Friday, July 15, 2011
In Profile: Tom Danley, Exploring The Possibilities Of Audio Technology
“What I do is to try the best I can to solve the problem at hand; however that happens, I really don’t give that much thought." - Tom Danley
In his role as director of R&D for Danley Sound Labs (DSL), Tom Danley has found a home for his unique talents that offers him more freedom to indulge his passion for invention and problem solving than ever before.
But although the Gainesville, GA-based company is Danley’s professional home, he does most of his work from the northern Illinois suburb where he was born and raised.
When he was growing up, Danley says, the area was far less developed. “Not in the middle of nowhere, but getting there, and literally, on the other side of the tracks. So I spent a lot of time slogging around in the mud and water in the woods.”
Though developers have discovered the area in the interim, it still offers an ideal combination of working environments, he says. “If I have an idea and want to measure something I can go downstairs at 11 pm and see if my idea is right, or if I’m stumped and need to go walk in the woods, I can do that, too.”
While the environment he works from may be familiar, the work the 58-year-old inventor and innovator does with DSL – and has done throughout his career – is almost entirely concerned with the unfamiliar; an ongoing exploration of the fringes of possibility in the development and application of audio technology that goes well beyond sound reinforcement.
Working with NASA hardware contractor Intersonics from 1979 to 1996, Danley designed and built hardware for sounding rockets, the KC-135 zero gravity “vomit comet” and the space shuttle program.
There he was awarded 17 patents for a variety of inventions, among them the Servodrive subwoofer and a variety of acoustic and electromagnetic levitation devices - the first of which was a sound source 100 times more effective than what Intersonic had used previously.
He’s also designed sonic boom simulators for BBN, an outdoor Flow Modulator-based unit for GTRI/NASA - affectionately dubbed “the speaker from hell” - and acted as principal scientific investigator on various research contracts, including one that explored the use of low frequencies to trigger avalanches.
Danley with a Space Shuttle payload at Intersonics.
Danley freely admits that from time to time his fondness for low frequencies has resulted in his scaring the wits out of those around him and himself. One of the early users of the TEF-10, he once demonstrated acoustic levitation in the Charlton Heston-narrated documentary The Mystery of the Sphinx.
Later, in the mid-90s, he was asked back to Egypt to measure the acoustics of the Great Pyramid with a TEF-12+, an experience he covered in detail for a Live Sound International cover feature in July 2000.
“In the pyramid, I scared the heck out of everyone. After the first TEF sweep the producer asked me to turn it up - I also went down lower - and it literally felt like it was shaking the place.” It was just air moving back and forth, he adds, but he moved closer to the chamber entrance anyway. “I tried not to act like it scared me, but, yeah, it did.
“I’d say that’s a bit of a pattern,” he continues with a laugh. “When I was a kid my uncle worked for the telephone company and he gave me a hand cranked generator out of an old telephone. With very little turning it gave you a nasty shock.”
Fascinated by this powerful force he couldn’t see, he decided to share his experience with others. “In third grade I had my entire class, including the teacher, hold hands and let them experience it. The shock wasn’t strong enough to hurt anyone, but it woke you up.”
It was an entirely different invisible force that prompted his lifelong fascination with audio, however - a fascination sparked by his interest in his grandfather’s mono hi-fi and later fueled by a clandestine exploration of the organ loft at the church he attended, where he first encountered “sound you could feel.”
“I used to help my grandfather clean up after services, but sometimes there were more interesting things to do, like exploring.”
Noticing the ladder to the organ loft, Danley asked if he could go up and have a look, but that first exploration was purely recognizance. “Three weeks later I saw the organist come in – she was a little lady that walked real slow, so I was able to sneak into the loft without her seeing me. She started playing and it was like, ‘okay, this is interesting’, but when she hit the pedals I didn’t know whether to run or what. It was just so powerful. That really stuck.”
In the short term, it prompted Danley to learn bass guitar and take regular trips to nearby railway tracks to feel the rumble of passing trains. Long term, it led to the development of the Servodrive subwoofer and ultimately to the creation of products like DSL’s SM Series loudspeakers, Synergy Horn, Tapped Horn and the JH-90 Jericho Horn.
“My father and his brother were inventors,” he continues. “My dad had a workshop, and he got an old motorcycle out of a flooded garage, got it going again and we rode that around the yard. When I was 11, I learned how to arc weld, and I really loved taking things apart.” It was an unusual childhood, he adds, but one that resulted in an insatiable desire to understand how things work and how to make them work more effectively.
Ironically, as passionate as Danley was about that pursuit, he disliked school in general and math in particular. “I took every shop class there was, but I had an attendance problem and math was my weakest thing. I knew how many marbles you could fit in a box car,” he says, laughing, referencing a long ago test question, “and thought what more math do I need than that?”
Doing The Homework
Soon, however, Danley realized that if he wanted to work in audio he needed substantially more knowledge of math and physics. “It was ‘well, you can design a horn if you can do this math’ and that was like, ‘oh no, this is the worst thing’.”
College was never on the menu, he says. “I didn’t think my family could afford it and my grades were so bad it didn’t seem realistic.”
Instead, after high school, he began working at Steamer Sound, a local loudspeaker company started by classmate TC Furlong. Steamer Sound didn’t last, but Danley continued to learn the math and science he’d previously neglected on his own. Still, he found it difficult to sustain a career as a loudspeaker designer financially and often settled for positions as an electronics technician.
Although live sound was only a small part of his career, he points to a gig at a local club, partially owned by Furlong, as a turning point.
Danley with his Servodrive creation – dig the white lab coat!
“It was the John Burns Band in 1974. I brought my reel to reel and plugged it into the mixer and recorded the full range signal.” Later that night, unable to sleep, he hooked the deck up and listened. “I was floored. It wasn’t album perfect, but it was close.”
What struck him was the huge difference between what he’d heard live and what he’d recorded. The realization that the only possible variable was the loudspeakers would substantially inform his future career and inventions.
Another breakthrough came during a stint at Northbrook, IL-based Data Specialties in 1976 after a co-worker gave him a Commodore Vic-20 computer. Seeing its potential, he immediately set about writing a program that cut down the time it took to do the math that was so integral to his work. “It took weeks to write, but you only had to enter the variables and press return to get an answer. Literally, it brought tears to my eyes.”
Eventually he walked across the driveway and applied at Intersonics, landing a job where his unconventional approach to invention was embraced wholeheartedly by company president Roy Whymark.
In addition to his work for NASA, while there Danley invented the Servodrive subwoofer, and with the blessing of Whymark, an audiophile himself, finally had the opportunity to start up a loudspeaker company as a division of Intersonics.
Hearing Servodrive in action on the massive Michael Jackson and U2 tours of the time remains a highlight. “What we did with high intensity acoustics, it was interesting and fun, but sound is what I love.”
Although profitable, Servodrive was tolerated more than encouraged, Danley says, recalling a near collision between visiting NASA representatives and metal rockers Manowar. “They were in their rock and roll clothes and the NASA guys literally plastered their backs against the wall so Manowar could walk by. It was classic, but I knew it wasn’t going to go over well.”
The 1986 Challenger shuttle disaster was the beginning of the end, he says, and in the years that followed, budget cuts and the growing influence of competitor Jet Propulsion Laboratory over decisions concerning what research would be done, and by whom, greatly diminished Intersonics’ role.
And as the NASA contracts dried up, Danley was forced to absorb some personal shocks as well - both the death of his father and the disintegration of his marriage, which left him solely responsible for the care of his two young daughters and threw into stark relief the importance of finding a home for his next inventions that would allow him to fully capitalize on their potential.
He transitioned to serve as chief designer at Sound Physics Labs (SPL) in Chicago, and when that company faltered, decided to license inventions such as the Synergy Horn and Tapped Horn to a pro audio manufacturer.
Danley Sound Labs president Mike Hedden with Tom.
Soon, however, another option presented itself via Mike Hedden – owner of SPL’s biggest distributor. When he floated the idea of forming DSL, Danley jumped at the opportunity.
“When Mike called and said, ‘how’s the speaker business sound?’ I thought it sounded pretty good. But most importantly he’s genuine; what he says, he does, and that’s a wonderful thing.”
