a washing machine overheated and its insulation failed, the full line voltage could energize the housing of the machine!

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Anyone who accidentally touched the machine and anything grounded, such as a water faucet, at the same time could be seriously shocked or electrocuted. 

Remember: current will always return to its source, whether the path is intentional or accidental. Electrons don’t care - they can’t read schematics! 

Trip The Breaker
To return this fault current directly to its source, many devices have a third wire connecting exposed metal to the safety ground pin of their plugs. The outlet safety ground is routed, either via the green wire or metallic conduit, to the neutral conductor at the main breaker panel.

This low-impedance connection to neutral causes a high fault current to flow, quickly tripping the circuit breaker that removes power from the circuit. To function properly, the safety ground must return to neutral. (Note that the EARTH connection had NOTHING to do with this process!)

NEVER, EVER use devices such as three- to two-prong AC plug adapters, a.k.a. “ground lifters,” to solve a noise problem! (Figure 3) Such an adapter is intended to provide a safety ground (see the fine print) in cases where three-prong plugs must be connected to two-prong receptacles.

If a proper safety ground isn’t available, always use a ground-fault circuit interrupter (GFCI). A GFCI works by sensing the difference in current between the line and neutral conductors.

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This difference represents current in the live conductor that is not returning in the neutral - the assumption is that the missing current is flowing through a person.

If the difference reaches about 5 mA (milliamps), an internal circuit breaker is tripped, removing power from the circuit. The GFCI shown in Figure 3 is unusual because it has a retractable ground pin that allows it to be used with a two-prong outlet.

Also consider two devices connected by a signal cable, each device having a three-prong AC plug. One device has a ground “lifter” on its AC plug and the other doesn’t.

If a fault occurs in the “lifted” device, the fault current flows through the signal cable to get to the grounded device. It’s very likely that the cable will melt and burn. Defeating safety grounding not only is both dangerous and illegal, it also makes you legally liable!

In a typical recent year in the U.S., consumer audio and video equipment electrocuted nine people and started 1,900 residential fires that caused 20 deaths, 110 civilian injuries, and over $30 million in property damage.

Current determines the severity of electric shock. At 1 mA or less, it’s simply an unpleasant tingle. But at about 10 mA, involuntary muscle contractions can result in a “death grip” or suffocation if the current flows through the chest.

Currents of 50 mA to 100 mA through the chest usually induce ventricular fibrillation that leads to death. The resistance of dry human skin is high enough to safely allow lightly touching a live 120-volt conductor, but normal skin moisture allows more current to flow as does increased contact area and pressure.

Lightning & Dirt
The earth itself is the return path for the current in a stroke of lightning. To protect people and equipment from lightning, we must make a connection to actual soil.

Overhead power lines are frequent targets of lightning. As a result, virtually all electric power distribution lines have one conductor connected to earth ground periodically along its length.

Before this was done, power lines effectively guided lightning inside buildings, starting fires and killing people.

The (NEC) code-required earth ground at the service entry panel serves to direct lightning to earth ground before it enters the building. For the same reason, the code requires telephone, CATV, and satellite TV cables to “arrest” lightning before it enters a building.

Because soil has resistance just like any other conductor, earth ground connections are not at zero volts with respect to each other or any other mystical or “absolute” reference point. Code allows the resistance of this earth connection to be as high as 25 Ω.

Since this is far too high to trip the circuit breaker under fault conditions, an earth ground should never be confused with a safety ground.

Safety ground must be connected to neutral at the main service entry panel. If more than one ground rod is used, Code requires that all must be bonded to the main utility power-grounding electrode. (Figure 4)

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Facts Of Life
Most sound (or video) systems consist of at least two devices, which operate on utility AC power. Although hum and other problems are often blamed on “improper grounding,” in most cases there is actually nothing improper at all.

Any properly installed, fully code-compliant AC power distribution system will develop small, entirely safe voltage differences between the safety grounds of all outlets.

In general, the lowest voltage differences (a few millivolts) will exist between physically close outlets on the same branch circuit and the highest (up to several volts) will exist between physically distant outlets on different branch circuits.

These normally insignificant voltages cause problems only when they occur at a vulnerable signal interface - more unfortunate than improper.

