Monday, August 26, 2013
Church Sound: How Many Drum Microphones Do You Really Need?
Alternative approaches where sometimes less is more (includes video)
Drum mikiing is an art form you might be missing. How many drum microphones do you really need? Do you have that many mixer channels to spare? Are you feeding your drummer the right mix?
Four microphones might be all you need.
My current stage setup includes an acoustic drum kit with eight microphones. Eight? Mmmm, that sounds about right. The larger the room, the more control you want over the drums.
What I mean by that statement is that in small rooms, you can get a lot of stage volume out of your drums without mics. I occasionally work in a small venue with ZERO drum mics…and it works because the room is small enough.
The eight microphones are easy to fill up when you have mics for the kick, snare, individual toms, cymbals, high-hat, and then the overheads. This gives great control of the whole drum kick and allows me to emphasize different parts of the drum kit depending on the needs of the song.
But what about when you don’t have enough channels, or enough mics, or wonder if you really even need that many in your room?
Welcome Glyn Johns
Glyn Johns is a recording artist who has worked with many of the big name bands from the classic rock era. He is even attributed to giving the Eagles their distinct sound.
One of the innovations in drum miking came from him and is called the Glyn Johns Technique.
This technique uses only FOUR microphones for capturing a full drum kit. I’ve even it seen using three, but for today, we’ll use four.
The first dynamic microphone is placed 6-to-12 inches from the resonance head of the kick drum. Gotta mic the kick!
The next two mics are used as overhead microphones. The first of these should be about 4 feet over the snare and it should capture the full sound of the drum kit.
The second overhead isn’t really on overhead at all but the idea is the same. Place it to the right of the floor tom, about 6 inches above the floor tom head and pointed directly at the snare drum. Consider it a side-fill mic that gives you a different kit sound. Use condenser mics for these two “overheads.”
Finally, mike the snare a few inches above the head.
You now have two overhead microphones that capture the full drum kit and you have control over the kick and snare. When mixing, bring in the overheads, mix those for the best blend, and then add in the kick and snare for filling out the mix as needed.
How does it sound? I found this unique video of someone playing with the Glyn Johns Technique to a Paramore song, minus the drum tracks. Enjoy.
What about the drummer? Pipe out an overhead to his monitor. Then, as needed, bring in the kick and the snare.
The key to making this work is in having the overheads capture the full kit sound. You don’t want them to be too heavy on one kit piece.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.
In The Studio: Pros And Cons Of M-S Recording
A very good way to get stereo imaging in certain situations
My pal and reader Gian Nicola asked about the pros and cons of M-S stereo recording, so I thought I’d respond with a passage from the upcoming 3rd edition of The Recording Engineer’s Handbook (due to be released in October).
M-S stands for Mid-Side and consists again of two microphones; a directional mic (an omni can be substituted as well) pointed towards the sound source and a figure 8 mic pointed towards the sides. The mics are positioned so their capsules are as close to touching as possible (see the graphic above/ left).
M-S is great for stereo imaging, especially when most of the sound is coming from the center of the ensemble. Because of this, it’s less effective on large groups, favoring the middle voices that the mics are closest to.
M-S doesn’t have many phase problems in stereo, and has excellent mono compatibility which can make it the best way to record room and ambience under the right circumstances. In many cases it can sound more natural than a spaced pair, which is covered later in the chapter.
If the source is extra large, sometimes using M-S alone will require too much distance away from the ensemble to get the whole section or choir into perspective, so multiple mic locations must be used.
If a narrower pickup pattern is required to attenuate the hall sound, then a directional mic such as a cardioid, or even a hypercardioid, will work for the “M” mic. Just be aware that you may be sacrificing low end response as a result.
For best placement, walk around the room and listen to where the instrument or sound source sounds best. Note the balance of instrument to room, and the stereo image of the room as well. Once you have found a location, set up the directional mic where the middle of your head was.
Listening to either of these mics alone may sound OK, or may even sound horribly bad. That’s because in order to make this system work, the mic’s output signals need an additional decoding step to reproduce a faithful stereo image.
The directional creates a “positive” voltage from any signal it captures, and the bi-directional mic creates a positive voltage from anything coming from the left, and a negative voltage from anything coming from the right. As a result, you need to decode the two signals to create the proper stereo effect.
While you can buy an M-S decoder, you can easily emulate one with 3 channels on your console or DAW. On one channel, bring up the cardioid (M) forward-facing mic. Copy the figure 8 mic (S) to two additional channels in your DAW.
Pan both channels to one side (like hard left), then flip the phase of the second ‘S’ channel and bring up the level until the two channels cancel 100 percent.