While Hedden had never manufactured loudspeakers before, he brings a wealth of business acumen to his role as president of DSL, which allows Danley to focus exclusively on R&D. Since 2005, DSL has grown exponentially and now counts the likes of IMAX, Cirque du Soleil and major educational, sporting and worship facilities worldwide among its clients.
As for the genesis of DSL’s revolutionary loudspeaker technology: “Well, I like things people say you can’t do. I had a friend who said ‘there are speakers that sound good and there are speakers that go loud, but there aren’t any that do both and that sounded like a target to me. So we’ll say, ‘we need a speaker that does this kind of a job’ and then it’s ‘O.K., how do you do that?’”
In the case of the Jericho Horn, the answer was complicated. “The combiner in the Jericho was the hardest thing I’ve ever done. Literally, it was four months of saying ‘wow, I wish I hadn’t said I thought I knew how to do this’, but it ended up being very similar to what I’d envisioned.”
In recent applications the box has proven itself a powerful and compact alternative to line array.
“With the interference pattern that a line array produces, if the wind blows, there’s a pronounced comb filtering effect,” he explains. “If you have a speaker like the Jericho that radiates essentially as one single source, the wind has almost no effect. It’s a giant difference in subjective sound quality.”
Although his number one priority is to help keep DSL growing, expanding what’s possible in sound reinforcement has also led Danley to ponder problems well outside the realm of live sound.
“I have ideas that cover a lot of subjects,” he says, among them the possibilities of alternative energy and propulsion technologies utilizing sound. “If you look at the motion and pressure involved with sound, the force that wind applies is actually in the same neighborhood, so I have a couple of ideas about how to capture energy from moving air.”
Clearly, he’s thinking farther outside the box than ever, but pondering problems that would give most people a screaming headache is just another day in paradise for Danley, an extension of what has driven him to invent and innovate his entire life,
“One of the things Roy Whymark said to me was, ‘what’s good about you is that you don’t know what you can’t do.’ What I do is to try the best I can to solve the problem at hand; however that happens, I really don’t give that much thought. You notice something in the mechanical world and you find an electronic analogy to that, or there’s something in the air motion that there’s a mechanical equivalent to.
“But that’s how a lot of good inventions come about - you take a principle from one area and apply it to another it’s never been used in.”
Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer. Find out more about Danley Sound Labs here.
This is one of those “thinking out loud” posts for me.
It’s been a topic of conversation at our church for the past several months, particularly as budgets have been cut (again).
Actually, I’ve been thinking about it on and off since I moved to Minneapolis in 2007; that’s when I first encountered paid musicians and techs in the church.
Prior to that, all the musicians and techs I’ve worked with had been volunteers. Honestly, it’s one of those topics that has left me still working on a position.
Before I dive into my still-forming conclusions, let’s consider both sides of the debate.
The Case for Paying Musicians (I’ll get to techs in a minute)
Those that support paying musicians in church are likely to point out that the church has a long history of supporting the arts and should continue. Paying the band—that is, artists who make their living playing or teaching music—is a continuation of that tradition.
Supporters would also agree that the musical worship time of the service is important, and paying for professional musicians will deliver better results with less rehearsal time.
It’s also important to note that a band that’s paid is under a little tighter control of the worship leader or music director. They tend to show up closer to the call time (or they don’t work as often), and it’s easier set and enforce expectations.
As a general rule, the quality of musicianship tends to be higher with a paid band, and that even makes it a lot more fun for the FOH engineer (who may be paid or volunteer). I’m sure there are other reasons to pay musicians, and the ones I just mentioned are all good ones. Honestly, I don’t really disagree with any of them.
On the other hand, where does it stop? Surely the front of house or monitor position requires just as much skill and training as does a band member, so should we pay those positions? Over the history of Coast Hills, that’s been the tradition.
However, based on my budget for the year, that tradition is coming to an end. When I was in Minneapolis, I always found it odd that the musicians were paid but the front of house engineer was not.
But what about the guy who helps out doing graphic design for the church? If he’s a freelancer, he’s an artist making his living doing design; if we want to support the arts, do we pay him as well? What about the teacher who leads a kid’s Sunday School class? Do we pay her also? Or the carpenter who helps out building sets for the Christmas production?
I’m not trying to be overly dramatic, but at some level, you can make the case for paying almost everyone who volunteers their time at a church.
Might we get friendlier ushers if we paid them? Maybe, but at what point does paying people to “serve” turn church into an attraction to be visited rather than a body that serves?
Part of the equation that further muddies the water is the distinction between bringing in outside musicians and contractors and people from the body.
In our case, we have both serving every weekend.
Actually, we often have three classes of musicians; outside contractors who don’t call Coast Hills their home; professional musicians that are part of our body, and are paid; and volunteers who may be project managers or firemen but also play a mean instrument.
This strange mix has never been a source of consternation (at least that I’ve seen), which is a testament to our team’s leadership. However, it is interesting. What is more interesting is seeing what happens when budgets get cut and people who used to be paid can’t be paid any longer. Some keep on playing, others sit out.
The Case For Volunteers
The other side of this coin is to use all volunteers—that’s been my experience for most of my church life. In fact, I’ve been a volunteer TD (tech director) far longer than I’ve been a paid one. I made my living working in the professional production world and gave my time at church.
The way I saw it, I’m not good with kids, I don’t like to greet people and I can’t sing. But I am a good tech, so that’s where I served. I’m sure I’ve given thousands of hours to the churches I’ve been a part of over the years, and loved (almost) every minute of it.
We talk a lot about putting ministry back in the hands of the people at Coast. When I use that phrase, I mean trying to find people who are gifted in various areas (in my case, tech) and empowering them to serve. For me, it’s not about saving the church money (though that is a nice side benefit) it’s about giving people the opportunity to serve.
It’s like giving of our finances; when we give, we benefit more than the church does. It’s about obedience and becoming more like Jesus (who is our example for being a servant). There is no better way to grow in our walk with Christ than to serve, and a big part of me thinks that when we bring in paid people from the outside, we deprive those in our midst of growing in their walk with Christ.
Sometimes however, a church is really just looking to save money and cuts out all the paid musicians and tech people. That’s one way to trim a budget, but if the expectations are not adjusted, few will be happy with the results. I
t takes a long time to get really good at what we do, and while it’s easy to make the decision in the board room, it’s a lot tougher in the field.
So where do I land on all this? I don’t know yet. I see the case for paying musicians, especially the ones in our midst. I love those guys and I know how hard it is to make a living as a musician; I want to support them.
I also know that the positions we’re talking about (musicians and front of house engineers) take highly specialized skill sets.
You can’t just cut a budget and say, “The band and front of house have to be volunteers from now on.”
I figure it takes a solid year to train someone to mix front of house at the level we expect at our church (unless the volunteer is committed to doing it every week, then it goes faster). And truthfully, few are cut out for it.
I’ve been talking with quite a few TDs about this recently, and it seems that there are a few positions in the church that need to be paid. Quite a few churches are starting to look at the front of house engineer as a necessary paid position because of it’s visibility, importance and the high degree of specialized training required.
Whether a paid front of house engineer is a requirement at your church will depend on the level of production excellence required. While I’m busy training new audio volunteers, I’m a bit nervous that when we start getting them on the board during a live service, their performance won’t be up to the level expected by leadership. And that puts us in an interesting position.
At the same time, some of my greatest experiences in life happened when I was volunteering at church. I want to open as many doors for that to happen as possible.
On the other hand (I told you this was a complex issue), everyone—and I do mean everyone, senior pastor and board included—have to be willing to accept the compromises that come with non-professional talent on stage and behind the board. It’s not going to be perfect. Notes will be missed, mics will be muted when they should be on. We all have to be willing to live with that.
What say you? Are musicians and techs paid at your church? If so (or not) how do you feel about that?
Mike Sessler is the technical director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
When you first started using a compressor, you were happy to just have a basic understanding of what the threshold and ratio do, right?
At some point, though, you need to learn how to deal with the attack setting.
I’ll give you a starting point today.
The attack setting simply tells the compressor how quickly it should compress the signal once it crosses the threshold.
