Tuesday, April 07, 2015
Real World Gear: Most Valuable Players
The latest developments in 2-way compact loudspeakers
Available in a range of sizes and cabinet shapes, compact 2-way loudspeakers are an MVP (Most Valuable Player) in the live sound world. They serve as mains, monitors, side fills, center fills, near fills, front fills and delays, and can quickly be ground stacked, flown, or placed on a stand. In addition, they can almost always be carried by one person.
Many (most) of these boxes are trapezoidal in shape, with many offering an enclosure angle for stage monitoring. Smaller 2-way boxes incorporate a single 8- or 10-inch woofer, but the most popular models offer 12- or 15-inch woofers for additional low-end performance.
Usually the woofers are accompanied by a compression driver on a horn or waveguide for mid and high frequencies, although ribbon drivers have also emerged as a viable option from certain companies. Also don’t overlook coaxial models where the individual driver units radiate sound from the same point/axis, which, when designed properly, can offer enhanced coherence.
When evaluating 2-way loudspeakers, start by defining the right box for the job – size, scale, portability, and so on. It all depends on the requirements of the application(s). Further, many models are available in either passive or active versions with (usually) class D amplification and DSP, and increasingly, networking.
In getting to a more refined “apples to apples” comparison, factors to consider include dispersion, power handling, sensitivity/maximum SPL, and mounting options.
—Dispersion, a measurement of the pattern of MF/HF sound that emanates from the box. This is stated in degrees for the horizontal and vertical planes.
—Power handling, which, for passive cabinets, is usually stated as an “RMS” or “continuous” rating in watts. An increasing number of these loudspeakers are now self-powered and also have onboard DSP.
—Sensitivity, stated in decibels (dB), is a measurement of the sound level the loudspeaker can produce with a given input signal, generally measured with 1 watt input at 1 meter distance. (By the way, we’re seeing an increasing number of manufacturers who prefer to provide a maximum SPL specification.)
—Mounting, which includes integrated flypoints as well as things like pole mounts that can come in quite handy for true portable applications.
Our tour of recent models that is linked here is intended, by design, to present the “state of the market” in terms of options. But for each type of model presented here, understand that there are dozens of potentially viable options, so further homework is strongly recommended.
The real workhorses of pro audio are traditional 2-way loudspeakers. They remain invaluable for a range of very good reasons, with versatility that translates to “great bang for the buck” topping the list. Whether placed on a stand for a speech at a groundbreaking ceremony, stacked on top of subwoofers at the local music venue, or flown at a corporate event, the ubiquitous conventional 2-way loudspeaker gets the job done
It can be tough performing live on a stage – all the noise, Noise, NOISE. This is the first thing in my mind when doing stage monitoring because too much noise can compromise performances and presentations.
It’s hard to be at your best when you can’t hear, let alone think, due to a sonic assault, and no amount of magic monitor mixing or technique is going to fix it.
The noise comes from everywhere. The backside of the house mains. The subwoofers. Various (and numerous) delayed signal reflecting back from the house. Stage sources. The audience. There’s only so much we can do, but minimizing stray energy on stage is a top priority, and then comes the work of devising a monitoring approach that’s best for the performers and production.
When setting up the house system, placement and people are the keys. Placement is where the main loudspeakers are stacked or flown, and people refers to the performers and the audience – we need to serve both.
If possible, the loudspeakers shouldn’t be too close to the stage, helping keep their excess energy off the stage while still insuring they fully cover the audience. Flying/stacking them even a few feet forward can make a significant difference. (Also try to keep their output off of reflective surfaces in the room.)
Loudspeakers on delay further out in the house, as well as front fills, are a good way to get even coverage throughout the listening area without having to crank up the mains too high. Digitally steerable arrays, which my company deploys regularly for both music performances and corporate events, provide an extra degree of welcome control. Cardioid and end fire configurations for the subs can help direct their energy outward rather than backward.
A very clean deployment of monitors and fills.
Now it’s time to turn our attention to the stage monitoring needs of the performers. Wedges, in-ear monitors, stage fills, or a combination?
It’s a direction determined by the specific gig, performer preferences, and what we (and perhaps the venue) have available in terms of gear. Everything but in-ear monitoring increases the noise quotient, but that’s usually (and simply) the reality; plus, some artists like amped-up monitoring.
Over the past several years, a wide range of active 2-way boxes have hit the market with cabinets offering a monitoring angle. They’re quite useful in being able to perform wedge, stage fill and small system main duties. Onboard DSP means they can be optimized for the specific application.
We’re also seeing more active options with traditional wedges. Active does require both a signal and power cable, but there are now options offering both within the same jacket, which can help reduce clutter.
Otherwise, the choice is passive wedges and loudspeakers. They can also be bi-amped (2-way, lows and highs) or tri-amped (3-way, lows, mids, highs), requiring separate amp channels for each section, and may also require an external crossover or processor. Active or passive, options include:
—Mini/personal monitors. Designed to be used very close to a performer, they can be located on the floor, placed on an instrument like a keyboard, or mounted on a microphone stand. The object is to enhance certain parts of the mix (i.e., the vocal), not usually provide the entire mix.
—Standard wedges. Purpose designed and positioned on the floor in mono or sometimes stereo configuration. The most popular types are 2-way designs with a 10-, 12-, or 15-inch woofer and a 1-, 1.5- and 2-inch compression driver mounted on a horn. This category also includes the previously mentioned active loudspeakers. And, a popular variation are coaxial designs that align the LF and HF drivers while occupying a smaller footprint.
—Larger wedges. Usually 3-way systems with a bit more thump and oomph, with the trade-off of being larger and heavier.
—Drum boxes. A specialty loudspeaker to reproduce low frequencies to help drummers better hear the kick (and sometimes bass guitar).
—Drum fills. A single larger loudspeaker, or a group of loudspeakers, to better serve the drummer. Often accompanied by a sub for extra thump.
—Side fills. Also called stage fill, loudspeakers at the sides of the stage, often joined by subwoofers, to augment the output of the wedges and/or IEMs. Some bands prefer side fill exclusively.
—Stage subs. Smaller subs used underneath or alongside a wedge to help augment LF, and these can also be deployed as part of the drum fill or side fill.
Welcome to the hot seat, otherwise known as monitor beach.
These options can be used alone or in any combination, and with IEMs. Just remember, the more loudspeakers on stage, the louder it’s going to be, and there’s increased chance of feedback, both in general and if stationary mics aren’t carefully placed. Plus more boxes can clutter the stage.
Making It Personal
Some performers love IEMs, others won’t use them, still others are in between, wearing them in tandem with monitoring via loudspeakers. Obviously IEMs cut stage noise and can protect hearing (if they’re not abused) through isolation. A lot of drummers prefer to wear headphones for further isolation.
Earbud options range from generics with replaceable tips of either foam or plastic to custom units molded for the user’s ears. The key is getting a good fit to provide adequate isolation as well as enough comfort so they can be worn for the duration of a show.
IEM/personal monitoring systems are usually wireless, with the performers wearing a small belt pack receiver with volume control. Wired systems are a less expensive option for relatively stationary musicians.
Systems are available in mono or stereo, the latter being the more popular choice. One way to deal with performers feeling too isolated is to add some ambience from the audience into their mixes, captured via an audience microphone or two placed on stage and pointed at the crowd, or flown/placed on stands in the audience area (or front of house).
Several personal mixing systems provide individual mini mixers to performers so they can tailor their own mixes. And a cool recent development is that many digital consoles/mixers now work with custom apps that allow performers to tailor their mixes via tablet or smartphone. Make sure these onstage devices can only access the one mix, and not affect other monitor mixes (or the house mix) by accident.
I mentioned that IEMs offer hearing protection via isolation, but keep in mind that they shouldn’t be turned up too loud. (Kind of defeats the whole purpose.) Also note that a squeal of feedback or loud thud from a dropped mic can be damaging to hearing, so compression and limiting, even brick wall limiting, may be needed to control any unexpected spikes.
Some shows may just use a few aux sends from the house console to feed the monitors. This can work well for smaller gigs with simple monitoring requirements, but on larger shows, a monitor console is necessary. It should be manned by a dedicated operator, and is usually placed stage side in sight of the performers.
Be sure to label IEM belt packs.
Inputs are shared between the consoles via a split snake in an analog transport system or just grabbed off the network in a digital transport system. The better analog splitters use transformers between the consoles to eliminate hum or buzz, but hard-wired splits can work OK if the system has no grounding and noise issues.
Monitor engineers can make use of a cue wedge at the console to hear each mix if there are wedges used onstage, or they may use IEM to cue up each mix and hear what the performers hear. Many digital consoles allow remote control via tablet, a boon for engineers because they can stand onstage next to performers during sound check and fine tune their mixes.
I’ll conclude with some tips that I’ve picked up over the years in working with monitoring over the years.
Acoustic Aiming Devices. I always carry some AADs (black painted pieces of wood) to tilt wedges to put the performers into the coverage pattern. If they can’t hear it clearly, they’ll want more volume.
Shakers. Some performers (drummers in particular) want to really feel the low end, and a seat shaker (a.k.a., butt kicker) can provide this while eliminating the need to add subs. They can also be used with smaller wedges to keep down stage volume.
Directional Subs. With monitor engineers usually at the side of the stage, the wash from the side fill subs can be problematic. A cardioid approach with these subs can help, and some manufacturers offer single cardioid units.
Parametric. Not every problematic tone falls right on the center frequencies, so parametric equalizers deliver added (and needed) precision.
Backup Wedge. Even if all performers are on IEMs, having an extra wedge comes in handy for use with announcers, guest artists and audience members brought onstage, as well as for talkback communication and in case an IEM system goes south.
Identification. Label all wedges with their respective mix numbers. It’s way easier and clearer for a performer to ask for more cowbell in mix 9 than ask for more “in that box.” Also label all IEM belt packs with mix numbers and performers names so they don’t accidentally grab the wrong pack.
Better Reception. Deploying quality antennas with IEM systems can help eliminate dropouts and other RF problems. Directional “paddle-style” (a.k.a., log-periodic dipole array) antennas can provide up to 6 dB of gain and Helical antennas can provide up to 10 dB.