Now pan the first ‘S’ channel hard left, the second “S” channel hard right, balance the cardioid (M) channel with your pair of “S” channels and you have your M-S decode matrix.
A nice additional feature of this method is that you’re able to vary the amount of room sound (or change the “focus”) by varying the level of the bi-directional “S” mic.”
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 blogs. Get the 3rd edition of The Recording Engineer’s Handbook here.
Thursday, August 22, 2013
In The Studio: Techniques To Get Great Sounding Vocals
Recording approaches for "the most important thing"
Someone once said: “A good music producer worries about the most important things” and a strong argument can be made that the most important things in pop music production are the vocals.
The singer is charged with artistically conveying the song’s lyric over a music track production that (hopefully) propels the song’s meaning and emotion across to the listener in an accessible and entertaining way.
Obviously the singer/artist/song are one of the main reasons engineers, producers, musicians and the studio personnel have jobs. They exist to facilitate the production of a song’s music and vocal performances. It is the focus of this article to deconstruct the process vocal recording in the studio.
To better understand the process of recording vocals and for illustrative and tutorial purposes, I’ve divided it into activities in two spaces: what goes on in the studio area and what’s required in the control room.
In The Studio
Recording studios come in all shapes, sizes and décors. There are only a few basic requirements conducive to getting a good vocal performance.
It does not take a special or a big room to record vocals but the studio’s size, acoustic properties and construction are just as important as a recording space as they are for acoustically louder instruments like drum kits, brass or string sections.
In the case of using a larger tracking room for overdubbing pop music vocals, engineers and producers prefer to “stop down” its size in order to record a dry vocal sound with little of the room’s ambient qualities included.
This, of course, allows them the freedom of adding whatever ambient effects they feel appropriate later in the final mix.
Gobos can help “stop down” the size of a studio. (click to enlarge)
Tall baffles or gobos are placed around the singer and mic to stop most of the room’s sound from being recorded along with the singer. If you are working in a large room with a pleasing decay time, there are plenty of reasons to record vocals sans any gobos.
The difference in ambience could work well to layer multiple tracks sung by the same person such as for double tracking or harmony stacking or for recording a singing group or choir.
You could capture a unique ambience possible only in that room instead of adding a simulation electronically from a commonly available digital reverb. I’m suggesting a high quality room like EMI’s Abbey Road Studio 2 — the Beatles’ playground!
If you are working in a small room or vocal booth, then dry is what you’ll get but make sure the dryness is not more of a tonality — an actual comb filter EQ effect caused by close, highly-reflective parallel walls, floors and ceilings.
Again, the use of a few gobos with soft, non-reflective surfaces will help kill those reflections.
You might try a Se Electronics microphone Reflexion filter — it uses a small screen of highly absorbent materials to surrounds the mic itself and prevents sound reflections entering the back and sides of the mic.
Especially good for acoustically bad sounding spaces like bathrooms, closets and hallways, a microphone filter “separates” the mic’s pick-up of the singer completely from the coloration of the surrounding space.
If you’re working in an “all in one space” studio, the room sound issues expand. You’ll have to eliminate noises from you computer’s fan(s), poor acoustics at the mic’s position, and external noises from A/C equipment or the streets outside, etc.
VocalBooth.com makes portable vocal booths — these look like old-time “phone booths” with a window and door and come in different sizes depending on how big the vocal singing party is going to be.
My own Tones 4 $ Studios is a single space setup used mostly for mixing, and for recording I use a product by RealTraps called a portable vocal booth.
A portable vocal booth. (click to enlarge)
It’s a pair of 2- X 2-foot absorbent panels that mount to a mic stand and forms a right-angle corner behind the mic and singer. This configuration does much more than a mic filter.
The portable vocal booth removes the influence of the sound of the adjacent walls, provides isolation from the rest of the room’s sounds — be it other musicians or the racket coming from my Pro Tools rig (computer, drives, power amp fans) as well as reduces external street noises.
Singers appreciate it for the sound and also because they can pin the lyric sheets to the panels directly in front of them.
The singer’s “station” consists of a boom mic stand to hang the mic over and above the music stand, microphone, pop filter (if required), music stand with light, headphones and control box, stool, small table to hold tea, coffee or water etc.
Or, in the case of a female demo singer I once recorded (whose name I can’t remember), a plate of strips of raw meat.
A metal music stand must be covered with soft cloth material to prevent sound reflection and checked to see if it vibrates sympathetically to the singer’s voice. Make sure it does not.