Note: Don’t confuse attack with ratio. Ratio tells the compressor how much to compress once it crosses the threshold.
When you’re dating a compressor, it’s important to take things slow.
My buddy Ian has been known to say that fast attack times kill music. For the most part, he’s right.
Music is all about rhythm and emotion. If your attack times are too short, you’re chopping off the transients on your tracks… losing that punch you’re trying to get.
While I’ll admit it’s fun to use a fast attack and watch the gain reduction meter bounce all over the place, it’s best to try to get into the habit of using slow attack times. The compression will be more subtle, and you’ll be less likely to over-compress the track.
Rather than starting at 5 or 10 ms…dial that puppy back to something like 50-75 ms. Let those transients through! Then let the compression shape the tone of the instrument AFTER the transient.
Missing The “Punch”
People talk all the time about wanting their mixes to “punch.”
So they talk about using analog summing, or buying fancy plug-ins.
I’m not against either of those, but before you go there, try increasing your attack times on your compressors.
You may just re-discover that punch you were looking for.
When To Go Fast
Of course, there are times to use fast attack times, too.
Whenever the transients of a given signal are too loud or need to be “tamed” a bit, you should try using a faster attack time.
The best way to do this is to set your ratio and threshold where you want them, then slowly dial back the attack knob. As the attack time shortens, the underlying audio will slowly become more and more “dull.”
That’s not necessarily a bad thing, and it might be just the thing you’re looking for for a particular track.
Here are a few examples of when it makes sense to use a fast attack time:
—Too much “thump” in the bass. Perhaps your bass player digs in really hard when he plays bass, so all his notes have a really loud transient, but it’s hard to hear the sustain.
Or perhaps you can’t get a good overall bass level in your mix without the transients overtaking the mix. Use a faster attack time to clamp down on these transients before they get out of control.
—Snare drum cuts through the mix too much. Sometimes you just can’t get the snare drum to sit in the mix. You turn it up, it cuts through too much. you turn it down, you can’t hear it.
A faster attack will let you turn it up without it chopping your ears off at the beginning of each hit.
—Lead vocals. Sometimes a slow attack on vocals can make it sound a bit odd.
If you’ve ever listened to a talk radio show where every phrase the announcer says has too much “punch” at the beginning? Yeah, that’s probably due to over-compression and a slow attack time.
How do you set YOUR attack times? Leave a comment below.
Design Principles For Distributed Sound Systems Serving Business Spaces
Enhancing the atmosphere with well-designed distributed systems. In Part 1 presented here, Rick Kamlet of JBL discusses key ceiling loudspeaker design issues.
Before you start designing a business sound system, identify what the customer needs and expects:
Fidelity Expectations. Does the client want a good, basic system; something a little better than average; or maximum bandwidth and maximum fidelity?
Sound-Level Requirements. Will the system be used strictly for background music, or does it really need to rock?
Usage. What kind of music will be played through it? For example, if it’s urban funk, then you’re going to have to think about the bass quality and SPL capability.
Form. Do they want in-ceiling or on-wall loudspeakers?
Coverage Requirements. How even do they want the coverage to be? Is it OK for it to perceptibly vary in volume within the space, or do they want it even throughout? Are there areas that don’t need to be covered at all, or where they might want lower SPL, such as the cash register area in a store?
Low-Frequency Coverage. How even does low-frequency coverage need to be? If you’re using subwoofers, is it okay to use a small number, meaning the sound will be loudest close to the sub(s) and softer elsewhere, or do the subs need to cover evenly?
Paging. Do they need paging? If so, how important is paging intelligibility?
Zones. How many zones need separate volume controls, different source controls or paging assignment?
Benchmarking. Do they want this system in order to keep up with a key competitor? If so, this can give you a benchmark.
Cost. Does the system they want fit into their cost requirements? If not, which functions can be adjusted to meet the budget, or is a budget reassessment advisable?
Collect this information and confirm your understanding with your customer BEFORE you start designing. You’ll have a much clearer idea of how to proceed when you know what your customer needs.
As we start thinking about the design, we need to then translate our client’s requirements into specific goals for coverage, sound levels and bandwidth. Once these requirements are identified, we can start thinking about loudspeaker and component selection, speaker layout patterns and speaker density.
We need to be able to correctly commission the system after it is installed, ensuring proper settings and optimal performance. Let’s start here with loudspeakers, and in future installments, I’ll look at other issues, such as loudspeaker layout, as well as SPL and equalization.
Have you ever heard, or followed, rules of thumb like “space ceiling loudspeakers as far apart as the ceiling is high” or “as far apart as the distance from the listener’s ear to the ceiling?”
While they are simple to follow, they aren’t very practical. The conditions upon which these rules were supposedly based — the needs of commodity-grade systems — just aren’t what’s required today. No hard-and-fast rule is going to apply in all cases. You need to find out what the customer needs before you start making plans around loudspeakers.
The main objectives in deciding about the placement pattern and density of loudspeakers in a distributed system are covering the area effectively, providing sound that is audible and intelligible over the entire area, and making sure the system is capable of sustaining whatever sound pressure level the application requires.
A Common Misunderstanding
The following misunderstanding about the coverage angle specification of loudspeakers can easily result in system design mistakes.
First, let me clarify that the term “coverage angle” is the angle at which the sound level is 6 dB down from the on-axis sound level. It is very common to see a polar coverage spec of the “coverage angle” on a spec sheet and assume that the speaker will actually cover this angle when installed. It seems like it would be the case, doesn’t it?
But the truth is that loudspeakers actually cover less area than their spec sheets would imply.
Polar vs. Listening-Plane Coverage
There are two different types of coverage measurements that get confused for one another. It is standard in the loudspeaker industry to state the coverage in a polar pattern — in a sphere that is 1 meter from the loudspeaker in all directions. The angle where the sound level is down 6 dB from the on-axis level is called the edge of the polar coverage pattern. This is what appears on spec sheets.
It’s a legitimate specification, but it does not represent what the coverage will be over a flat listening plane, as in any room, because it doesn’t take into account the difference in distances that people are from the speaker.
For loudspeakers projecting from a ceiling onto a flat listening plane, the sound has to travel farther off-axis (to the sides) than it travels on-axis (directly below the speaker) resulting in a much greater drop-off of sound level off-axis.
The result is that the actual coverage angle (at -6 dB) on the listening plane is more narrow than the polar spec. Some ceiling loudspeaker manufacturers use their polar measurement to claim extraordinarily wide coverage. Do not use this specification to lay out coverage patterns of ceiling loudspeakers!
To Illustrate, imagine a loudspeaker with a 180-degree polar spec. If you were to incorrectly interpret this as 180-degree coverage on the listening plane, then one loudspeaker would be all you would ever need for any application. Imagine a single loudspeaker trying to cover an entire department store or restaurant.
In fact, you will see that unless a loudspeaker can send more sound to the sides than it does directly on-axis, it never covers more than 120 degrees.
The system designer needs to work with the actual coverage over a flat listening plane because that is the plane in which we live, listening at a height of 3 to 6 feet above the floor, depending on how tall we are and whether we’re standing or seated.
This is called the listening-plane coverage specification of the loudspeaker. The listening-plane spec represents the reality of the speaker’s coverage for the listeners. Laws of physics dictate that the listening-plane coverage is always more narrow than the polar coverage pattern.
Figure 1 shows a speaker that has a 140-degree polar coverage (i.e., its 6 dB down points). We can see that it would be a mistake to assume that this speaker can cover 140 degrees over the listening plane.
In fact, the level at the edges of a 140-degree pattern is actually more than 15 dB down compared to on-axis — not 6 dB down.
It’s interesting to note that the same proportions hold true for any ceiling height: No matter how high the ceiling is, the off-axis distance is farther away by the same proportion. So for the loudspeaker in this example, whether the ceiling height is 8 feet or 20 feet, the listener who is at the edge of the 140-degree pattern, who you might have thought would be at the 6 dB down point is really at the 15 dB down point!
The actual listening-plane coverage depends on the polar plot of each loudspeaker. On average, the coverage of the listening plane from a speaker with a 140-degree polar coverage is usually between 90 degrees and 110 degrees (see Figure 2).