In The Pan. Stereo IEM mixes have more depth and are easier to listen compared to mono. I place performers center in their own mixes and pan the other instruments to the left and right (depending on where those instruments are onstage). And we tailor it together from there…
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.
Compact Benefits: Small & Mid-Sized Digital Consoles
In the beginning there was resistance to digital consoles, but the tide has turned and the advantages of digital live sound workflow are so numerous that we now take many for granted.
An obvious advantage is size and weight. Compact digital desks weigh 50 pounds or less, and a 100-meter Cat-5 snake fits in the back of stage box racks. Analog consoles and their associated outboard gear, multi-core snake and splitter take a couple rows in a truck instead of the back seat of a crew cab.
With digital snakes, the hums and buzzes of yesterday are mostly gone, while putting preamps on stage improves sound quality. Gain-sharing means consoles don’t need a splitter, and mix engineers don’t have to listen to their mixes through transformers.
“In-the-box” mixing, where mix processing is performed entirely within a digital console, means engineers can save shows and open them on identical or similar consoles. They can email that file to the next gig, whether across the country or around the world, “cc-ing” a copy as a backup.
Though layered consoles with fixed architecture have been an annoyance to those who grew up with analog, newer designs improve workflow with the ability to “spill” control group members onto fader banks and customization of layers can improve mixing. Remote control of a console from something as inexpensive as an iPad has become a standard feature and a powerful tool, allowing monitor engineers to stand beside performers while front of house engineers check the entire listening area – both with controls in hand.
USB 2.0 has enabled digital console manufacturers to provide multi-track I/O for recording and playback to laptop-based DAWs – and even direct to hard drives in some cases – providing affordable “virtual sound checks” to many applications. This simple innovation allows engineers to easily check a previous show, test sound systems with a previous performance, practice mixing or tweak a show file, teach others to mix, and afterwards, easily mix a show down for distribution using the same console.
Another advent is simple 2-track recording and playback using USB “thumb drives,” allowing engineers to walk away from a console with a board mix they can easily listen to and then e-mail to others if they like. It also provides simple and foolproof playback of walk-in and intro music with no moving parts.
Thirty years ago, 32-input channels was a standard that covered any band and even festivals. Of course, this was before the days of double-miking nearly everything on stage. When I worked at Sun Sound Audio, we took our flagship 32-channel Soundcraft 800 to many New England concerts, colleges, festivals, and even on my first tour with Crystal Gayle.
By 1989 I found myself at the Capital Theater in Port Chester, NY helping Jonny Podell put Duane Allman’s and Dickie Betts’ bands back together (again), with two drum kits and The Band thrown in as an opener, all mixed on that same Soundcraft 800-32. Yes, we used a few Y-cords, but today I’m sure that show would “require” 96 channels. Point being, 32 channels can do a lot, and with today’s digital workflow, so much more.
Yamaha first introduced the “0” series in the ‘90s, and the youngest sibling, the 01V, has long been a benchmark for compact digital consoles. Stan Miller and Bernie Becker took 14 of its predecessor, the Pro Mix 01, on Neil Diamond’s tour as sub-mixers feeding stereo stems into a 24-channel Yamaha PM3500. In honor of this trailblazer, we lead off the listings here with the 01V96i – to this day thousands are used in smaller venues and recording studios.
With the impending release of his new album next week, Joe offers an insight on something he picked up while working with one of the songs.
The lesson: sometimes allowing things to go against what you think you should do, in order to get the best performance and make the most of a song, is the right way to go.
The song is “Forgiveness,” a simple, soft, intimate piano ballad-type track. Joe shares what it sounded like at the beginning of the recording process when he first started working on the album, creating the scratch tracks using an automated click track.
As he explains (and you’ll quickly hear), the click wasn’t working with this song; in fact, quite the opposite. It seems in direct opposition, working against the song and performance. How did he resolve the issue? Check out the video to find out.
Church Sound: The Biggest Key To Mixing Like A Pro
In my 30-plus years as a professional mixer, I’ve mixed thousands of events ranging from live concerts for thousands of people, to television broadcasts reaching millions, all the way down to intimate club shows for artists in front of high-level music executives where the artist’s career may hang in the balance.
But I must confide that I’m the most anxious while mixing events for houses of worship.
Perhaps it’s because, as a believer, I realize just how much is at stake for the listeners during a worship or praise event, but honestly, there have been times that the butterflies have felt more like dive bombers in my gut when sitting down at the console for one of these events.
But if mixing all of the varied styles of live events in my life has taught me anything, it’s simply this: the only way to truly neutralize the butterflies, or the dive bombers in this case, is with preparation.
Over the past decade or so, I‘ve lead a number of seminars and workshops on the topic of “mixing” and, without fail, I’m constantly amazed by the attendees’ reactions at the conclusion, when they arrive at the stark realization that successful mixing is based on much more than learning to operate a complex console, or some highly touted routing or EQ manipulation.
Now granted, knowing how to actually operate the mixer is important, but as I have been known to say from time to time—“a great pipe wrench does not make a great plumber.”
So what I’m getting at is this: mixing— especially music and event mixing—is an approach and a mindset, not simply a task. And it requires a method. That method has to include a way to anticipate moves before they need to happen.
The Problem & Solution
I’m sure you all have heard the event where it feels like the guy mixing has set his watch about two minutes slower than everyone else’s.
The pastor’s mic comes on after he starts speaking, the audio for the video comes up well after the video actually begins, you hear the backing vocals well after they start singing their parts, you hear the guitar solo about a bar after the guitarist starts playing it, the worship leader prays at the end of the song with all the effects glaring on his or her vocal—for the first half of the prayer.
All of these kinds of happenings are signs of a mixer who is not anticipating but simply reacting to what is happening. The ol’ tail is officially wagging the dog.
No matter how accomplished a tonemeister we become, or how many consoles and effects we know how to operate, if we mix like this, we’re imparting some serious disruption to the worship experience and severely handicapping the pastors and performers from creating any kind of “moment” for the congregation to experience.
So how do we combat this? At the root of changing this is some simple preparation and a change of mindset by you as the mixer.
And I want to put up the disclaimer here: Digital consoles can be very effective in helping us manage certain tasks during an event. However, they do not afford us less preparation. In fact, they put a bit more pressure on us to pre-prepare.
A very good friend of mine and a fabulous mixer in his own right preaches and practices the following mindset to interns and budding audio professionals. And in my opinion, he’s right on the money. You and your team may be well-served by employing this little ditty: “If you’re early, you’re on time, and if you’re on time, you’re late.”
Granted, in the house of worship world, because of the high proportion of volunteers involved, time, generally speaking, is not money. But don’t let that lull you into believing time is not cherished by those involved. In fact, it may be even more highly scrutinized because people are freely offering it.
One of the sure-fire ways to see your team’s focus and enthusiasm wane is to have them waiting around for you as the sound guy to make even a simple patch, or find that elusive extra DI that you didn’t know you were going to need until the extra player showed up. What may only take a few minutes can seem like an eternity to a musician or a service director.
Frankly, you and your team’s energy and rehearsal time needs to be spent refining and learning what is going to transpire during the service—not doing what is generally considered utility or task work.
Three Important Planning Documents
So, with that in mind, I always take the time and initiative to assemble three documents, in this order: a stage plot, an input list, and a cue sheet.
Nothing you do as audio mixer will save you and your team more time and energy, and in turn make you look more on top of your game, than taking the time to prepare and implement these three documents.
For the stage plot, work with the service director to assemble it. It essentially shows where the pastor, actors, musicians, and their instruments will be located on the stage. You can also include information on where you’ll place monitor wedges or which musicians will be working with in-ear monitors, where you might need power drops, etc.
This will allow you to have mics, DIs, stands, and monitors all within arm’s length of where the musician will set up before they walk in the door, allowing for nearly instant patching of your sources. It can also help you preempt and diffuse difficult transitions between events and elements of your service.
For the input list, you—not the director—should take the initiative to question the musicians and find out what instruments they’ll be bringing, and then list it out in a simple spreadsheet. Lay it out just as you would lay out your console and have it show everything you deem relevant about a given console input. Include info such as what microphone or DI you’ll use, what snake line or stage box the mic will be patched into, what kind of mic stand you’ll need.
Possibly have a “misc.” column that lists things such as quarter-inch cables needed, etc. You can even get console-specific and list what inserts you’ll need for a given input and include notes about its associated patch. This is especially helpful when using analog mixing systems.
There’s nothing wrong with having extra items handy and ready to go at a moment’s notice. Maybe the guitar player will bring an extra acoustic just in case of an arrangement change, or the drummer may bring an extra drum for a given song.
It will always be better to have extra DIs sitting patched and ready to go and not need them than it is to scramble to find and patch them while everyone is waiting on you to make it all happen. The depth of your preparation is only limited by your imagination and what your situation typically demands.
For cue sheets, again work with the service director and make simple but detailed notes for every “audio event” that happens in the service.
I usually layout every event that happens from “walk-in music” to “CD-R record ready” to “verify drama mics and stand by” to “mute acoustic during the breakdown” to “walk out music fades up” etc. With many digital consoles, you can incorporate a cue sheet note into every recallable scene on your console and have it displayed prominently on the GUI. Very handy indeed.
Believe it or not, this seemingly simple process has many overarching side effects. First, it’s a huge time-saver for everyone involved when time is the most coveted factor in rehearsal. It will also help you dramatically build self-confidence because you’ll be on top of the game and prepared for the job at hand, which will please the musicians and pastors and make for a much better performance.
And finally, it will relax you and allow you to stay focused on your most important duty: anticipating all the moves you’ll need to make throughout the service, and not simply reacting to them.
Robert Scovill is a veteran of professional concert sound and recording, and has mixed over 3,000 live events in his career. His engineering and production talents have been enlisted by a veritable “Who’s Who” of marquee music acts, including Tom Petty & The Heartbreakers, Matchbox Twenty, Prince, Rush and Def Leppard. Scovill’s body of live sound and recording work has garnered numerous industry accolades including six TEC Awards for technical excellence and creative achievement in sound reinforcement.