The entire station should be placed on a rug to mute any foot tapping and stop sound reflections coming from the floor. All mic, headphone and power supply cables should be dressed away so nobody trips and pulls over a multi-thousand vintage condenser mic over.
I try to locate the station under dimmable studio lighting for this reason and also for reading lyrics and for seeing the singer’s hand gestures and signals from in the control room — even if the studio is darkened.
The “look” of this setup—rug style, gobo colors etc. is up to the producer and artist’s tastes and preferences.
It should look warm and inviting to the artist and help set up the vibe of the session. I think this all helps in subtle ways—it is more special treatment for the artist and transforms the space.
However, for some artists and producers, none of this matters, especially if scheduling, cost, budget and availability impinges on the optimum choice for a studio. At those places, you may have to do all this “remodeling and redecorating” yourself.
Two looks at a singer’s station. (click to enlarge)
With respect to the visual sightline to the control room, most of the time eye contact is wanted — remember, the producer is acting as the listening audience and the artist will look for reassurance or emotional “feedback” from him/her, the engineer and anybody else in the session in the form of facial gesturing or even body language.
I’ve worked in studios that used closed-circuit TV to see the artist singing who could not see us back in the booth. I can’t prove any connection, but I bet the quality and emotion of the performance will be different—but depending on the singer, maybe better or maybe worst.
There are three microphones choices for vocal recording: condenser, dynamic and ribbon. The differences are vast and the right choice can make or break the vocal sound and performance.
Most of the time, a large diaphragm condenser mic is used for vocals for its ability to capture the loudest to the softest of sound and nuance.
The large diaphragm offers a big surface area to pickup low frequencies and most modern condensers have huge dynamic range specs meaning it will be difficult to distort them with close, loud singing.
While old vintage condensers sound wonderful, I find them (depending on their condition and upkeep) a little more finicky, temperamental and a little unreliable compared to some of the newer mics coming from Germany.
So in the world of condenser mics there are a lot of great choices. I like the whole line of Brauner mics, Neumann (both new and vintage models), an AKG C12, Sony’s out of print C-800G and vintage C-37, Manley Reference, David Bock, and Dave Pearlman mics, and John Peluso remakes of classic vintage mics such as his 2247 SE or P12 models.
Dynamics in the studio work great for loud and brute force singers. There is nothing like the urgency of the sound brought on by a good dynamic mic. Some singers must physically hold the mic to “produce” their vocal sound because they are used to working it during live shows.
I’ve tried to let them sing their vocal that way if there is no handling noise and minimal P-popping. I’ve sometimes given the singer a handheld dynamic mic while standing in front of stand-mounted condenser mic. I would record both mics to two tracks and later go between them in the mix.
The list of good dynamics is long and here are a few worth using for studio vocals. I like Shure SM7A or B, Electro-Voice RE20 or RE27N/D, and Heil Sound PR 40, PR 22, PR 20 or PR 20 UT.
Ribbon mics have always been favorite vocal mics, dating back to the 1930s. Today the modern versions are better than ever with wide-open sound, more gain and rugged ribbons less prone to damage from close vocals like the old classic models.
In general, ribbons are great for harsh or bright sounding voices that need some mellowing. I like the AEA R84, Shure KSM353, and the Coles 4038 with its “brontosaurus bottom end.”
The mounting, positioning, distance from the singer, and even the angle of the mic all weigh heavily on the finished vocal sound.
I like to use a heavy floor stand and boom. I try to position the boom’s counter-weight opposite the singer — out of the way. The counter-weight should be padded in case someone does not sufficiently tighten the stand’s height and it slips and comes crashing down.
I learned a lesson years ago when, in a hurry, I (or the other assistant) didn’t fully tighten a mic boom overhead of session drummer Earl Palmer’s kit.
Halfway through the session it came down and the counterweight hit him in the head. The producer nearly called the session while ol’ Earl stopped bleeding. (Sorry again Earl!) I prefer to use a good shock mount microphone holder and hang it so the mic’s capsule end is about eye level and aimed at the singer’s mouth.
Check with your singer(s), who will have a definite preference as to the way they like to project sound towards a studio mic. It is better to angle the mic down rather than allow the singer to sing straight into the mic’s capsule.
Microphone angled down toward the singer (above), and directed straight at the singer. (click to enlarge)
Windscreens — Pop Filters
With the mic angled and above the source, you may not need to use a pop filter, but your singer must keep from pointing upwards at the mic; this will defeat the whole purpose.
So if the singer can sing straight ahead just below the bottom of the mic without tilting up, then no windscreen is needed.