How do you convert polar coverage to listening-plane coverage as you design sound systems? There are two ways.
One is to use a computer program that does the conversion for you. If you have a copy of EASE, place the speaker in the ceiling, set the listening plane height to the typical application height and see how much area it covers (at the 6 dB down point).
Usually that’s enough, but you can also do a little bit of math and figure out what the real listening plane coverage angle of the speaker is. JBL Pro created a simple program, “Distributed System Design,” to do this same conversion for its loudspeakers. I think there are some programs available that have a more generic database of loudspeakers.
The second way to compute the listening-plane coverage is to start with the exact polar plot of the speaker and use a conversion table. (Real polar plots directly from test equipment are more accurate than an artists’ redrawing.)
Polar plots are usually normalized to the on-axis value, which is usually labeled “0 dB.” For every angle off-axis, there is a “difference-figure” between this normalized on-axis value and the volume at that angle.
To convert to listening-plane coverage, add the dB Correction Factor figure from Table 1 for that angle off-axis to the figure from the polar plot. If you’re doing this correctly, the coverage pattern is getting more narrow than the original polar plot.
By using the actual polar plot of the speaker and applying these correction factors from the chart, the angle that results in a figure of -6 dB is the angle of coverage for the loudspeaker. This angle is the real 6 dB-down angle for that loudspeaker when it is projected onto the listening plane. Remember that this coverage angle is valid regardless of the ceiling height.
Example 1. If we look at the polar plot of our hypothetical speaker with 140-degree polar coverage, we see that at 70 degrees off-axis (140 degrees total for both sides) the level is down 6 dB compared to the on-axis level.
By looking at the polar-to-listening-plane conversion chart, we need to add -9.3 dB to this -6 dB figure to find the actual level on the listening plane at this off-axis angle.
We find that the level of this 140-degree loudspeaker (as specified by the polar coverage) is actually -15.3 dB, not -6 dB, down at 70 degrees off-axis.
Therefore, listeners located at this off-axis angle will hear sound that is more than 15 dB down from the level they hear when they pass directly underneath the loudspeaker. This is a very large difference.
To find the actual 6 dB down point of the speaker for the listening plane, take the actual polar plot of the speaker and at every increment of 5 degrees off-axis, apply the correction factors from the polar-to-listening-plane conversion chart.
The 6 dB-down angle is that angle at which the new figure reads -6 dB (polar dB down plus the additional dB down from the correction factor). While the final resulting angle depends on the actual polar plot of the loudspeaker, it can generally be said that most loudspeakers with a nominal polar coverage of 140 degrees can be expected to reach -6 dB between 45 degrees and 55 degrees off-axis, resulting in an actual listening-plane coverage between 90 degrees and 110 degrees.
Example 2. Let’s look at a loudspeaker that has a 180-degree polar coverage. Let’s further assume that it is a mythical “perfect” loudspeaker where the volume doesn’t go down at all at any angle. Its polar pattern when placed in a ceiling is a perfect half-circle.
To find the real 6 dB-down point, we apply the correction factors and find that at 60 degrees off axis (120 degrees coverage), the sound is 6 dB down (again, polar dB down plus the additional dB down from the correction factor).
Therefore, the real listening-plane coverage of a perfect 180-degree loudspeaker is only 120 degrees. Now, given the fact that speakers with a “spec” of 180-degree polar coverage can actually be down as much as 6 dB at the 180-degree point and still have a spec of 180 degrees, the fact is that virtually all loudspeakers have a real coverage of less than 120 degrees.
A couple of interesting points follow from this discussion. First, let me point out that subwoofers resemble Example 2. They have omnidirectional coverage of 180 degrees (when in the ceiling). So they, too, have listening-plane coverage of only 120 degrees.
Second, while it is not within the scope of this article, you can see that this principle applies to all loudspeakers projected anywhere. The fact that the listening plane in any venue is hardly ever a perfect sphere around the loudspeaker means that you will almost always be projecting speakers onto a somewhat listening plane; and the coverage is almost always going to be more narrow than what the polar spec would indicate.
A student once mentioned to me that he had done some church designs taking a protractor and scribing the coverage pattern onto the floorplan drawing. When the system was installed, they had more holes than expected. After a class on coverage, he realized that it is a matter of projecting a loudspeaker onto a listening plane so that the actual coverage is going to be narrower than what he plotted.
Let’s talk about broadband vs. single-frequency coverage specifications.
You may have done all the conversion work above, and it might apply only to one frequency!
If the loudspeaker does not have similar coverage at all frequencies, then it covers wider or narrower angles at different frequencies.
Some ceiling loudspeakers specify their coverage only at a particular frequency, often at 2 kHz. The reason 2 kHz if often used is that 2 kHz is the most important octave for speech intelligibility. However, other octaves a still very important for intelligibility and all octaves are important for music.
Many types of loudspeakers do not provide even coverage throughout the audio spectrum. Every spot within the listening area ends up with a different frequency response and different sound level (see Figure 3).
In addition to this making it impossible to set a coverage angle, it also defeats attempts to equalize because whatever spot you choose to set your EQ, it’s different everywhere else.
But there is a good solution: Choose loudspeakers that cover similar angles throughout a broad frequency range. This is one advantage of having loudspeakers with small diameters (so they don’t start beaming until a very high frequency), or having multiway loudspeakers with goo crossover networks where higher frequencies are reproduced by small drivers (tweeters or compression drivers for high-power loudspeakers).
Horns on the high-frequency drivers further help in providing even coverage at all frequencies. See Figure 4 for a well controlled loudspeaker.
Let’s take one more look at the old rule of thumb that says to space the loudspeakers as far apart as the distance from the listener to the loudspeaker.
First, this rule was based on an incorrect assumption that all loudspeakers covers 90 degrees. Even for loudspeakers where this coverage is true for some frequencies, it may not be true for all frequencies.
In today’s high-quality sound systems, you need to consider real coverage, not polar or incorrectly assumed coverage.
Next time I’ll focus on SPL and equalization of distributed systems.
Rick Kamlet is senior market director, commercial sound, JBL Professional. Published with permission by JBL.
A detailed discussion on which types of microphones are best suited for choir mic'ing, how many mics should be used, and how they should be positioned
Audix has released a new “How to Mic a Choir” instructional series of videos. (View video series below.)
The series offers six chapters, featuring Dean K from Audix and guest Travis Cibolski, engineer and technical director for Sunset Presbyterian Church in Portland, OR, providing a detailed discussion on which types of microphones are best suited for choir mic’ing, how many mics should be used, and how they should be positioned.
Additional subjects covered include microphone basics, polar patterns, the 3 to 1 rule, and the complexities involved with mic’ing a large choir in a contemporary setting where music is an integral part of the service.
Audix VP of sales and marketing Cliff Castle explains, “Today’s contemporary houses of worship have integrated elements of audio and video into their services to the degree where new challenges must be met. In addition to the typical audio challenges facing technical directors, there are now other issues to contend with, for example, choir set up requirements may change from week to week and there will likely be music added to the mix.
“Also, the choir microphones may need to have very low profile due to the integration of video screens into the service,” Castle continues. “As a result, choosing the correct choir microphones are critical. Audix began developing products for overhead choir applications 10 years ago.
“With the introduction of the Micros and the MicroBoom system, we have simplified the process. We find that it is extremely helpful to use video to demonstrate how to go about miking a choir every step of the way. These videos are beneficial to all viewers, regardless of their level of audio expertise.”
Turbosound Flex Array Propels Southard Audio Into Larger Venues
Southard plans is to add 16 more of the TFA600-HW wide throw boxes and 12 more TSW218 subs in the future
Located in Mt. Crawford in west central Virginia, Southard Audio is a regional sound company that during its 30-plus years in business has made the move from local events to large arena-sized shows.
Two major initiatives have powered this expansion: a long-term cooperative partnership with Richmond-based Soundworks, and a mutual reliance on Turbosound Flex Array loudspeakers.