In the world of systems integration, talking about the diminishing margin in hardware starts to feel like the movie “Groundhog Day.” We all know it is an issue, we all feel it in our bottom lines, but we struggle to bridge the shifting economics.
While we all have a blast talking about our problems, can we agree it is more fun — or at least more productive — to solve them? In order to solve our problems, we first must understand the root of them.
In the case of commercial integration, diminishing profitability is rooted in a few things. While most notably it is the diminishing margins on hardware, it is important not to ignore many commercial integrators have shifted from being product suppliers to service providers. In this shift, not all integrators have been created equal.
As margins have moved from product to project, it is upon the business leaders to solve the conundrum of how to make the intangible projects as dependable as the hardware that once drove big margins. A great place to start is by looking at where margins get lost in a project by spending some time analyzing our respective organizations to determine if we can get those margins back.
Are you looking for your project profitability? Here are eight places to start.
1) The Wrong Suppliers. I won’t pull any punches here. Every company should take a close look at all vendor relationships, their programs with those vendors and what improvements can be made. This can drive two-to-four percent to the bottom line if it is made a focus from top to bottom.
2) Over Incentivizing. Sales are unquestionably the lifeblood of your company. But if your sales team is overcompensated, that immediately eats up potential profits. A good rule of thumb is a sales person should be paid (including benefits) no more than 25 percent of the gross margin they are creating. In my experience 20 percent is more manageable.
3) Design Efficiency. Customers don’t want drill bits, they want quarter-inch holes in the wall. If your design is heavy in equipment dollars then you probably have to reduce services and high margin dollars to meet the budget. Can you design more efficiently?
4) Forgetting G&A. Some clients don’t want to see your charges for administrative support, but that doesn’t mean those costs don’t represent 5-10 percent of your revenue. Since someone needs to order the equipment, bill the client and process your payroll, those costs need to be considered when proposing projects. Whether that adds $30 to a small job or $30,000 to a large one, it adds up and is highly problematic to the bottom line if it isn’t accounted for.
5) Time Management. Chances are you are paying your installers and technical resources for a full day’s work. The question is: Are you getting a full day’s work? Starting 15 minutes late and taking 10 extra minutes for lunch don’t appear on the surface to be a big problem. But if you have five, 10 or more technical staff all doing the same thing, those 25 minutes can turn into thousands over a year. Divide that by your hourly rate and your projects are bleeding unnecessarily.
6) Capturing Freight. With margins on hardware continuing to decrease, getting your freight pricing right is critical. Using tools to estimate inbound and outbound freight and properly charging for this is key. Also, using covenants to charge appropriately for change orders that require air freight or other highly expensive means of product acquisition are helpful in improving overall project profitability.
7) Missing the Little Things. In the quoting phase, it is easy to forget a cable, DA, or mount. When estimates are being turned quickly this can become even more problematic. “Check 11 times, cut once,” is the mantra. It is very hard to charge for hardware adds that were mistakenly forgotten on your end, so thoroughly audit your equipment list and consumables before finalizing your quotes.
8) You Aren’t Charging Enough. I know this sounds obvious, but I used to ask myself after winning a bid or project what we did wrong. While there was a certain kidding in that gesture, there was always the fear that the winning bid on price-sensitive proposals was really the loser. Bottom line: winning projects by underestimating the true cost or requiring unrealistic precision is a great way to not make money.
Driving margin back into your projects is a game changer for your bottom line. You have other ways of trying to solve the same problem, but these eight steps are critical to maintaining your company’s profitability.
So where are your projects losing margin? More importantly, what are you doing to get it back?
Daniel L. Newman currently serves as CEO of EOS, a new company focused on offering cloud-based management solutions for IT and A/V integrators. He has spent his entire career in various integration industry roles. Most recently, Newman was CEO of United Visual where he led all day to day operations for the 60-plus-year-old integrator.
Having trouble figuring how to use your effects during mixing? Here’e a set of rules that can help you choose the best effects for each track more efficiently, courtesy of The Mixing Engineer’s Handbook.
Rule 1: As A General Rule Of Thumb, Try To Picture The Performer In An Acoustic Space And Then Realistically Recreate That Space Around Them. This method usually saves some time over simply experimenting with different effects presets until something excites you (although that method can work too). Also, the created acoustic space needn’t be a natural one. In fact, as long as it fits the music, the more creative the better.
Rule 2: Smaller Reverbs Or Short Delays Make Things Sound Bigger. Reverbs with decays under a second (and usually much shorter than that) and delays under 100 milliseconds (again usually a lot shorter than that) tend to make the track sound bigger rather than push it back in the mix, especially if the reverb or delay is stereo.
Rule 3: Long Delays, Reverb Predelays, Or Reverb Decay Push A Sound Farther Away If The Level Of The Effect Is Loud Enough. As stated before, delays and predelays (see below) longer than 100 ms (although 250 is where it really kicks in) are distinctly heard and begin to push the sound away from the listener.
The trick between something sounding big or just distant is the level of the effect. When the decay or delay is short and the level loud, the track sounds big. When the decay or delay is long and loud, the track just sounds far away.
Rule 4: If Delays Are Timed To The Tempo Of The Track, They Add Depth Without Being Noticeable. Most engineers set the delay time to the tempo of the track (see below on how to do this). This makes the delay pulse with the music and adds a reverb type of environment to the sound. It also makes the delay seem to disappear as a discrete repeat but still adds a smoothing quality to the element.
If you want to easily find the right delay time to the track and you have an iPhone, grab my “Delay Genie” app from the iTunes App Store. It’s free and will making timing your effects to the track incredibly easy.
Rule 5: If Delays Are Not Timed To The Tempo Of The Track, They Stick Out. Sometimes you want to distinctly hear a delay and the best way to do that is to make sure that the delay is NOT exactly timed to the track. Start by first putting the delay in time with the track, then slowly alter the timing until the desired effect is achieved.
Rule 6: Reverbs Sound Smoother When Timed To The Tempo Of The Track. Reverbs are timed to the track by triggering them off of a snare hit and adjusting the decay parameter so that the decay just dies by the next snare hit. The idea is to make the decay “breathe” with the track. The best way to achieve this is to make everything as big as possible at the shortest setting first, then gradually get longer until it’s in time with the track.
Of course, the biggest part of adding effects to a mix is experience, but following these rules will provide a perfect place to start.
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website, and go here for more info and to acquire a copy of The Mixing Engineer’s Handbook.
Capturing The Moment: Microphone Techniques For Live Recording
Perhaps the most exciting type of recording comes in the live realm, whether it be in a club or concert hall or stadium.
Many musicians and bands want to record live because they feel that’s when they play best. The goal, then, is to capture the performance so it can be brought back alive.
Remote recording is exhilarating. The musicians - excited by the audience - often put on a stellar performance. Usually you only get one chance to get it recorded, and it must be done right. It’s on the edge, but by the end of the night, especially if everything has gone as planned - what a great feeling!
Challenges abound. The monitors can feed back and/or leak into the vocal microphones, coloring the sound. The bass sound can leak into the drum mics, and the drums can leak into the piano mics.
Then there are other mic-related gremlins - breath pops, lighting buzzes, wireless system glitches, and even electric shocks.
How to get around the potential problems? Let’s have a look at some effective mic techniques that work well when recording in the live realm. And note that these are tailored more to “pop” music performances.
- When using directional mics, position them close to the source. Close mic’ing increases the sound level at the mic, so less gain is needed, which in turn cuts background noise and leakage.
Unidirectional mics (cardioid, supercardioid, hypercardioid) do the same thing by attenuating off-axis sounds. Also, their proximity effect boosts the bass up close, without boosting the bass of distant sounds.
- Use direct boxes and guitar pickups to eliminate leakage. Or use pickups mixed with mics.
- Consider using headworn noise-canceling mics on vocals. A noise-canceling or differential mic is designed to cancel sounds at a distance, such as instruments on stage or monitor loudspeakers. Such a mic provides outstanding gain-before-feedback and isolation. The mic must be used with lips touching the foam windscreen; otherwise the voice is cancelled.
- Use wireless mics correctly. If dropouts can be heard, move the wireless receiver (or remote antennas) closer or to a point where a stronger signal can be realized. If distortion occurs with loud yelling, turn down the gain-trim pot in the mic.
- Prevent hum and buzz.Keep mic cables well separated from lighting and power cables. If the cables must cross, do so at right angles to reduce the coupling between them, and separate them vertically.
If hum pickup is severe with dynamic microphones, use dynamic microphones with humbucking coils built in. Routinely check the microphone cables to make sure the shield is connected at both ends. For outdoor work, tape over cracks between connectors to keep out dust and rain.
- Prevent electric guitar “shocks.” There may be a ground-potential difference between the electric guitar strings and the sound system mics, causing shocks when both are touched. It helps to power all instrument amps and audio gear from the same AC distribution outlets.
That is, run a heavy extension cord from a stage outlet back to the mixing console (or vice versa). Plug all the power-cord ground pins into grounded outlets. This prevents shocks and hum at the same time.
Further, try putting a foam windscreen on each vocal mic to insulate the guitarist from shocks. As a bonus, a foam windscreen suppresses breath pops better than a metal grille screen. If you’re picking up the electric guitar direct, use a transformer-isolated direct box and set the ground-lift switch to the position with the least hum.
- Try mini mics and clip-on holders. Nearly all microphone manufacturers offer miniature condenser models. These tiny units sometimes offer the sound quality of larger studio mics. If clipped on musical instruments, they reduce clutter on stage by eliminating boom stands.
Plus, the performer can move freely around the stage. And because a miniature clip-on mic is very close to its instrument, it picks up a high sound level.
Often, an omni mic can be used without feedback. Note that “omni’s” generally have a wider, smoother response than “uni’s” and pick up less mechanical vibration. Clutter can also be lessened even when using regular-size mics by mounting them in mic holders that clip on drum rims and mic stands.
As always, there is no one “right” way to mic an instrument. The suggestions here are techniques that have been proven to work, but never hesitate to use what feels best for your situation.
Vocal. Cardioid dynamic or condenser handheld mic, maybe with a presence peak around 5 kHz, and always with a foam windscreen to reduce breath pops.