If the singer cannot keep straight ahead or wants to sing directly into the mic, you’ll have to use a screen. There are several great models out there and for the perfect popping storm — singers with an extreme popping problem try Pete’s Place Blast Filter.
Middle Atlantic has a more conventional two-stage nylon mesh type.
The Stedman filter is also a good choice because, like the Blast Filter, it’s metal and washable.
Pop filters change the sound slightly. There is a greater or lesser loss of super high frequencies depending on the particular filter. But there is another method to reduce plosives — an ordinary #2 pencil.
Although not as effective for big pops, this trick will kill most small pops.
Simply strap the pencil vertically in line with the mic body’s length (assuming you are hanging the mic vertically) using rubber bands (don’t use tape) so that the pencil bisects the face of the capsule.
The pencil will disturb the puff of air from a P pop and divert the impact from the capsule.
The pencil-on-the-mic “trick”. (click to enlarge)
Headphones for you singer are very important. I have several different models I bring if the studio’s selection sucks.
All three of these models are closed-back, circumaural earphones that attenuate ambient noise and keep the cue mix from leaking out.
I like Shure SRH840 phones for their fat and loud sound. Ultrasone HFI-680 are bright phones your artist may prefer, and finally, AKG K271 phones or some variant offer the most unvarnished truth of the sound.
Try to get your singer to keep both ears covered with the phone cushions to prevent spill. The phones should fit well and make sure a powerful amp drives them.
Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics.
Wednesday, August 21, 2013
Maximizing The Mix Of Live Recordings
Ten ways to make your live mix shine in post production
Anyone who’s ever mixed a live recording knows that it’s a lot different than mixing tracks recorded in the studio.
How? Dealing with audience tracks for one, and leakage for another. With that in mind, here are a series of steps you can use to maximize that mix.
1. Set Mixing Priorities
Assuming that you’re not going to replace instruments that are played badly or don’t sound that great, your mix may be determined not by the instruments themselves but by how they’re played.
The best strategy is to emphasize the strongest performances and keep the weakest ones low in the mix.
As an example, while it’s normal for most engineers to build their mix around the drums (although some mixers start with the bass and others with the lead vocal), that process might not work well if the drums are the weakest played part.
Emphasizing a stronger part will take the focus off some shaky playing, and even though the mix might not have the punch you’d like, the final product will be better for it.
2. Place The Audience
A big question for someone new to mixing a live show is how to balance the audience against the instruments. Generally speaking, the level of the audience is determined by the sound of the room and by the needs of the artist.
If the sound of the audience tracks is pretty good, they can be placed higher in the mix to add some “glue” where it’s appropriate.
Likewise, if the artist wants the energy of the audience as a prominent feature the audience level can also be raised.
If the sound of the audience and room sounds bad, the tracks have little room sound and just audience, or the artist or producer prefers a drier sound, then the audience tracks are brought back in the mix.
Regardless of where the audience tracks end up in the final product, don’t wait until the end to bring them into the mix. Check them against every instrument to see what the interaction is.
3. Embrace The Leakage
For those who normally mix studio tracks, instrument and monitor leakage is a departure from what you’re used to. The tracks are never as clean as in the studio, unless all the instruments are recorded direct (even then, the vocals will have some sort of leakage).
The real trick is to use that leakage to your advantage to fill out the sound. Don’t try to get a clean mix because it won’t likely happen. Best to get a general balance first and gently message levels with the leakage in mind.
4. Gates Can Help
One way to clean up the leakage a bit is to use a gate on any instrument with excessive leakage, but use it judiciously. If a gate is set to decrease the level of an instrument to infinity (mute it), it may sound unnatural and probably affect the sound of other instruments leaking into the one you gated.
Best to set the gate to gently decrease the level by 3 or 6 dB and see how that affects the sounds of other instruments.
5. Pan To The Picture
If the show is being mixed to picture, then you’re pretty much stuck with panning as you see it. That being said, it almost always helps with the phase if everything is panned the way the band was set up on stage.
6. Be Careful Of Phase
Speaking of phase, you can’t always be sure of the polarity of the microphone cables and signal chain, so it’s best to check every track as you add it to the mix.
Check both positions of the phase switch and select the one with the most low end.
This is especially important with audience mics, which can be subject to comb effects because of their distance from the stage and each other.
7. Be Careful Of Adding Effects
Because you already have a lot of leakage on different instruments, adding effects to one instrument or vocal can sometimes spill over onto other instruments or vocals, which might begin to wash the mix out.
Also, depending upon how large the venue is, there may be a lot of built-in ambience that you can use instead of adding something artificial (unless it’s a unique effect required by the song).