“Our relationship with Turbosound goes back more than 20 years,” notes founder and president Mike Southard. “When it was time to expand into a line array system, it was easy to see the benefits of the Flex Array. It’s a great-sounding PA, very loud for its footprint, with easy and straightforward rigging – a perfect complement to our existing inventory.
That system was put to the test at a recent 11,000 seat show featuring Snoop Dogg at the Coliseum of West Virginia University. Advance work and on site operations for this event were handled by Southard Audio partner Jason Misterka.
“This was our first arena event for the school, and it was the largest concert audience the school had seen at this venue,” recounts Misterka. “As you can well imagine, an arena event with a national hip-hop act requires a substantial amount of horsepower. We ran the prediction software and were pretty amazed at the numbers, and during the show we had all the level we needed - quite a feat for a mid-sized line array.”
The main front of house system for the show was two arrays comprised of 13 Flex Array boxes each. Five TFA600-H boxes topped each array to handle the long throw duties, complemented by the wider dispersion of eight TFA600-HW speakers beneath.
Subwoofers were Turbosound TSW721, a single-driver, 21-inch horn-loaded design. A total of 24 subs were set up as a delay arc, with six stacks of four subs spread across the front of the stage.
“The delay arc sub array is a key element. You lose a little of the overall energy, which is why we used so many enclosures,” Southard explains. “But it gives you a much more even distribution of bass information throughout the venue and virtually eliminates the “power alley.” This way audience members on the extreme edges of coverage still hear a balanced, musical sound.”
Turbosound also played a big role on stage, with a mix of ground-stacked Aspect 880 loudspeakers providing sidefills, augmented by eight more TSW721s for bass.
“It was amazing, really,” says Southard. “When I look at how we use these speakers on our other jobs, the sidefill system was basically a PA system that would be sufficient for FOH for a 1500-seat room under normal circumstances.”
“We had to satisfy the touring engineers in terms of volume, coverage and bass, and the WVU Coliseum was a real challenge,” Southard recalls. “It’s a very lively room, and the very defined coverage of the Flex Array helped keep energy off the reflective surfaces, which was a big advantage. At the end of a long day, I’m happy to say that everyone went home happy.”
The company’s current inventory includes 16 Flex Array boxes (half TFA600H, half TFA600-HW), 22 Aspect TA-880s, 24 TSW721 subs, and 8 TSW218 subs. Southard’s current plan is to add 16 more of the TFA600-HW wide throw boxes and 12 more TSW218 subs in the future.s.
“Working in partnership with Soundworks in Richmond is a key part of our success,” concludes Southard. “Since we’re about two hours apart, it makes more sense to cooperate rather than compete. We combine our inventories as needed and make sure our amp racks are wired compatibly, which greatly extends both our capabilities.
“Steve Payne and I consult with each other on major purchases and try to have mutually compatible inventory. In fact, Soundworks owned the Flex Array system that I auditioned before buying my own. This way, both companies can do larger events than we could do separately. It’s a great arrangement.”
Church Sound: Stage Monitoring & Keeping Those Performers Smiling
An overview of approaches and techniques that help with improving and optimizing monitors in simple settings
A stage monitoring system is, simply, a complete and independent second sound system for the performers rather than the audience, made up of monitor loudspeakers (also called monitors for short or wedges due to their shape), power amplification and signal processing.
A monitor system can also have its own mixer/console, or receive a feed (or feeds) from the main system console. Note that adjustments made to the main system mix do not affect the monitor system mix, and vice versa.
People who sing and play instruments derive much enjoyment from both listening to and participating in music. Music sounds best when it is clear and balanced, when you can hear instruments and voices at appropriate levels. And singers and musicians play their best when experiencing these circumstances.
There are several keys to performer satisfaction with monitors and monitor mixes.
Let’s start with positioning the performers. On more than one occasion I’ve been asked to come to a church because “we can’t hear the monitors”.
Upon arrival, the monitor loudspeakers themselves are actually tuned pretty well, and I can stand on the platform or riser and hear crisp, clear, and well balanced sound.
This is when I’ve learned to ask where the people who “can’t hear” are physically positioned on the riser, and the answer, predictably, is that they’re almost always somewhere out of the primary monitor coverage field - too far to either side, too far forward, or too far back.
Getting performers to stand in the coverage field is usually a training issue - if necessary, the sound operator should show them where to stand, or, consider repositioning the monitor(s). The vast majority of times, it comes down to using what is already there (repositioning) rather than the need to add more equipment.
Other times I’ll come to a facility and experience monitor sound that is feeding back or poorly tuned - the equalizer (EQ) is not adjusted properly. EQ can help eliminate feedback, get rid of annoying, lingering overtones, and provide a way to “shape” the sound so it is pleasing to the ear.
What I find is that monitors tuned by novices are frequently bass heavy and “thick” in the mid frequencies, when instead, it should be rich and full, crisp and distinct. How to tune using an EQ is a complete topic in and of itself, and has been covered in a previous article.
Assuming monitors are well tuned, the next consideration is relative balance in relation to the main system. Aside from the problem of sound bleeding from the stage into the house, monitors that are too loud can also cause considerable distress to the performers.
Surprisingly, though, often when performers tell me they “can’t hear the monitors” it is because the monitors are too loud, rather than too soft. Being too loud robs all of the ambiance - a sense of space and the sound within that space.
This causes a lot of discomfort, and really, what the performers should be saying is not “we can’t hear” but rather, “we’re not happy.”
Ever notice how cool it is to sing in a parking garage, a gymnasium, or a canyon? We enjoy the sound of our own voices returning to us from a distance, and this phenomenon also occurs, albeit at a much smaller scale, with stage performances.
But if the monitors are too loud, the sound is too dry, in your face, and quite unmusical.
If you’re faced with this situation as a church sound operation, try this at rehearsal: tell the performers that you’re going to shut the monitors off, so at the beginning of the song, they’re only going to hear the sound of the main system.
Then let them know you’re going to slowly bring up the master monitor volume, and to raise their hands when the mix is clearly but gently lending enunciation to the sound they’re hearing from the room.
Assuming the voices and instruments are good quality, in tune and well presented, this is a nirvana situation for most performers. Sound is big and full in the room yet crisply defined by the monitors.
On smaller platforms/risers this method is particularly successful. Once the basic master level is set, tweak and adjust individual levels or do some physical repositioning until all the players and singers can hear everything in balance. The result is very musical and your players and signers will have a lot more fun.
When it comes to multiple monitor mixes, well, I may well get a lot of flack for saying it, but multiple mixes are frequently not necessary, particularly in smaller to mid-sized stage situations.
Players can position themselves so they can all hear each other adequately, and usually, the vocals are most prominent in the monitors while the instruments are far less prominent.
Now, on wide stages/platforms (especially those that are not very deep), multiple monitor mixes might be a good idea. The performers at stage right have a hard time hearing the performers at stage left, and vice versa. Multiple mixes allow them to request sonic information from the other side of the stage in their own mix.
Multiple mixes require separate signal pathways, from the aux on the console/mixer, to the equalizer, to the power amp, to the monitors. As such, it requires more investment in equipment.
Multiple mixes can also be necessary because many performers simply differ on what they prefer the monitor mixes to be. This preference is partially what gives rise to a number of earworn monitors (either wired or wireless) that have a remote control station for each performer to individually adjust the mix to their own liking.
Communication between the performers and sound operator is also vital.
Performers must understand that the sound operator is not on stage with them, so if they want monitor sound changes and improvements, they need to tell the operator, and further, to be as specific as possible. Some praise when the monitor sound is good never hurts, either.
Operators, in turn, need to understand that requests and complaints aren’t usually personal, just a desire for improvement and the expression of frustration at a subpar situation. Address each issue with patient attention, and it usually works out.
And don’t be shy about asking performers if they’re happy and comfortable with the monitor sound. This can both head off problems before they start and result in further improvements.
Remember, some performers may be holding their tongues due to less-than-professional operator responses in the past.
Kramer Electronics Enters International Licensing Agreement With InfoComm For Educational Content
New agreement expands the original agreement to include all markets, in all countries
Kramer Electronics has entered into an agreement with InfoComm International to offer InfoComm licensed courses throughout the world.