Lips should touch the foam for best isolation. Aim the rear of the mic at floor monitors to reduce monitor pickup and feedback. Use a 100 Hz low-cut filter and some low-frequency roll-off to reduce pops and to compensate for proximity effect. Acoustic guitar. Consider using a cardioid condenser on guitar, between the sound hole and 12th fret, a few inches away. Roll off excess bass. Aim the mic downward to pick up less vocal. Other approaches include using a direct box on the guitar pickup and placing a mini mic near the bottom edge of the sound hole. Roll off excess bass. (Figure 1)
Saxophone. Mount a shock-mounted cardioid on the instrument bell. Or, try a mini omni or cardioid condenser mic clipped to top of bell, picking up both the bell and tone holes a few inches away. (Figure 2)
Figure 1: Some acoustic-guitar mic’ing techniques. (click to enlarge)
Electric guitar. To add some guitar-amp distortion, mic the amp about an inch from its speaker cone, slightly off center, with a cardioid dynamic mic. A leakage-free alternative is to use a direct box and process the signal during mixdown through a guitar-amp modeling processor or plug-in.
Electric bass, synth, drum machine. Go with a direct box.
Leslie organ speaker. Cardioid dynamic mic with a presence peak a few inches from the top louvers. Add another mic on the lower bass speaker.
Figure 2: Mobile techniques for saxophone. (click to enlarge)
Drum set (toms and snare). Cardioid dynamic mic with a presence peak, or a clip-on cardioid condenser mic, about 1 inch above the head, 1 inch to 2 inches in from the rim, angled down about 45 degrees to the head.
Drum set (cymbals). Using one or two boom stands, place cardioid condenser mics (flat or rising high-frequency response) 2 feet to 3 feet over the cymbals. The mics can be spaced 2 feet to 3 feet apart, or mounted “XY” style for mono-compatible recording. A stereo mic can also be used effectively. (Figure 3)
Figure 3: A strategy for mic’ing all parts of a drum set. (click to enlarge)
Drum set (kick drum). Remove the front head or go inside the hole cut in the front head. Inside, on the bottom of the shell, place a pillow or blanket pressing against the beater head. This dampens the decay portion of the kick-drum’s envelope and tightens the beat.
Place a cardioid dynamic mic with a presence peak and a deep low-frequency response inside a few inches from the beater. For extra attack or click, use a wooden beater and/or boost EQ around 3 kHz to 6 kHz. Cut a few dB around 400 Hz to remove the papery sound.
Drum set (simple miking). For jazz or blues, sometimes you can mic the drum set with one or two condensers (or a stereo mic) overhead, and another mic in (or in front of) the kick. Note that there may be a need to mix in another mic near the snare drum.
As an alternative, clip a mini omni mic to the snare-drum rim, in the center of the set, about 4 inches above the snare drum. With a little bass and treble boost, the sound can be surprisingly good. Put another mic in the kick.
Metal percussion. Use a flat condenser mic about 1 foot away.
Bongos or congas. Place a cardioid dynamic near each drum head.
Grand piano. Tape a mini mic or boundary mic to the underside of the raised lid in the middle. For stereo, use two mics: one over the bass strings and one over the treble strings.
And for more isolation, close the lid and tweak EQ to remove the tubby coloration (usually cut around 125 Hz to 300 Hz). Or, raise or remove the lid. Place two flat condenser mics 8 inches over the bass and treble strings, about 8 inches horizontally from the hammers, aiming at them.
One other approach is to put the bass mic about 2 feet nearer the tail, aiming at the sound board. (Figure 4)
Upright piano. Use two cardioid mics facing the sound board, a few inches away, dividing the piano in thirds.
Figure 4: Not one, but two piano-miking methods! (click to enlarge)
Xylophone or marimba. Deploy two flat-response condensers 18 inches above the instrument and 2 feet apart.
Banjo. Tape a mini omni mic to the drum head about 2 inches in from the rim, or on the bridge. Or, place a flat-response condenser or dynamic mic 6 inches from the drum head, either centered or near the edge.
Fiddle/violin. Mini omni mic. Put a small foam windscreen on the cable 1.5 inches behind the mic head. Stuff the foam in the tailpiece so the mic head “floats” between the tailpiece and bridge. Another approach is to use a cardioid dynamic or condenser mic about 6 inches over the bridge.
Mandolin, bouzouki, dobro, lap dulcimer. Flat-response cardioid condenser about 6 to 8 inches away from a sound hole is often the best option.
Figure 5: Three ways to handle that pesky acoustic bass. (click to enlarge)
Acoustic bass. Try a flat-response cardioid a few inches out front, even with the bridge. Or, tape a mini mic near an f-hole and roll off excess bass.
Another option: wrap a cardioid dynamic mic in foam and stuff it in the tailpiece aiming up. Cut EQ around 700 Hz for tailpiece miking. (Figure 5)
Brass instruments.Place a ribbon or cardioid dynamic about 8 inches from the bell.
Woodwind instruments.Use a flat-response cardioid condenser placed 8 inches from the side - not in the bell.
Flute. Try a cardioid mic near mouthpiece, and using a foam pop filter. Or, use a mini omni clipped on the instrument, resting about 1.5 inches above the zone between mouthpiece and tone holes.
Harmonica. A very closely placed or handheld cardioid dynamic mic is usually the way to go.
Accordion, concertina. Employ a cardioid about 8 inches from the tone holes on the piano-keyboard side. Mini omni mic taped near tone holes on the opposite side (because it moves).
Audience. This is an interesting one! It can be done with two spaced cardioids on the front edge of the stage aiming at the back row of the audience. (Figure 6)
Figure 6: Get the audience into the action! (click to enlarge)
Another way is to use two spaced cardioids hanging over the front row of the audience, aiming at the back row. Or, try two mics at front-of-house (FOH).
To prevent an echo between the stage mics and FOH mics, mix the on-stage mics to stereo, then delay that stereo mix relative to the FOH audience mics until their signals align in time.
Keep in mind that each of these techniques involves some compromises in order to fight background noise and leakage, but with some careful EQ, they can put you well on the way to a quality recording.
AES and SynAudCon 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.
Knowing the audio path through a mixing console is absolutely critical to sound tech/engineer success.
Using this information, an engineer can quickly troubleshoot the likely causes of common problems, and can even narrow down the possibilities of unexpected major problems.
It can also prevent mistakes because you know what the audio is doing at each stage of the console, and it instills confidence as you sit behind the console, fulling knowing the the ins and outs (sorry for the pun) of the equipment.
Finally, it provides a foundation of understanding which makes it easy to move from room to room or console to console and not be thrown for a loop.
For instance, you might think “the second red knob on my old console was always set to 12:00, does that mean the second blue knob on this console should be set the same?” However, after carefully studying a console’s signal path, you’ll know exactly what that knob is and where it is in the audio signal chain (even consulting the owner’s manual if necessary.)
You want to be an excellent all-around driver of vehicles, not a specialist who only knows and drives a Chevy Malibu 2-door with the small V6.
In general, the controls that you tend to “set and forget” are at the top of the console, meaning you have to actually reach for them. The controls that need more adjustments along the way are closer to your hands.
The channel strip tends to lay out generally “in order” as it applies to the audio signal flow – Gain, then EQ, then the Fader, for example. But this is a very broad overview. There is much more detail to be examined…
Figure 1: Yamaha DM2000.
So how do you learn the signal flow of your particular console? You break out the manual!
It will contain what is typically called a “block diagram.” Now, block diagrams like Figure 1 for a Yamaha DM2000 can be headache-inducing nightmares. So I recommend that you take the time to create your own simplified signal flow. Just follow the lines on the block diagram to determine the signal path.
It’s also recommended that you make it in linear, vertical orientation so that it helps you visualize the flow better. You can use any drawing or paint program to make one.
I’ve created a few signal flows for study. These can be extremely valuable learning tools.
Here is the signal flow of a Mackie 1604 VLZ:
Figure 2. Click to enlarge.
And here is a larger format Yamaha IM8•40:
Figure 3. Click to enlarge.
Finally, here is an APB Dynasonics Spectra-C/56:
Figure 4. Click to enlarge.
Now that’s we’ve taken a look at some different signal flow diagrams, let’s review exactly what the different components you’re likely to see on block diagram are doing.
Gain: A level adjustment designed to optimize each signal coming into the console.
Pad: If you turn the gain all the way to the left and the signal is still too hot, then you should engage the pad, which will reduce the incoming signal by a preset amount (usually 20 dB or so).
HPF: A high-pass filter is a circuit which sharply decreases low frequencies, reducing mike handling noise, stage rumble, and plosives (p-pops).
Polarity: A simple switch which flips the polarity of the input. (Polarity is sometimes incorrectly called “phase”). Useful for eliminating phase-cancellation when using multiple mics on the same source (both the top and bottom of a snare drum, for example).
Insert Loop: A patch point for connecting outboard gear, such as a compressor or effects unit.
Direct Out: An individual channel output after the gain stage, but before EQ or fader involvement. Most often used for feeding multitrack recorders.
Aux Mix: A separate mix of each channel which has its own output, which can be used to feed stage monitors, a recording mix, sends to a reverb unit, or other uses.
Pre/Post: An indication of where the Aux mix splits off from the main signal. If it’s labeled as “Pre” or “PreFade” mix, then its level is completely independent of the channel’s fader. If it’s labeled as a “Post” or “PostFade” mix, then the aux’s level will also be affected by the channel fader as it is adjusted.
PFL: The Pre Fade Listen works as a “solo” button for the engineer’s headphones. You can isolate an individual channel, and hear changes you make with the EQ. Because it is pre-fade, it does not matter where the fader is at the time.
Group/Subgroup: A tool used to help the audio tech during a service or performance. Rather than have to independently mix 32, 40, or even up to 56 channels on a console, you can assign, for example, all of the drums to one fader called a “Subgroup.” The Subgroup does not affect any aux sends, it only affects the main mix. So I can raise or lower the level of all eight drum mics on one fader.