What’s all of this have to do with hum and buzz? People have a strong tendency to blame “dirty” AC power for audio-video system noises.

But in fact, AC power is a utility much like a public highway - used by huge trucks as well as sports cars.

Eliminating noise problems by “purifying” the AC power is much like re-paving the highways to fix a car’s rough ride. A much more cost-effective and practical approach is to eliminate the problem that allows the power line to enter the signal path in the first place.

This is analogous to replacing bad shock absorbers in a car to isolate it from rough roads. Finding and eliminating these coupling points will be topics of upcoming columns. 

Always make electrical safety your top priority!

Bill Whitlock has served as president of Jensen Transformers for 20 years and is recognized as one of the foremost technical writers in professional audio.

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Real World Gear: Column & Line Source Architectural Loudspeakers

The characteristics of line source columns – wide horizontal coverage, minimal vertical coverage above and below the enclosure and coherent sound in the vocal range – are all attractive for intelligible sound reinforcement in reverberant public spaces

Mention column loudspeakers to grumpy old sound men and Shure’s Vocal Master immediately springs to mind, as every ’70s band used it. Line sources have been around for a half-century.

For portable sound applications, audiences tend to be small. Installed systems are often employed in reverberant public spaces – houses of worship, auditoriums and passenger terminals – predominately for public address.

The characteristics of line source columns – wide horizontal coverage, minimal vertical coverage above and below the enclosure and coherent sound in the vocal range – are all attractive features for these kinds of venues.

Driver spacing determines the highest frequency at which a column of identical drivers acts as a line source, while the height of the column determines the lowest frequency with directivity. As with modular line arrays, a short system might efficiently throw the midrange, but leave a puddle of low-frequency mud behind the enclosure.

Inexpensive column speakers remain popular because they offer a compromise solution to installers who need efficiency in the vocal range combined with even coverage and a skinny profile that makes them acceptable on the walls of public assembly spaces.

Investigations into line source coupling behavior and pattern control tell us that loudspeaker cones exhibit coupling behavior up to a frequency whose wavelength is half the distance between adjacent acoustic centers.

Another old sound guy image is the column of JBL 2123 10-inch midrange drivers in Clair S4 cabinets. With their frames squared off to provide closer coupling, their acoustic centers could be placed 9.5 inches apart, providing coupling to 800 Hz.

In architectural columns 6.5-inch diameter cones, when tightly-spaced, will couple up to about 1,000 Hz. Four-inch cones couple to about 1,600 Hz and 2-inch cones to 3,300 Hz.

Above these frequencies top and bottom lobes appear in the polar response, however restricting the high frequency response of some of the cones can reduce lobing. This can be achieved with passive filters in the cabinet, or with active filters used in DSP-driven steerable columns.

A specialized version of the line source column is called “digitally steerable” with individual amplification, delay and equalization for each driver, allowing the column’s vertical coverage to be tilted down (or up) and focused for short or long throws, though its horizontal coverage remains fixed.

The advantage is that a column speaker can be placed flat on a wall, while its coverage can be tailored to a specific listening area. One advantage to digital steering is that the entire coverage pattern can oriented downwards. Mechanically tilting a passive array adjusts the farthest coverage, while leaving the coverage towards its sides near the original height.

More demanding applications, such as music, require both greater bandwidth and dynamic range, leading to two-way designs that incorporate separate high-frequency transducers in addition to the column of tightly spaced speaker cones.

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In modular line array enclosures, compression drivers are mounted onto “isophasic” manifolds that convert sound from their round opening into a tall, thin opening whose output is in phase along its entire height.

Columns lack both the budget or depth for this, so high-frequency drivers in 2-way column speakers are reproduced by a few closely coupled HF drivers, since their height only needs to be a few inches, or by the use of magnetic planar or “ribbon” drivers, which provide the required in-phase output in a tall, thin HF driver.

It’s common for line source columns to be combined as multiple cabinets to achieve better performance as taller systems for bigger rooms. Longer columns provide pattern control reaching to lower frequencies. A nine foot column can provide control to 125 Hz, so combining three 3-foot columns can increase low frequency performance.

As with miniature line arrays, some systems have companion LF columns which employ long-excursion small-format woofers to extend pattern control to lower frequencies.