Up until now, Kramer was the only manufacturer offering InfoComm licensed material in the U.S only. This agreement expands the original agreement to include all markets, in all countries.
Kramer is the only manufacturer that offers InfoComm Licensed Content to their partners around the world, and the company is also offering more classes that carry CTS Renewal Units (RUs) than any other manufacturer in the industry.
“We are extremely happy to now be able to professionally deliver InfoComm licensed content around the world,” states Dave Bright, president of Kramer Electronics US. “We are committed to providing this valuable content to our customers and potential customers wherever there is a demand for it around the globe, and we are specifically intent on helping as many AV professionals around the world as possible to become InfoComm CTS holders.”
Over the past two years, Kramer has developed a niche by successfully offering the CTS Prep Course from InfoComm regionally in the United States. Since the inception of the CTS Exam Prep class, Kramer has provided CTS Exam Prep training to more individuals preparing for the new CTS Exam, than any other entity in the industry.
Students who have attended the Kramer led CTS Exam Prep class have a very high pass rate on the exam itself. As a result, many more people are taking and passing the CTS exam and the amount of CTS holders in the United States is growing rapidly.
Recently, Kramer Asia Pacific held its first CTS Exam Prep course under the new agreement which was conducted by David Penrose.
The 3-day course was held in the offices of Electronics and Engineering (E&E) Singapore, and six participants from E&E spent the three days going through the same content given by Kramer US. These participants will be taking the CTS exam in the near future. Kramer hopes to be offering more classes, in more countries, in the very near future.
“A strong AV industry is dependent on a well-trained workforce,” said Randal A. Lemke, Ph.D., executive director and CEO of InfoComm International. “Kramer has worked with InfoComm in the United States to train and help industry members to attain the CTS certification. As we begin this joint effort to take the same training around the world, it is important to say how appreciative we are of Kramer’s efforts as a leading company in training our industry and being a strong supporter of the CTS and InfoComm.”
As always, the Kramer non-licensed course offering is continuing to expand including new courses on sales and technology. For example, Kramer has recently added courseware on Digital Signals in AV (Plug and Pray), Understanding the Analog Sunset and Twisted Pair Cabling Best Practices.
“We are committed to the InfoComm curriculum and to offering our own educational programs that support the CTS renewal program,” states Clint Hoffman, Vice President of Marketing for Kramer Electronics U.S. “We are also committed to making the educational experience as accessible as possible anywhere we can around the world. We often bring our educational offerings, especially the CTS Exam Prep class, directly to our customer’s location or to an area nearby that is an easy commute for the students.”
Kramer Electronics has printed a new brochure that covers the educational and training offerings contained in their Kramer Academy. The newly printed Kramer Academy brochure can be obtained from Kramer by downloading or ordering it directly from the Kramer website.
Despite the preponderance of exceptional drum samples and loops on the market, for certain genres of music (notably country and rock) there is no substitute for a great session drummer playing on a well-recorded and mixed drum kit.
One thing that samples and loops can’t provide is the great rhythmic instincts an accomplished live player draws upon when responding to a specific song.
However, getting a great player (while certainly a significant element) is not the entire story.
The appropriate treatment of the drums in a mix with EQ and compression can make the difference between a lifeless, vague sound and an exciting, textured and genuinely rhythmic drum track.
Even though the drummer plays the entire kit as a single instrument, the miking of individual drums and cymbals can make for a very complicated mix scenario.
The reason I reference country and rock music specifically has to do with the fact that in these genres the sounds of the individual drums and cymbals are not only singled out by individual microphones placed on each of them but also their sounds are exaggerated to create an even more dramatic effect.
Consider, for example, the tom fills in Phil Collins’ “In The Air Tonight.” By contrast, jazz drums are often treated as a more cohesive, unified sound and it’s not unusual to use a simple pair of overhead mics to capture the sound of the entire jazz drum kit.
In this article, I’m going to go drum by drum providing EQ and compression settings that will, hopefully, provide you with a jumping off point to getting great drum sounds in your mix.
Because of its all-in-one mixing board channel approach, I’ll be using Metric Halo’s Channel Strip plug-in with its EQ, compression and noise-gate to illustrate my comments about various EQ and compression settings.
As the heartbeat of the contemporary drum kit, the kick drum sound we’ve grown accustomed to hearing is both boomy and round on the bottom and has a nice, bright click in the high mid range. It’s the balancing act between EQ and compression that gives the kick drum its ability to stand out in a mix.
Kick Drum Settings: Click to enlarge.
Beginning with EQ, the best way to accentuate the lows and highs is to remove some low-mids. I’m a big believer in cutting as opposed to boosting EQ to achieve a desired effect. As a result, I tend to pull somewhere between 2 to 4db at between 350hz-450hz.
Then, after removing some of this low-mid mud from the sound, I can enhance the clicking sound of the beater hitting the head of the kick drum by boosting around 2db in the 2k-3k range. I’m providing approximate dB and frequency range settings because depending on the kick drum, mic placement and, of course the drummer, all of these settings will vary. Use these general ranges as a jumping off point and then trust your ears.
As far as compression settings go, the trick is to preserve the transient attack of the kick drum with a fast but not too fast attack time (9ms in this instance) and then a quick release (11ms) so the compressor is ready to respond to the next kick drum hit. The ratio I use is a relatively mild 2.5:1 and I adjust the threshold until I hear the kick sound I’m searching for.
Finally, in order to give the kick drum sound some separation from the rest of the kit, I use a noise gate and adjust the threshold to allow the kick sound to come through while essentially muting the majority of the other drum/cymbal sounds. Also, while setting the attack to the Channel Strip’s fastest “auto” setting, I allow for a long (400ms) release.
This particular miking trick is one that can be used to bring great low-end presence to the kick drum.
By way of explanation, a short stand holding essentially the woofer of a speaker is placed in front of the kick drum and picks up predominantly the low frequencies.
When blended with the kick drum mic, the sub-kick generates great power in the lowest part of the frequency.
In order to accentuate the most important elements of the sub kick’s sound, I tend to use a low pass filter approach to my EQ that removes all frequencies above 500hz and drops off even more dramatically below 100hz.
Sub Kick Settings: Click to enlarge
This is to make sure that only the essential parts of the sub kick’s sound come through.
The sub kick should be felt more than it is heard. In terms of compression, a ratio of approximately 5:1, a relatively slow attack (120ms) and medium fast release (57ms) allow the sub kick’s tone to stay present and full underneath the sound of the kick drum’s regular miked sound.
Then, I’ll use a noise gate with a fast attack (20ms) and slower release (200ms) to keep out any other kit sounds that might otherwise bleed into the sub kick sound.
Along with the kick drum, the snare drum is essential for driving a rhythm track. Poor EQ and compression techniques can leave it sounding thin, dull and generally uninspired.
In order to accentuate the best parts of the snare sound with EQ, I’ll boost the low end of the snare by 2-3dB at around 80hz, cut 2-3dB between 350-450hz and then boost again, if necessary, for more high-end brightness, by 1-2dB at 5k. These three EQ points are a great place to start to sculpt an interesting snare sound.
Snare Settings: Click to enlarge
Compression on a snare is a real balancing act where too much will take away the energy of the performance and too little will make it practically impossible to find an appropriate level for the snare in the mix.
I use a ratio of 2.5:1 with a very quick attack (2ms) and release (11ms). If you’re finding that you’re losing the snap of the snare, slow your compressor’s attack a little but remember that slowing the attack too much will take the compressor too long to grab onto the sound and will leave the snare much less manageable in the mix.
Adjust the threshold settings until things sound right to your ear. This basically allows you to decide how much overall compression you’ll be applying. Don’t overdo it or the drum will lose its energy but don’t go too lightly or the snare won’t stand up in the mix. Gating the snare is a trial and error process as well.
Depending on whether the snare approach in the song is aggressive or soft will have a lot to do with your threshold settings. Like on the kick drum, I use the very fast “auto” attack and a slower release on the gate in an attempt to keep out the ambient sounds of the cymbals, toms and kick.
While obviously a cymbal, the hi-hat is often used more as a rhythmic element than a tone color like some of the other cymbals in a drum kit.