VCAs & VCA Groups: A VCA stands for Voltage Controlled Amplifier and is a common way to “automate” certain things on a mixing console. You can assign multiple channels to a VCA (just like a Group), but the difference is no audio is passed through a VCA.
Instead, the VCA acts like a remote control to channels which are assigned to it. Where it gets really interesting is that channels that are assigned to a VCA Group do not have to share a common audio path at all. This means you can have the entire band on one VCA fader, even if they all are routed to different mixes and Subgroups!
Something to keep in mind with VCAs that you don’t have to worry about with Groups: a VCA provides the exact same function as adjusting a channel’s fader (including any changes to it’s Post Aux mixes). This is different from a Subgroup, as a sub would only affect the house mix.
Bus: This is an electrical term rather than an audio term. Technically, an aux mix, a Subgroup, a master mix, a mono output, a matrix output, etc. are all buses. The only way this term becomes important to an audio tech is in the possibility that you get some “bus distortion,” which may not show up on the meters.
Matrix Mix: A completely different kind of output available only on the larger consoles. It’s sole purpose is to create an alternate mix to be used for recording, for routing a different mix to a different room, or for any other specialized purpose. You will not see a Matrix split on the above audio signal flows.
Why? Because they are not made up of individual channels. A Matrix mix is created solely from mixing the Main Outputs and Subgroup Outputs. So a Matrix Out is created downstream from any individual channel functions.
Jeremy Carter is a veteran of the pro audio industry with extensive experience designing and operating church audio, video, and lighting systems.
Somebody has been feeding misinformation to our pal Roy here.
He wrote in twice and I have taken the liberty of mixing and matching excerpts from both his missives.
Good Sir -
I was hoping you would grace us with your suggestions on miking guitar amps, placement and such.
When you go to concerts, do you ever see the taped “X” on some poser’s 4 x 12 cabinet? Either he or his guitar tech has painstakingly determined that this is the single best-sounding speaker in the box.
They’re absolutely convinced that—after combining his input with the 115 dB conniption fit that the bass player is having, along with the distorto-spazz attack that the other guitarist is experiencing on his brand-spanking-new seven-string Korn/Official Slipknot ax from Guitar Center, plus the wild flailing of the drummer on his chrome double-kick kit with 14 cymbals, topped by the bestial howling of the pseduo-satanic singer—every person in the audience will nod their heads knowingly and say: “I’m sure glad they didn’t mic one of those other twelves!”
Personally, I put the mic on the edge of the speaker, about 2 inches off the cab. When it’s a 4 x 12, I use one of the upper speakers—unlike my colleagues, who seem to really love that wave of 150 Hz that rolls around the stage about a foot off the ground. I encourage players with smaller amps to put them up on a case lid or a chair to get it out of that hell-stream.
As well, is it true that compressors are only triggered by low frequencies?
No, that is an urban myth. Perhaps you were misled by the true fact that it takes more amp power to push subwoofers than high-frequency loudspeakers.
An Old Soundman trick is to knock down 25 and 31 on your graphic EQ at front of house. That way, the amp is not struggling to reproduce those frequencies, which are not important in 99.99 percent of today’s popular music.
Of course, the gang in the Live Audio Board will say that by doing so you’re totally warping your phase response.
Don’t listen to them, Roy!
Also, what are the pros and cons of miking/DI-ing bass amps?
The DI brings the sound to you without any of the ambient noise onstage being included. It can seem kind of dry, though.
The mic brings you the sound of the speakers (the tone of which the player allegedly enjoys.) However, it can also pick up a lot of the crap noise swirling around in general.
Generally I first put up the DI and then add some mic around the edges for fatness.
And finally, the band my bar is hosting this weekend has a stand-up bass, and it sounds awesome!
Perhaps you’re unaware how seldom that is the case with upright basses—or maybe you are, and that’s why you used the exclamation mark.
The Old Soundman
There’s simply no denying the love from The Old Soundman. Check out more from OSM here.
I’ve often joked that my funeral ceremony will include a small monitor wedge next to the casket, with a sign on my chest that reads “I’d like a little more of me in the monitor.”
It’s a way of finding humor in a comment that sound mixers hear all too often.
On a more serious note, watch out! Trying to accommodate “more of me” can quickly provoke an endless string of errors that can turn the strongest mix master into a blithering idiot by the time the night’s over.
This is especially true if you end up mixing both monitors and front of house for shows, which is often the case in my world.
Case in point. Some years back I was asked to mix a 4-piece band, the usual things – vocals, drums, bass, guitar and a couple of monitor mixes up front.
Pretty simple, right? An hour or so later, a few extra not-quite-expected musicians rolled in. Another vocal, acoustic guitar, horns, and oh yeah, another guitar amp.
“We need another wedge mix for the new vocal and acoustic, and we’re going to move this over here and put that mic over there…” And voila, the nightmare began. What used to be mix 1 was now mix 4, and 3 was 2 – or was it 2 was 3?
Up came the guitar amps, accompanied by requests to make them louder in the wedges. No, not this wedge, that wedge over there.
And then, feedback! Every head in the place turned my way. I had lost control of the mix. The band was frustrated, and I was frustrated.
How can we avoid this dilemma? It’s not rocket surgery or brain science, or in other words, not all that hard.
It starts with the artist(s) understanding what you’re trying to accomplish, meaning you should explain it to them at the outset. You don’t have to be a control freak, and allowing each artist to express their wants and needs is imperative. (One at a time, please.)
Assuming the band arrives early enough, give yourself time to set things up during sound check to avoid hands and fingers going every which way – and from everybody – during the show. More of me in this one, and half of him over there, and not so much guitar over on that one (and hold the mayo).
Minor adjustments during a show should be expected, of course. As performers’ ears become accustomed to the SPL on stage, more or less level will be needed.
Nevertheless, the artist needs to be informed (gently if possible) that jacking up the backline amp “so I can hear my tone” only makes the situation more difficult for the whole band (and the sound person, not that this is about me…).
It’s important to have a method to apply during setup. Here are some things that have proven to help.
Use those pre-fade switches so monitor levels aren’t bobbing up and down with the channel faders. Use like wedges with like amplifiers with like power levels (if possible). Determine the locations of the wedges, and be ready explain to the performers the downside of moving a wedge into a microphone pattern.
Keep the master fader down. Dial in the wedges so stage volume is where you want it to be, and then bring up front of house.
Use cardioid microphones (preferably your own). If the performers insist on using theirs, be sure to know the specs. One not-so-great mic on stage, especially in the wrong spot, can really ruin your day. Label all mixes, and keep them organized on stage and at the console.
And remember: someone needing more of something on stage is not an emergency. Think it through, find your controls, make small adjustments.
All of it allows you to relax into your job and focus on delivering the audience the best possible front of house mix.
Greg Stone has worked in live sound since 1976 and is the owner of Hill Country Ears Sound Company (www.hillcountryears.com) in south Texas.
Acoustic guitars are one of the most dynamic and expressive instruments used in modern music. They have a broad frequency range covering almost the entire audible spectrum.
The instrument can serve many different roles in an arrangement as a harmonic, melodic, or percussive element. All of these factors will affect how you approach both the recording and mixing acoustic guitars.
The first thing to do is think about its role in the song/mix:
Is the guitar the main harmonic instrument?
Is it supposed to enhance or thicken other harmonic instruments like electric guitars?
Is it enhancing the melody, playing arpeggios or single notes?
Is it a percussive element more like a shaker just to fill out the rhythm section?
Figure out the goal, then you can begin processing.
Depending on the guitar, microphone, pick, player and role in the arrangement, acoustic guitars always require at least a little equalization to fit in a mix.
Cutting of nonmusical sub frequencies (everything below 100 Hz) is a must. A low shelf may be required to tame the remaining low-end. In the midrange, take out any unflattering frequencies causing “boxiness” or anything that might mask the vocals.
A broad, gentle high shelf boost can enhance articulation and an overall more sparkly sound. The general strategy is to remove what you don’t need, then enhance what you do.
An acoustic guitar performance is often extremely dynamic, which is often a problem. A compressor will even out the differences between loud and soft notes and help keep the level more constant throughout the track. It’s easy to take it too far, so keep the gain reduction to no more than 3-4 dB.
The compressor’s attack control will allow you to clamp down on the beginning of the notes, or when set slower (about 20-50ms) will enhance the transient which helps it stand out more in the mix.
At this point you may want to adjust the sound envelope further though use of a transient shaper. This tool can increase or decrease pick attack, and separately adjust sustain.
Sending your acoustic track(s) into a stereo reverb will add space and distance to the sound. The trick here is to not wash out the sound in a ton of reverb, but just to move the move the mics back a bit into an imaginary room.
Usually, a short room type verb with the highs rolled off is the best bet. Try to make the reverb sound like an office or large bedroom – short decay with and a bit of early reflections. The predelay control will separate your source from the room.
And so, following the above strategy you should now have an acoustic guitar track with a balanced frequency response, even dynamics and sense of space around the instrument, (if that was what the track required).
The track is ready to be put in the mix with the other instruments. Tweak the EQ in context with the other tracks, adjust the reverb send amount, and even EQ the reverb if needed.
Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.
A live echo chamber can be a considerable asset for any recording studio, that is providing that it is a good one.
That’s the problem — how do you construct a good echo chamber? When someone builds a chamber, they hope it will turn out great and pray it won’t turn out absolutely dreadful and good for nothing but storing echo plates.
The truth is there are a number of complex variables which will make each chamber unique. These factors which effect the chamber include the type of wall construction and the selection of materials used on the inside surface.
Probably the most important consideration is the cubic volume and physical proportions of the chamber. This leads to the first question to be asked before a chamber can be built. What space is available?
Most times echo chambers occupy surplus space. The more space that is available to start with, the easier the construction job since the builder won’t have to deal with odd angles or cramped building conditions.
The size of the chamber can generally vary from 1,000 cubic feet to about 2,000 cubic feet of internal volume. 1,500 cubic feet seems to get excellent results within a workable space.
A small chamber won’t get optimum results and the largest chambers are a luxury since they occupy a space large enough to be usable for other purposes.