Alternately, traditional subwoofers can supplement a column’s LF response.

Take our Gallery Tour of the latest column and line source loudspeakers on the market.

Mark Frink is Associate Editor of Live Sound International.

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Wednesday, June 24, 2009

Understanding Loudspeaker Specifications And What They Can Tell You

Knowing how and why loudspeakers do what they do is the critical step in wringing the most from any sound system

What’s wrong with this scenario? A clean stream of audio, processed through state-of-the-art digital processing devices and amplified by an advanced power amplifier, is delivered to the audience through a cone of paper pulp inside a wooden box.

The weak link in this chain is the box, more commonly known as a loudspeaker. Final sound quality of any system is wholly dependent on the ability of loudspeakers to reproduce the glorious audio presented to it by the upstream components.

In the past century, commercial loudspeakers have gone through slow, steady advancement in development. In contrast, the electronic portions of systems have undergone fundamental changes in physical attributes and methodology, near the speed of light.

However, just because the transducers (loudspeakers) have not kept pace with their electronic counterparts does not mean all is lost.

Quite the contrary, amazing performance cannot be realized with today’s best designs. Knowing how and why loudspeakers do what they do is the critical step in wringing the most from any sound system.

What’s Inside?
A loudspeaker’s task is to convert the electrical signal of the audio system into acoustic energy that humans perceive as sound. In most instances, the closer this output emulates its input, the better, because a known value of input will remain true on the output (i.e. high fidelity).

It’s also important to understand that sub-standard input sources will always detract from sound quality, regardless of the excellence of any given loudspeaker.

Professional loudspeakers usually contain multiple drivers (components) in a single enclosure. The most common design is referred as “two-way,” with two components teaming up to provide output.

Two-way designs usually consist of a 15-inch or 12-inch diameter cone woofer and a smaller-format (1- to 1.5-inch) compression driver coupled to a horn offering a defined coverage pattern.

For woofers, which reproduce the lower frequencies, the enclosure provides a plane of operation, a means of directly radiating output into the surrounding atmosphere.

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In contrast, compression drivers, which handle high frequencies, would be practically inaudible if directly sent into a coverage space. Thus the driver must be mated to a horn, to match the driver’s output with the surrounding atmosphere by unfolding the soundwave at a defined rate and dispersion pattern.

As a result, the efficiency of the horn/driver setup is raised dramatically, with only a few watts of input necessary to deliver room-filling levels.

Inside the loudspeaker enclosure, input signal is divided between the two components by a passive dividing network, commonly called a crossover, which directs the low frequencies to the woofer and the high frequencies to the compression driver.

Conventional loudspeakers work much like a car engine, with the cone action of the drivers akin to the movement of the pistons. The back-and-forth motion of each is the source of power needed to accomplish the work - turning a drive shaft or propagating sound waves.

Auto engines use gasoline as fuel, while loudspeakers use the electrical output of a power amplifier. In both cases, matching the fuel to the motor is essential to optimizing the performance.

Loudspeakers and power amplifiers have a give and take relationship, with the amplifier pushing against the loudspeaker’s natural state of equilibrium (equal pressure inside and outside of the cabinet) and the loudspeaker pushing back against the amplifier’s varying output.

The ability of an amplifier to control the driver’s excursion is termed its damping factor and is important to overall performance. Optimizing this interface is one of the reasons for the recent surge in powered loudspeaker offerings from many major manufacturers.

Key Specifications
Given the plethora of loudspeaker brands available, finding the best unit for a given application can be akin to the proverbial needle in the haystack.

Narrowing the field is made easier by the tenacity of reputable manufacturers’ adherence to industry standard specifications. These specifications include ratings for frequency response, sensitivity, power handling, and directivity.

Frequency response is a measure of how well a loudspeaker reproduces input signals across the 10 octaves of human hearing.

The engineering obstacles that prevent a single speaker component from achieving a perfect score on the frequency charts start with the physical makeup of sound waves and the limitations imposed by the need to make the loudspeaker enclosure practical in size and weight.

Low-frequency waves can be as long as 56 feet, while highs can be as short as one-half inch. Building a device that will faithfully reproduce these extreme sizes and everything in between is nearly impossible.

Further complicating the design is the need to keep the box as small as possible and aesthetically pleasing.