Making sure it has its own sonic space and speaks clearly without being too loud and distracting is what EQ and compression are about in this instance.
For EQ, I’ll again use a shelving approach at around 200hz that will effectively clear out low-end information that is non-essential to the hi-hat sound.
If I’m interested in bringing in a bit more high-end shimmer and sizzle, I’ll boost between 1-3dB between 6k and 8k again using my ears to tell me what’s working.
Hi-Hat Settings: Click to enlarge
In general, I tend to stay away from compression on the hi-hat as it tends to find is own dynamic range without too much additional help.
Low (Floor) Tom
A well-mixed set of toms can make all the difference between drum fills that are exciting and those that go by without catching the listener’s ear.
Starting with the low tom, I tend to look for the places in the frequency range that bring out both the boom and the snap (similar to the way I approach the kick).
In order to accentuate the low quality of this drum, I’ve found that a dramatic cut (12dB) at around 500hz allows the drum to speak clearly. Also, to add the high-end snap, a semi-aggressive boost of between 4-6dB at around 3k will do the trick. Compression also adds a lot to this equation.
Floor Tom Settings: Click to enlarge
A ratio of around 4.5:1, a slower attack of 120ms and medium slow release of around 90ms will help the sound remain full and resonant. For the threshold, I simply adjust until the tom rings properly.
Gating is another major factor for toms as the large diaphragm mics placed on these drums tend to pick up a lot of the extraneous sounds from the rest of the kit. I set the gate with the quickest “auto” attack and a slow 400ms release and then adjust the threshold until I’m hearing only the low tom come through when it’s hit. For the “tweak heads” among us there’s a slightly more accurate and labor-intensive way to do this.
By going into the actual sound files in your DAW and deleting all but the tom hits themselves, you can create a perfectly gated tom track.
High (Rack) Tom
Like the low tom, the high tom has it’s own frequencies that should be cut/accentuated to bring out the sweetest parts of the sound.
For EQ, I’ll do another big cut of around 10dB at 600hz and I’ll make a similarly big boost of around 7dB at approximately 2k.
For compression, I use a slightly more aggressive 6:1 ratio slower attack (100ms) and a quick release (25ms).
As with the low tom, I’ll gate the high tom using the identical gate attack (fastest “auto”) and release (400ms).
Rack Tom Settings: Click to enlarge
The key to the threshold is to adjust it until only the high tom punches through keeping the channel essentially muted for the rest of the time.
A final note on the toms, as all tom sizes, tunings and even drummers are different, you’ll need to play with these settings until you find the sweet spots.
Overheads / Room Mics
Given that we’ve made a real effort to isolate and enhance each of the individual drums in the kit, overhead mics serve the dual purpose of capturing the cymbals and integrating the blended sound of the kit back into the sound of the drums. I pay attention to three specific EQ points in order to give the overhead mics a clean, balanced tone.
First I’ll use a high pass filter (shelving EQ) at the very low frequency of 40hz to clean up any unnecessary sub-sonic rumbling. Then I’ll pull around 5dB at between 100 and 200hz to prevent any low-mid buildup.
Overhead Mic Settings: Click to enlarge
Finally, if necessary, I’ll enhance the overall brightness of the cymbals/kit with a small 1-2dB boost at around 5k. For compression, I’ll set the ratio at about 3:1, the attack at around 110ms and the release at a slightly quicker 70ms.
The threshold should be adjusted to make sure that the overhead/room sound blends with the overall kit mix. Finally, adjust the volume of the overhead mics in the mix until you pick up just enough of the room to put some air and depth back into the kit.
Limiting the Sub Mix
A final trick to add punch to the overall drum kit is to send all of the individual tracks to a stereo sub mix and place a limiter like the Waves L1 on that stereo auxiliary track.
Sub Mix Settings: Click to enlarge
By adjusting the threshold until the attenuation is between 5-7dB, you’ll find that the kit has a really satisfying overall punch and presence.
While I’ve been painfully specific about EQ, compression and gate settings, it’s important to remember that every mix situation is different. Use all of these settings as a jumping off point and then use your ears to tweak the sounds until you’re happy. Good luck!
In the music business for over 20 years, Cliff Goldmacher is an engineer, producer and owner of recording studios in Nashville and New York City. A multi-instrumentalist, Cliff has recorded, played on and produced thousands of demos for major and independent publishers, brand new songwriters and Grammy winners. The demos Cliff has recorded have ended up as major label cuts, in feature films and on television.
For more audio/sound related content and resources, go to Audiofanzine.
Kenton Introduces MIDI USB Host For Interfacing With Standard DIN MIDI Connections
New device can also provide up to 500mA of USB bus power to the USB device if required
The new Kenton MIDI USB Host provides convenient interfacing of USB-enabled devices directly to those with standard 5-pin DIN MIDI connections, either in or out, without the need for a computer…
This enables, for example, direct connection of a USB controller or keyboard to another MIDI instrument that has only a 5-pin DIN MIDI connection.
Mains powered, the compact and rugged MIDI USB Host features a USB port (USB A socket) and MIDI In and Out sockets (both on 5-pin DIN).
MIDI data received at the MIDI In socket will be sent to the USB device, while MIDI data received from the USB device will be sent to the MIDI Out socket.
Additionally, the MIDI USB Host can provide up to 500mA of USB bus power to the USB device if required.
Kenton MIDI USB Host Specs:
Power Input: 5V DC (regulated) – use only the supplied adapter
Power: 90mA, 2.1mm plug (centre positive) – 510mA available for attached USB device
MIDI ports: 1 x In, 1 x Out (both 5-pin DIN)
Weight: 100g (excluding power supply)
Dimensions: 110 x 55 x 32 mm
Power supply: A 5V power supply appropriate to the destination country is supplied with the unit.
Leads: No leads are supplied with the unit
The MIDI USB Host comes with a 12 month (from purchase date) back to base warranty, (i.e. customer must arrange and pay for carriage to and from Kenton Electronics Ltd).
Flexible, Scalable Digital Intercom Systems For A Wide Range Of Applications
Digital intercom systems provide a foundation for point-to-point and group multi-connections
As digital has taken over much of the audio signal path, so it goes as well with intercom systems.
Digital intercom systems provide a flexible and scalable foundation for point-to-point and group multi-connections, and they’re enjoying increased usage across a wide range of live productions and installed applications such as churches, theatre, performing arts centers, stadiums, broadcast facilities and more.
Analog intercom systems often incorporate a patchbay that facilitates connecting (usually via shielded twisted pair, a.k.a., microphone cable) individual stations with a base station. A common concern is a bad cable that causes a short, taking down a channel.
Further, many of these systems are “partyline” – whoever has a headset with a microphone or a station with a mic can join the conversation – whether they’re wanted or not.
The result, for example, is production staff that ends up talking over each other, hampering communication. I’ve even heard a story of a show director sending a stagehand up to a spotlight location with a roll of duct tape, presumably to tape that particular spotlight operator’s mouth shut.
Digital intercom systems eliminate those types of problems in addition to providing vastly increased functionality. The producer in our example above could have the stagehand mute the mic of that spotlight operator from the cozy comfort of the production booth simply using PC software to “click” the mute button for that headset.
Many digital intercoms provide the capability to talk station to station, create an ad-hoc group, reconfigure groups, connect over the Internet to a remote location, bring in a phone call in and route it to a specific station or channel.
The recently introduced ASL Digital Intercom System, which supplies several handy features in addition to intercom capabilities.
Systems usually include a base station with a matrix that offers a scalable foundation for a dozen channels, on up through thousands of channels.
Digital technology provides a lot of flexibility in setting up user-to-user as well as group-based multi-connection communications. Accompanying software suites makes it pretty simple to configure a system for particular applications.
Digital intercoms typically operate on networks that can be configured in daisy chain, line or star topologies, with the system professional (designer/installer) able to choose the approach that is most efficient and cost-effective for the given application. And flexibility continues to improve.
For example, the ASL Digital Intercom allows direct person-to-person communication (like a telephone call) from any user station (including beltpacks) to any other user station, without tying up a channel.
In addition, this system offers text messaging from a keypad loudspeaker station to any other user(s).