When determining if enough area is available, it is necessary to remember to allow for figuring the wall thicknesses. There is also a need to consider a wide enough passageway around the structure for a hammer to be swung. If this isn’t planned for putting on the exterior sheeting is going to be a difficult proposition.
Figure 1. Click to enlarge.
It is also suggested that the space be large enough to accommodate a chamber with a minimum interior dimension of 7 feet. A chamber with a side shorter than this will usually give unsatisfactory results.
The Floor Plan
Once it has been decided that there is enough space, the next step is to design a floor plan (Figure 1). When laying out the sides of the room none of the walls should be parallel or even near to parallel. The ceiling should not be parallel to the floor. This is very important if the room is to have maximum random reflections and a smooth decay. It’s at this stage that a bit of math should be introduced (1).
T = Const X M/α S
T = Reverberation time
V = Total volume of the room in cubic units
S = The total combined surface area of all sides in square units
α = The average absorption coefficient Constant:
.049 if measurements are in feet.
.161 if measurements are metric.
The American National Standards Institute (in S-l. 1-1960) defines reverberation time of a chamber as the time it takes for the mean square sound pressure level to decrease to 60 dB after a steady state signal has ceased (2). Generally this level is referenced to 500 Hz although some information relates the level to 1 kHz.
The following equation for figuring decay time was developed by Wallace Clement Sabin (1900). Since his time there have been a number of alternate equations developed but the original equation continues to be the most popular. This is due partly to the simplicity of the computation and the similarity of the resulting data.
Sabin determined that the reverberation time was related to the volume of a room, its surface area and the total amount of absorption.
As the formula would indicate, the time is directly proportional to the volume and inversely proportional to the surface area of the room. With this being the case, the longest echo will be obtained when the required volume is achieved with a minimum of interior surface area.
Trying to achieve more diffusion in the chamber by building accordion type splays will have the effect of cutting down the delay time because the total surface area will have been increased.
Another point to be remembered when figuring out the length of the sides is that none of the dimensions between any two opposing walls should be the same or a multiple or fraction of any other two opposing walls.
Figure 2. Click to enlarge.
There are a number of ratios which serve as guides when figuring acceptable proportions, including the Golden Section; (5^1/2 + 1): 2 :( 5^1/2 - 1).
This relationship was proposed by the Greeks and divides a line in such a manner so that the smaller dimension is to the greater as the greater is to the whole. There are four or five other ratios which have been proposed and accepted to varying degrees, (3) but the one most often used is Sabin’s 2:3:5 relationship. (Figure 2) (3)
The angles used for the intersecting corners should not be severely acute. In practice, the simplest way to arrive at the wall angles is to build the sides so that two of the joining sides meet at right angles.
Their opposing walls are constructed similarly, but without permanently nailing down the floor plate. Once these walls have been framed, the entire unit can be angled inward. For the average chamber moving one end of each of these two walls in by a foot or so should be sufficient.
Rigidity And Isolation
Two things which are important in making a good live echo chamber are maximum interior wall rigidity, and the total isolation of the chamber from its surroundings.
The more rigid the wall is, the less energy dissipated when a sound wave hits it, hence the surface is more reflective. The isolation is important since it’s essential that the rooms broadband ambience be very low.
Figure 3. Click to enlarge.
Echo chamber wall systems which achieve these goals can be constructed from a variety of materials. There are a few chambers that have been built with walls, floors and ceilings of poured concrete. Such a design if properly executed will get very good results, but will be expensive to build, very permanent and extremely heavy. It will be there forever, providing you haven’t underestimated the strength of your sub- floor.
The second most popular approach is concrete block walls. They are easier to build, and a bit easier to tear down, but once again you have the weight problem.
The most popular and cheapest type of construction is a wood frame design made of 2” x 4” and 2” x 6"s. It is the easiest to build with hand tools and more importantly is relatively light compared to concrete and block.
In recording studio construction it is very popular to use 2” x 4” staggered stud construction since the results achieve a better transmission loss than a conventional wall. (Figure 3) A staggered wall uses two rows of studs 8” apart (2” x 4” alternating 16” centers) on 6” top and bottom plates. Every other stud is flush to the opposite edge of the plates. In this way the wall sheeting of the two sides is only connected at the top and bottom plate.
A standard stud wall uses a 2 x 4 plate. The sheet rock covering both sides is connected through each of the common studs. The transmission loss between these two types of construction with insulation is STC 36 dB for the standard wall and STC 49 dB for a staggered wall. (4)
The staggered wall would be the preferable wall design if transmission loss were the only consideration, however, rigidity is a more important factor in the design of the inside shell, hence standard construction is preferred. (Figure 4)
Figure 4. Click to enlarge.
The reason for this is that staggered construction does not allow enough space for cross bracing. The interior shell should have a cross brace splitting the span of every stud, joist, and beam. There should be no unbraced span longer than eight feet.
The preferred layout is to use a staggered exterior wall, and a standard interior one. These two walls should not be coupled in any way and hopefully will not only be decoupled from each other but from the rest of the building.
Floating Walls and Floors
A very important part of the de-coupling of an echo chamber is isolating it from the building it sits in. What sort of isolation is needed will depend on the specific situation and the funds available.
The drawings shown use Celotex for de-coupling. As can be seen, the outer shell is de-coupled from the floor of the building, while the echo chamber has a completely floated floor.
If the location requires extreme de-coupling then it might be necessary to construct the outer shell on its own completely floated floor. It is also likely that something other than than Celotex will be needed.
Machine rubber is a good alternative to Celotex. It seems to work acceptably when used in a quiet environment but it does seem to compress a great deal and might break down with time. A thickness of or more is necessary.
In an extreme isolation situation involving low frequency vibration more severe measures will be necessary. The worst case may need a floated concrete floor on spring isolators, but once again you have considerable weight and expense.
The best material which has been found for most situations are Fiberglas decoupling blocks. As shown, they are two inch cubes, covered with latex (to keep the moisture out) and are specially designed for floating floors. They can be used for both concrete and timber designs.
When they are used with concrete, a sheet of plywood is laid over them and a border runs around the plywood to create a pouring form. Be sure all cracks in the form are sealed so that none of the concrete will seep and re-couple the floating floor to the structural one.
When the blocks are used with wood construction, they should be set under three or four 2” x 6” headers. The timbers form the base for the echo chambers floor joists. Isolators should be placed about a foot or so apart along the entire length of the headers.
The only problem with the block spacers is the gap it leaves between the floating floor and the structural floor. The solution is to fill the space with Fiberglas and run the sheetrock down to close to the floor caulk.
These blocks are available from a number of suppliers. The blocks can also be used to isolate ceilings from walls. For this application the blocks seem to work best if they are split in two.
A chamber built on a concrete ground floor can be further de-coupled by using a concrete saw to cut a slot around the perimeter of the chamber. This helps quite a lot in isolating the rest of the building from the chamber. The newly formed slit can be stuffed with Fiberglas but a hard sealer should be avoided since it will slowly harden and compress and re-couple the slabs.
Insulation, Sheetrocking and Sealing
All the walls should be liberally stuffed with foil backed Fiberglas insulation (R-11). Note: when working with the glass, be sure to always wear gloves, goggles, a mask, and clothes that won’t allow the glass to touch your skin.
Both sides of each wall are covered with sheet rock. Two layers of 2” sheet rock is specified for the interior wall but two layers of 5/8” will work just as well or better. When two layers are used one should overlap the other so that none of the seams coincide.
All seams should be taped and sealed so the room will be airtight. In addition to standard seam Hydroseal which is a black gummy adhesive is highly recommended.
It is normally used as a roofing sealer and will stick to anything. It should be used liberally at every structural intersection of the sheeting or the stud construction. You end up using gallons of this stuff and practically glue the building together.
The purpose of the Hydroseal is to close every crack or seam. The smallest crack should not be ignored. If you have a 1/32” crack along the floor, the actual total area of that hole is considerable.
Figure 5. Click to enlarge.
A final note on Hydroseal. It is suggested that trowels are unnecessary for the application. Wood shims made from construction debris works a lot better. These are good for only about one or two applications. It is almost impossible to clean Hydroseal off anything, including the applicator.
The Chamber Surface and Its Application
After the walls have been built and the last inside layer of sheet rock has been taped and sealed, the chamber is ready to have its reflective surface treatment applied. The reflective walls of the echo chamber can be made from a number of different materials.
The key to how good a particular material is for this application depends on how rigid it becomes after installation and what its absorption coefficient is. Referring back to the earlier formula, the length of the echo is inversely proportional to this coefficient.
The absorption coefficient of any material is defined as “the ratio of sound energy absorbed by a given material to that which arrives at the surface from the source.” (5)
Hence, a porous surface will have a much higher number than a reflective one. This figure will not only be different for each material but vary widely depending on the frequency. Figure 5 shows an absortion coefficient chart of materials versus frequency. As can be seen, plaster has a remarkably low number and is relatively flat. The plaster which Scott uses is Keen Cement (absorption coefficient of .015). He says, “when it is properly applied, it is as smooth as a baby’s bottom.”
Before plastering the surface, the sheet rock has to be prepared so that it will hold the plaster. This is done by nailing on the actual buttonboard.
The actual plastering involves the application of two different layers. The bottom one is a brown coat layer and is put on as a preparation for the top coat of Keen cement. It is strongly suggested that the actual plastering be done by a very good professional.
The mixing of the two cements is nothing more than following the directions on their respective bags. One of the most important things to remember about the plastering process is the amount of time required for proper curing.
As often happens the chamber is complete after the final coat of plaster is applied and there is a desire to use it right away. If the chamber doors are closed prematurely, ventilation will stop as will the drying of the walls. How long it takes for a wall to properly dry changes with the temperature, ventilation, and humidity of the environment, and can vary from a few days to a few weeks.
The longer you can continue to ventilate a new chamber, the better. Scott suggests hooking up a dehumidifier in the room during the drying. He added, “you’ll be surprised how much water comes out of those walls.” This is a crucial part of getting the best end results.
“I have been in chambers that were built years ago that have walls that have never completely cured.” Needless to say such a room has a poor reverberation.
Some chambers sound bad only because the plaster coat was improperly applied. In such cases all that might be necessary is a re- plastering. It is also likely that an additional layer of buttonboard will be necessary so that the new plaster will have a firm base.