When a frequency response is stated, it will have associated with it a range-defining number such as “+ or – 3 dB (decibels).” Without the range limiter in the equation, the numbers are meaningless, because any loudspeaker can reproduce almost any frequency at some level (i.e.- 45 dB).

The standard range of +/-3 dB actually grants a relatively wide window of 6 dB, and should be considered the maximum usable variance. For full-range designs, a frequency response of 50 Hz to 15,000 Hz (15 kHz) can serve as a benchmark of standard performance.

Sensitivity, like frequency response, only carries meaning when parameters are clearly defined. The common specification for sensitivity includes the phrase “1w/1m” or one watt, one meter.

If a loudspeaker’s data sheet lists sensitivity at “87 dB@1w1m,” the sound pressure level on axis one meter from the front of the loudspeaker will be 87 dB when one watt of power is applied to the loudspeaker’s input terminals.

The application of two watts should yield a measurable output of 90 dB at the same position, given that a doubling of power results in a 3 dB increase in output.

Therefore, an input of 256 watts means an output of 111 dB, which will drop to 93 dB at a distance of 8 meters from the loudspeaker due to the Inverse Square Law - a doubling of distance results in a 6 dB drop in output at a given location because the doubling requires a four-fold increase in coverage area.

In comparison, a more sensitive speaker (92 dB@1w1m) with the same frequency response will deliver 98 dB at 8 meters distance. The 5 dB gain in output is free in that it cost no more input to get more output. As with all designs, however, there are tradeoffs and sensitivity is just one part of the puzzle.

Yin To Yang
Power handling is the yin to sensitivity’s yang, as an increase in one typically results in the decrease of the other.

For instance, an easy way to increase power handling is to make the components more robust, but the added mass needed to improve the drivers’ power handling causes those components to be less sensitive, with the common result of no increase in overall output level.

Power handling capability is important, though, to ensure the long-term viability of the loudspeaker system. Given the uncertainties that are the heart of live sound, it’s possible for a loudspeaker to receive an input spike that is magnitudes above the average operating level.

For those inevitable peaks, adequate functionality at extreme levels can be the difference between a system’s last gasp and continuing performance.

Problems arise when loudspeaker manufacturers publish inflated numbers regarding the ability of their products to deal with high input levels. The lowest number stated references the loudspeaker’s long-term power handling.

Depending on the brand, long-term may refer to a period of four, eight, or twenty-four hours of continuous operation with a defined input signal.

Program power is often rated at twice the long-term amount and represents the wattage of variable broadband music material the loudspeaker can handle for a shortened time.

By the way, an impressive – but useless - rating is the peak program (or impulse wattage), which represents the power the system can sustain for a brief moment.

Directivity refers to a loudspeaker’s capacity to control the location of its output. Low directivity is suitable for placing broad coverage within a short throw distance, while high directivity excels in situations where pattern control is of primary importance.

Big Waves
Due to the physical attributes of sound waves, directivity control is frequency dependent. It’s relatively simple to control the dispersion of small high frequencies, but quite difficult to control where a 10-foot-long wavelength (100 Hz) might go.

Generally, the larger the mouth opening of the horn, the lower the pattern control remains effective. For instance, a horn mouth of two feet corresponds to control down to about 500 Hz.

Therefore, a horn pattern rated at 60-degrees by 40-degrees will only display those degree coverages above the cutoff frequency dictated by the horn dimensions.

High frequencies, although easier to control, tend to cluster together in a tight manner, forcing horn designers to develop techniques such as constant directivity to help insure the even spread of high frequencies across the horn’s operating range.

Naturally, such control by force creates new anomalies that require further innovative techniques to solve.

While modern commercial loudspeakers are fundamentally close to their ancient ancestors, given current technology’s computational abilities and rapid innovations, the loudspeaker of the future may eventually break free of its current bonds/limitations.

But for now, the issues of frequency response, sensitivity, power handling, and directivity are the defining parameters for objective judgment. How these specifications actually translate to overall sound quality goes to the subjective – but that’s another story.

Kent Morris is noted for his church sound training abilities. He has more than 25 years of experience with A/V, has served as a front-of-house engineer for several noted performers and is a product development consultant for several leading audio manufacturers.

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