In working with digital intercom systems, it’s useful to have a solid understanding of audio and signal flow as well as good working knowledge of networking, telephony and VoIP (Voice over Internet Protocol).
The key to successfully setting up a system is to have a clear plan as to how the system is to be implemented (i.e., using existing wiring, using a facility’s structured cabling, establishing new cabling, and so on).
Further, there needs to be a thorough understanding of any additional connections (like connecting an existing legacy 2-channel partyline system or connecting in a remote location) that need to be made.
Really, though, many of the same principles for designing and implementing a legacy system apply to designing and setting up a digital one.
Here’s a glossary of terms it’s good to be familiar with when working with digital intercom systems:
DTMF – Dual-Tone Multi-Frequency signaling, used for telecommunication signaling over telephone lines in the voice-frequency band between telephone handsets and other communications devices and the switching center.
MADI – Multichannel Audio Digital Interface, an industry-standard electronic communications protocol that defines the data format and electrical characteristics of an interface carrying multiple channels of digital audio.
SIP – Session Initiation Protocol, a signaling protocol widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP).
TCP/IP – Internet Protocol Suite, the set of communications protocols used for the Internet and other similar networks
AES3 – The digital audio standard frequently called AES/EBU used for carrying digital audio signals between various devices.
Star Topology – In its simplest form, a star network consists of one central switch, hub or computer, which acts as a conduit to transmit messages. Thus, the hub and leaf nodes, and the transmission lines between them, form a graph with the topology of a star.
Daisy Chain – Connecting each device in series to the next. If a message is intended for a device partway down the line, each system bounces it along in sequence until it reaches the destination.
Digital intercom systems are usually interconnected just like a computer network, commonly referred to as a LAN (local area network), and just like most audio networks, a digital intercom usually requires its own separate network.
It may run on the same kind of cable as a LAN, but still needs to be wired as its own network.
Note, however, that a couple of years ago, Clear-Com introduced a direct link from its IP-enabled V-Series panels to its Eclipse digital matrix intercom frames over an existing WAN or LAN Ethernet connection, allowing users to establish intercom communications in locations where direct cabling is lacking by employing the existing local IT infrastructure.
Most systems can be interconnected with CAT5, fiber optics and even good ol’ coaxial cable.
For control and configuration of the system, software is loaded on a computer, and the computer is connected to the network.
The computer may be on its own LAN and connected to many other computers, a connection can be made from the computers LAN to the digital intercom systems network giving control and configuration capabilities from anywhere that a computer is connected to that LAN network.
Clear-Com V-Series panel that enables a direct link of the company’s Eclipse digital
matrix intercom frames over an existing WAN or LAN Ethernet connection. (click to enlarge)
With the use of DSP (digital signal processing), very large digital intercom systems can be configured and reconfigured all from a software GUI (graphical user interface).
The bottom line is that digital intercoms, harnessing these technological advancements, can literally be configured into a networked system that spans the globe. Gary Zandstra is a professional AV systems integrator with Parkway Electric in Holland, MI.
APR Audio Brings Allen & Heath iLive Digital Consoles To Dual Stages At Glastonbury 2011
The systems were networked to provide input sharing and remote control via the iLive Editor PC software.
Rental company APR Audio provided systems for the West Holts and John Peel stages at Glastonbury 2011, selecting Allen & Heath iLive digital mixing systems to manage both front of house and monitor mixing duties, which were also networked for audio and control sharing.
On the John Peel stage, APR chose an iDR-64 MixRack connected to the largest control surface in the range, the iLive-176, for monitors. Twelve onstage mixes were provided, plus LR side-fills and up to eight stereo IEM feeds.
Engineer Fabrizio Piazzini additionally used the iLive MixPad app to make final tweaks. For front of house, an iLive-144 Control Surface was chosen, connected via CAT5 ACE to a modular iDR10 MixRack installed onstage.
The systems were networked to provide input sharing and remote control via the iLive Editor PC software.
“One of iLive’s great assets is that it is a networkable system, the benefits of which really come into their own in the fast paced festival environment. For instance, engineers can check FOH settings from the stage without battling through the crowds and mud,” explains Allen & Heath product manager Leon Phillips, who joined the APR team on site.
“Using iLive Editor software, we could drag appropriate channels from the generic patch directly onto the iLive surface layers in real time even while the engineer had already started working on the desk.”
A broadcast feed was also provided, split from the iDR-64 mic preamps and patched via the ACE network to 48 XLR outs on the iDR10 MixRack.
“Working with a generic festival layout is tricky, as you can end up with active channels spaced out all over the board and in different layers but this was not the case with iLive,” explains house engineer Stuart McKay. “The surface worked really well when it came to quickly putting bands’ channels next to each other, and hiding channels/outputs that I did not want visiting engineers to touch.
“Engineers who had never used iLive before soon got their heads around it and found the layout simple and intuitive. The way that ACE was implemented to feed the broadcast sends was really cool and the gain sharing that was in place between monitors and FOH worked really well.”
On The West Holts Stage, APR provided an iLive-144 surface at FOH, and an iDR-64 was built into the stage box input rack.
A compact iLive system, comprising an iDR-16 MixRack and iLive-R72 rackmount control surface, was used to manage compere mics, DJ, and BGM duties at front of house. Delayed near-field monitoring was also controlled by this set up at the front of house tower.
Jay Ruston Deploys Metric Halo ChannelStrip In Recording & Mixing New Anthrax Album
"We still want powerful drums, and to make that happen, we have to be clever. It takes some pretty aggressive processing.” - Jay Ruston
Producer/engineer Jay Ruston has been recording and mixing tracks for artists such as The Donnas, Jars of Clay, Steel Panther, and Meatloaf for about 20 years, and recently utilized the Metric Halo ChannelStrip plug-in for the upcoming Anthrax release, Worship Music, working with Anthrax producer and guitarist Rob Caggiano,
“I heard about Metric Halo and specifically, ChannelStrip, from Rob and from an article I was reading about some mixers and albums I admire,” explains Ruston. “A lot of them commented about how helpful ChannelStrip was for drum tracks. For heavy metal, it’s the drum tracks that need the most post-production processing.
“Twenty years ago, producers recorded a few stereo pairs of guitars, and the drums were able to sit powerfully in the mix without any conflict. These days, everyone gets that incredible crunchy guitar sound by layering massive numbers of tracks. But we still want powerful drums, and to make that happen, we have to be clever. It takes some pretty aggressive processing.”
Ruston downloaded the ChannelStrip demo version and was impressed not only with the sound, but also with the usefulness of the plug-in’s presets. At the time, he was putting the final touches on Steel Panther’s latest album at that time, and wasn’t entirely satisfied with the snare sound he had achieved.
“I brought ChannelStrip in on every single song and ran a parallel snare track through the ‘deep snare’ preset,” he says. “When I blended it back in, the whole thing became way punchier. I was glad I discovered it in time! When the drummer heard it, he flipped out, it sounded so good.”
For Anthrax, Ruston found that the kick and snare were too loud in the room mics, which was especially disappointing because he really needed the room to get the right atmosphere on the drums. He used ChannelStrip to hard-limit the room tracks and to pull back some of the high-end wash from the cymbals.
Then he mixed that back with an unprocessed track to get the right mix of punch and room. “It sounds incredible,” he says. “Way bigger and way punchier than I would have guessed possible. And while I think it might have been possible to achieve similar results with other tools, the whole thing happened very quickly and intuitively with ChannelStrip. It’s a very musical plug-in.”
He continues, “I know everyone talks about how an EQ needs to be musical, and ChannelStrip has that for sure. But it also has that element of surgical precision that allows me to home in on and fix very specific problems.”
In fact, Ruston finds the ChannelStrip EQ so “musical” that he has stopped relying on his Trident’s EQ. Although it was nice to include on some very special parts, relying on outboard EQ made recalling a mix time-consuming and somewhat unreliable.
“In 10 minutes, I had replicated the Trident’s EQ sound with ChannelStrip. It’s interesting that the ChannelStrip EQ is just as responsive as an analog EQ. A few dB often does the trick, whereas with other digital EQs, I often have to go to 6 dB or more to really hear an effect.”
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