Connections And Doors
The chamber should have an IN and OUT opening so that the speaker and microphone lines can be kept separate. The pipe used should be flexible and have an I.D. of 3/4”.
Increase the diameter if there is a need for more than two or three lines per pipe. The smaller it is, however, the less possibility there is for any leakage. Plastic hose works well because it is flexible, has gentle curves and there are no ridges inside for wires to hang-up on.
Make the hole through which the hose is run as small as possible since you want to maintain the integrity of the wall system as much as possible. Once the hose has been installed, any gap between the hose and the wall should be completely caulked with Hydroseal.
A door is obviously needed somewhere in the chamber. Be sure the access to the door and the passageway leading to it is large enough to get a good sized speaker in and out of the chamber.
The two doors used should be of solid construction, not hollow. They are mounted on completely separate frames and jambs to coincide with the de-coupled inner and outer walls. (Figure 6) Machine rubber should be used between the two frames where they almost touch (%” to between the walls.
Figure 6. Click to enlarge.
Rubber should also be used completely around the jambs of the two doors. When they close the rubber should compress and make a tight seal. Since these doors are generally used infrequently, an elaborate closing is unnecessary.
Additional Construction Notes
A light should be installed somewhere inside the room. The switch for it can go anywhere convenient including right on the fixture. Be sure that de-coupling practices are maintained while running the AC conduit and mounting the fixture.
All the walls are built with studs on 16” centers. When laying out the studs, be sure to take into consideration that the drywall is 8’ or 12’ high and 4’ wide and the centers of every third stud needs to line up with the edges of the drywall. #16 nails are used for all end nailing and #8 for toe nailing.
Drywall nails are used on the first layer of drywall, but because of the added thickness #8 nails are used on the second layer. The buttonboard should be nailed on with #8 nails.
Be sure to estimate lumber lengths thoughtfully, to limit the amount of scrap. However what is left should be used for blocking. The greater the waste, the more the chamber (and for that matter any type of construction) will cost.
Speaker(s) and Microphone(s)
Changing the speakers and microphones or altering their placement will change the sound heard in the control room. Deciding what type of speaker sounds best or what microphone should be used becomes a matter of taste. This is equally true as to where they are placed in the room.
Providing there was adequate space to start with and care was taken with the design, construction, and isolation considerations, it is probable that the chamber will end up sounding very good.
With a little bit of luck, it might turn out to be the sort of chamber that gains a reputation for itself as well as the studio that has it.
1 - Sound Recording Practices, J. Borwick; “The Acoustics,” by Alex Burd, Oxford Press, page 27.
2 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 25.
3 - Acoustic Design and Noise Control, M. Rettinger, Chemical Publishing, page 87.
4 - Handbook of Multi-Channel Recording, F. Alton Everest, Tab Publications, page 261.
5 - The Audio Encyclopedia, H. Tremaine, Sams Publications, page 44.
Scott Putnam has been extensively involved in studio construction, having designed and built a number of echo chambers including two for Kaye-Smith in Seattle, and a pair for the Record Plant. As a builder he has collaborated on various projects with a number of acoustic consultants and architects including Jack Edwards, George Augspurger, and Scott’s Father, Bill Putnam.
Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.
Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.
Latin Sounds Fuel Concert Audio For Enrique Iglesias On Tour
Line check is over, the PA is set to go, and right now it’s relatively quiet as Brad Divens savors a moment when he can kick back and provide a little context for his work.
“This is a dynamic show with a lot of energy – and very visual,” he says of Enrique Iglesias’ current tour, which brought him to the front of house helm to preside over a PA provided by Sound Image (Escondido, CA and Nashville). “And I must say it’s a very fun show too. A fun show to mix, and a fun show to attend. No one could come out here and not find themselves enjoying it and having a good time.”
If the 15,000 or so primarily female fans who turn out every night are any indication, Divens’ assessment is more than accurate. “The fans are totally engaged,” he adds. “They know all the words to every song and sing along. Enrique isn’t afraid to get out among them either, up close and personal. We frequently set up a ‘C’ stage when we have the space, and it’s directly behind my mix position.
“When Enrique appears there, everyone goes from looking at the main stage to rushing the C stage. I’m in their direct path, of course. At these moments there’s not much I can do but literally duck and cover.”
Mix engineer Brad Divens (left) with systems engineer Jim Miller pre-show at the Avid VENUE Profile house console where there’s plenty of plug-in capability. (credit: Jesse Adamson)
The show may be packing arenas, but as Divens alluded to earlier, it doesn’t lack intimacy. Capable of rocking out with a power pop tune and then turning on a dime to unleash ballads and quiet acoustic numbers, Iglesias moves like a magician among his fans from all angles, appearing at a second, circular “B” stage found downstage center where the thrust starts to go into the audience, walking directly into the crowd itself, or disappearing entirely only to suddenly materialize on the C stage, where his band is already playing.
“When Enrique pops-up on the C stage, the crowd goes into a frenzy and can easily hit 108 dBA,” Divens says. “I generally sit at 101 A-weighted, peaking at 103, but when he’s on the C stage I pull it back just a little to make things a bit more dynamic. When the crowd realizes we’re no longer at a normal show level, it gets really quiet and you can hear a pin drop as he addresses everyone and introduces the next song.”
The “C Stage” behind front of house. (credit: Jesse Adamson)
Divens is joined on the tour by Sound Image systems engineer Jim Miller and monitor engineer Eddie “El Brujo” Caipo. As a primary PA caregiver, Miller can quickly run down the tour’s cabinet count with speed and alacrity: “In the main hangs there are 14 L-Acoustics K1s per side along with four Kara down fills,” he quickly rattles off. “There are subhangs using a half-dozen K1SBs per side, side hangs comprising 12 Kudo cabinets each, and a pair of rear hangs also utilizing a dozen Kudo boxes apiece.
In The Box
A VENUE Profile from Avid is Divens’ console of choice, loaded with five DSP cards and offering 96 channels.
One side of the main PA, with L-Acoustics K1 mains and subs, as well as Kara and Kudo. (credit: Jesse Adamson)
Using snapshots to incorporate reverb and delay time changes, small fader moves for the band’s DJ, keyboards, and other things he doesn’t have on the VCAs, the show is fairly consistent from night-to-night, giving him the luxury of not having to worry about things changing a lot level-wise.
Everything is played to a click-track, keeping the tempo the same, and insuring that the delays for vocal effects follow suit.
The delays for the PA will change depending upon trim heights and whatnot, but are deftly managed and kept honest by Miller.
Effects come from a Waves Mercury package, and include TruVerb on vocals, Renaissance Reverb on drums, and a Waves doubler, which provides a harmonizer on vocals and acoustic guitar. Echo Farm adds a final complement with a pair of delay options.
“I’m totally in-the-box on this show,” Divens explains. “There is no outboard gear whatsoever. Beyond my Mercury package, I also came loaded with an SSL bundle, Soundtoys, Crane Song, and SPL Transient Designer.”
Within Divens’ sonic blueprint, the SPL Transient Designer plug-in adds attack to toms and snare, while the SSL channels are spread across drums, guitar, and Iglesias’ vocals. Crane Song, in Divens’ estimation, is “good on anything, really.”
Heard within this application on overheads and hi-hat, he says that, “On one level, you can think of Crane Song as an analog modeling plug-in. You get different types of tape saturation, you have odd and even harmonics, and I can even develop a different type of compression curve using one of the settings. I don’t do this kind of thing here on this show with the K1s, but on other PAs I use Crane Song right on the master bus. It provides sparkle and shimmer on the high-end I might otherwise lose in the digital world.”
Monitor engineer Eddie “El Brujo” Caipo at his VENUE Profile. (credit: Jesse Adamson)
Within Soundtoys there’s a plug-in called Decapitator that he uses on Iglesias’ vocals on the verse of one song. Incorporating five compressors modeled to their overdriven limit, the plug-in tears up the signal, adding distortion – hence the name.
“The thing I like best about my Profile is that I can create a little hybrid desk,” Divens explains. “I tailor my inputs exactly to my needs. I’ve been doing this for years now – making use of all the groups, processing channels independently. I always take advantage of things I can do easily in the digital world. Why not? Especially if you’re getting the results you’re looking for. Turn the knob till you like the way it sounds.”
Tight & Controlled
Across the aisle in monitorworld, Caipo also takes charge of things with the aid of a VENUE Profile. Earning his “El Brujo” moniker early on in his career based on an uncanny ability to make even the crappiest house PAs sound good or at least decent enough, Caipo uses a combination of McDSP, Waves, and Avid plug-ins.
Plenty of antennas on hand to help insure solid RF performance. (credit: Jesse Adamson)
“I have McDSP ML4000 on my IEM mixes,” he relates. “It gives me a tighter and more controlled sound, as well as the flexibility to control certain frequencies and keep my mixes clean. For Enrique and drums I’m using Waves SSL Channel and SSL Buss Compressor. In terms of reverbs, I like Avid’s Reverb One for instruments, and ReVibe for both Enrique’s vocal and backing vocals.”
Just like Divens, Caipo basks in the mucho-fun factor that the show offers to those mixing. He uses snapshots for every song, noting that, “It works great because I get to scope it to change what I need for everyone in the band and Enrique. There are tons of cues on this show, but for Enrique they’re done mostly live: I recall a snapshot, then simply proceed to make the rest of the cues live as needed.”
Everyone onstage is on in-ear monitors. For Iglesias, Caipo chose a Sennheiser 2050-XP transmitter and Ultimate Ears UE7s, while the rest of the band uses Shure PSM 1000 systems.
All of this gear routinely finds itself in a world where the lead vocalist is known to run all over the place (yes, even in front of the PA of course), and performances occur on one of three stages spread across great distances in cavernous arenas built in densely-populated urban areas that are already RF hell.
Given this climate, it should come as no surprise that Caipo and his monitor tech Chris “Sharpie” Sharpe occasionally encounter a challenge or two.
“We’re running nearly 50 channels of RF between microphones, IEMs, and the backline” Caipo acknowledges. “Sharpie is instrumental in keeping all that under control, as are Shure’s Wireless Workbench and our GX-8 Distribution Systems from Professional Wireless. Sometimes we may have to work at it a little, but we can always rely on having a good, solid show.”
For Iglesias, Caipo is currently using a Sennheiser 5200-II BK-L transmitter with a Neumann 105 S capsule. “I like the top-end on this mic,” he notes. “It captures his voice extremely well without sacrificing the warmth at the bottom-end. It also enhances the breathiness of his ballads.”
The input list in general is vast, eclectic, and diverse. No one’s AR guy walked in and took down a blanket order covering everything. Starting at the bottom on drums, a Shure BETA 91A can be found at kick. Telefunken M80-SH mics are spread across snare top, while an Audio-Technica AT4051 covers snare bottom.
A Neumann KM 184 stands-in on high-hat, beyerdynamic Opus 88s are on toms, and AT4050s are the choice for drums and percussion overheads. “Big Drum” is what its name implies, and gets a Sennheiser e 604. BETA 98H/C mics from Shure round out the percussion portion of the stand on bongos and cajon.
A perspective of the stage and set during the show. (credit: Steve Jennings)
Direct inputs are also widely employed. There are two guitar players, each of which play three guitars.
“All the guitars are run direct via a Line 6 HD500X,” Caipo says. “We used to use DIs and each would be mono, and to be honest, they didn’t sound too good that way. But all that improved dramatically when we made the switch, and now each guitarist can process all three of his guitars with the same pedal. Using different patches, they’re all sent via a single stereo output to the console. The guitarists can program their own sound as they see fit in terms of reverb, amp emulation, and so forth. As a result, what they hear in their IEMs provides a true reflection of what their guitars actually sound like.”
Supporting Iglesias’ 10th studio album, Sex and Love, the tour wound its way through the American Southwest this winter prior to landing for two dates in Rosemont, IL and Minneapolis, and then moving on to San Juan, Puerto Rico. If Divens, Caipo, and crew, plus the arenas full of screaming fans know a thing or two, fun will indeed continue to be in endless supply at each stop.
Gregory A. DeTogne is a writer and editor who has served the pro audio industry for the past 30 years.
A few weeks ago I came across an old post at another blog that described a trend, lousy church sound. You can read the post here.
I’ll warn you, there are a lot of things going on in that post, and it may take you a few passes through to get a handle on what he’s saying. (I’ve read it five times and I’m still not 100 percent sure…)
My intention is not to attack the author of the post, as I believe he makes some good points. But he makes some statements that I think are worth unpacking here.
Pro Level Requires Professionals
One statement he makes that I’m in general agreement with is this: “They [churches] haven’t yet realized they can’t invest in pro equipment without hiring a pro to run it.”
I’ve been saying this for quite a while now, and I’ve seen it happen at quite a few churches. They start off as a small church in a small room with simple, analog equipment that the volunteers figure out fairly well. As they grow, they build a new building and install a fancy new digital console and no one knows how to use it.
What the church needs is a technical director who can train the volunteers on the new gear and keep it running smoothly. Sadly, most churches discover this too late. There are a couple reasons for this failure.
1) Church leaders don’t realize how complex technology is. Marketers tell church leaders that all they have to do is buy the latest digital console and their problems will go away. This leads them to tell their integrator they want to go digital.
The smart integrator will talk about the need for training for the team, but in the interest of saving money (which is generally needed because the church is trying to build a bigger church than they can actually afford), the training gets cut from the budget.
After the grand opening, when the integrator has gone home, the volunteers stare at the new console like deer in the headlights and things go downhill from there. The reality is, digital audio consoles are complex devices, and they require someone who knows how to run them properly to set them up. Some are easier than others, but all are complicated. Without training and support, the team is set up to fail.
2) It always comes down to the people. I’m always amused that churches are more than willing to pay a healthy salary for a worship leader and will put him or her on the leadership team of the church.
At the same time, churches will often expect volunteers with no training, support or guidance to manage incredibly complex AVL systems. If the church does finally see the need to hire a technical director, it will often bring in a part-time person or will only pay slightly more than minimum wage.
But in fact, the person behind the console is just as important to the overall sound and worship experience as the person on stage. If one is worth a reasonable salary and status, so is the other. Neither will do well without the other.
If you lead a church that is going into a building project that will include a whole new technology system and you don’t have the hiring of a technical director on your radar, get on that. I can pretty much promise you will be disappointed if you don’t.
At this point, you might think I’m down on volunteers. In fact, the author of the original article implied that volunteers cannot possibly ever run a complex digital console. However, I disagree.
Blanket Statements & Volunteers
The author’s next premise is that volunteers will never be able to run a modern sound console. To wit: “Churches are discovering the complexities of modern worship. In other words, you can’t have a new mixing console that resembles the cockpit of the space shuttle and expect a volunteer to (ever) be able to get it to work right.”
I think there are two problems with this statement. First, the only console I can think of that resembles the cockpit of the space shuttle is the Midas XL8. And the few churches that have installed those have professional operators on staff because the consoles themselves cost more than a quarter million dollars.
Second, lumping all digital consoles in with the complexity of an XL8 (or perhaps a Studer Vista X) is really unfair.
But volunteers actually can mix on digital consoles. I know this first-hand as I’ve trained people to do it. Because I know so many technical directors in churches all over the country, I know they also have teams of volunteers who do a great job mixing every weekend. I know of volunteers who mix on various Yamaha, DiGiCo, Allen & Heath, Avid, and even Midas PRO Series consoles.
The one thing that almost all of these churches have in common is that they have a professional technical director on staff who maintains the console and trains the volunteers. As technical production systems become more complex, this is almost mandatory if great results are expected.
When I was tech director at Coast Hills, I had a volunteer who got good enough mixing on the DiGiCo SD8 that most people in the congregation couldn’t tell if it was me or him behind the console. Of course, I did a lot of the setup work that helped him be successful, but from an operating/mixing standpoint he could do a great job.
Great volunteer teams have a great leader. I’ve come across a few churches that have a great all-volunteer tech team, but they seem to be the exception, especially once the church reaches about 1,500 in average weekly attendance. By that time, the building is big enough to have pretty complicated production gear, and most volunteers simply don’t have the time to dedicate to learning every the intricacies of it all.
The teams that do really outstanding work almost always have a staff tech director leading them, a person who takes the time to learn all of the ins and outs, develops processes and systems, and trains the team to be successful. So while I support the idea that it takes a pro to maintain pro-level gear, I reject the notion that it’s impossible for volunteers to ever get it right. They just need the proper support and training.
It’s also important to remember that really, all of the equipment we use for production in modern churches is pro-level gear. For example, my friend Norm Stockton is an accomplished professional bass player, and he plays MTD basses for a living. I’ve had the pleasure many times of mixing when he’s playing. But this doesn’t preclude, say, a math teacher with an MTD bass from putting in the time to become proficient enough to play well enough for a church service.
Likewise, just because the DiGiCo SD8 console is out on tour with more than a few bands, it doesn’t mean that the console is too complex for a high school student to learn to mix on. As with anything, it comes down to time, dedication, natural aptitude and proper training.
Financial Aspects In Context
The author of the original article also said this of a new PA system: “One church spent $125,000!”
Here is where some context can come in handy. The exclamation point indicates to me that he thought that was a large figure. And to be sure, $125K is a lot of money.
However, it may not be excessive. In fact, depending on the room, that may be a good down payment. As church auditoriums get bigger, the amount of PA needed to cover the area well and with sufficient level gets expensive. In fact, spending $300,000 to $500,000 on a system for a 3,000 to 4,000 seat room would not be out of line.
Now, $125,000 might be a lot of money, especially if the room in question is 200 seats. On the other hand, $125,000 is about right for a 700-800 seat room. Unless of course, you’re simply amplifying speech.
I talk with churches nearly every day about technology upgrades and very few have a clue about how much it really costs. After we walk them through the process, they get it, but few do at the beginning.
This problem is compounded by the fact that during a building project, the AVL integrator too often gets left out of the budget process. The architect might put an allowance in there for technology, but again, most times it’s way low. When the integrator is finally brought in, they have to either work within the inadequate budget (more likely) or the church needs to raise more funds (less likely).
Back to our original $125K budget proposition, the author talks about how bad such a system sounded when he heard it. I wonder if he considered that perhaps it’s because the church spent only $125K, instead of the $200,000-plus it may have really needed?
While I agree that spending $125K on a PA only to have it sound “10 times worse than before” would be disappointing, perhaps the fault lies with the church that in an effort to “save money” didn’t spend enough. I’ve seen more than one system that wasn’t done well due to lack of funds, and we usually take it out to put in a good one. As the saying goes, churches that can’t afford to do it right the first time will almost always find the money to do it again.
Good People Should Be Paid Well
One more statement by the author that I’d like to address: “I know of one megachurch that just hired an excellent soundman away from another megachurch – they’re paying him $60,000 a year and he was making $30,000.”
I can’t tell if he thinks the $60k salary is excessive, but I’d say it sounds about right for an “excellent sound person,” depending on what part of the country you’re in. Out here in SoCal, that would be a good opening offer. And for the person making $30,000, I would say his previous church was very likely way underpaying him—which is probably why he left.
Churches that pay a senior pastor $150,000-plus, a worship leader $90,000-plus, and a lead tech person $30,000 will likely be disappointed with the long-term results. Especially if they skimped on the system.
What’s really required here is to look at the big picture. Whenever we throw out random numbers, we can incite shock and awe, but without knowing the context, it’s hard to know what is really going on.
This is another reason why it’s so important to have a relationship with a great integration company to help guide the process. Good integrators will help right-size the system for the room, budget and team. When they are brought in early and allowed to do their job well, everyone will be happy with the results. Skip this at your own peril.
Well, that was fun, wasn’t it! Now that we know some of the reasons for lousy church sound, next week we’ll return to our regularly scheduled programming on how to make it better.
Mike Sessler now works with Visioneering, where he helps churches improve their AVL systems, and encourages and trains the technical artists that run them. He has been involved in live production for over 25 years and is the author of the blog Church Tech Arts.
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