8. Be Careful With EQ
Just like with effects, any EQ added to one instrument or vocal that has leakage on the track can affect the sound of other instruments as well.
This is a good reason to build your mix first, then add any individual EQ to see how it affects everything else in the total mix.
9. Use The High Pass Filter
Just like in studio recording (maybe even more so), the high pass filter is your friend. The HPF can clean up the sound considerably on every instrument and vocal, even kick and bass.
Especially in a live recording, there’s a lot of low frequency information that gets recorded (like leakage from the subs) that isn’t useful and the HPF effectively eliminates it, which cleans the mix up like magic.
10. Don’t Over Compress
Once again, just like with effects, any compression added to one instrument or vocal that has leakage may affect the sound of another instrument as well.
Compression on individual tracks is almost always needed to control the dynamics of a live recording, but you’re better off to start with a gentle 2 or 3 dB of compression, then increase if needed until it begins to negatively affect the sound of another instrument.
It’s also best to build your mix first, then add compression. If you need more compression than what you can comfortably add to individual tracks, you can always squash the stereo bus.
Follow these 10 steps, and you’ll find that you’ll be able to produce a better sounding product – and in less time than you think. Happy mixing!
Bobby Owsinski is a veteran audio professional and the author of several books about live and recorded sound.
Tuesday, August 20, 2013
Eight Ways Ways To Boost A/V Project Profitability
Driving margin back into your projects is a game changer
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 CI’s 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.
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.
Go to Commercial Integrator for more content on A/V, installed and commercial systems.
In The Studio: How To Succeed As A Freelance Audio Engineer
It’s a slow climb, but you can turn a pursuit into a career
Hi folks. This article is going to be a bit different. It’s not about compression, vocals, reverb, or something of a musical nature. It’s about creating a career in music.
While it may not be the most popular article I’ve ever written, I think it will be pretty valuable.
There are two types of people in the world: people who want to work in music, and people who work in music.
People often confuse the idea of getting paid to work in music with working in music. Music is a tough field, because in your first five to ten years in the business you don’t really make enough money to support yourself. Or in my case, you actually end up going into a bit of debt. Possibly a lot. Depends how good you are at kitchen work.
The point is, working in music (really in the arts in general) requires more motivation than your average job, because your first pay check doesn’t exist and there’s no promise that it ever will. However, I promise you it will come.
There are two basic principals to getting a career in a music going: The first is to get really f-ing good. The second is to get known. And the two go hand in hand.
Back In The Day
At the very beginning of my career there was a lit path. You worked as an intern at a studio, and worked your way up to assistant engineer or assistant producer, and eventually became a b-lister, and then eventually an a-lister with your own clients and assistants. Here, the process allowed you to become good and become known in the same movement. I’m lucky in the sense that I caught the tail end of this. Unfortunately those lights have faded.
As these lights went out, and that path became enshrouded, I realized I needed a new way. I left my job at the studio and began to create my own world. I stopped expecting clients to come to me — rather, I had to figure a way to reach out to potential clients. I also had to start relying on myself alone for support and experience.
First, the success must be an inner success. If you aren’t succeeding in your own sense of self, you can’t show people how truly great you are. This isn’t something that can be faked. So you set your own bar high.
In terms of skill level, I consider the people who charge five times what I charge to be my competition. ‘Good for what I charge’ is never the goal — really, ‘great period’ is the goal. And on every record.
The second issue in this is being in competition with oneself. Making excuses is the equivalent of giving up. So what if the vocal is a cheaply tracked mp3 and the music is printed as a 2-track. The end listener wants to listen and enjoy it — so make it work.
It’s important to walk in with this mindset, because your first clients will be your toughest.
Generally speaking, the guys with the smallest budgets are also the guys with the least experience — and those are going to be the first clients you start picking up. This is your boot camp. You spend the extra time it takes, even with very little (or no) pay to get great results from inexperienced artists.
Not only does this get you prepared for whatever the more demanding artists will throw at you when you start charging a lot more, but it also accelerates your career. That initial set of artists are the ones who will spread your name around. And in the absolute best case scenario, the music you worked on gets spread around as well.
Eventually, more experienced and demanding artists start seeking you out and they force you to elevate your game — which in turn gets you even more experienced artists. All the while, the artists you’ve been working with are elevating themselves.
What’s important here is that at no point do you stop going the extra mile to put forth the absolute best results — even if it means working two days on a mix when you only get paid for one.
Second, the success must be an outer success. When you’re really good — let people know it. Put together a demo reel, a website, business cards. Go out to shows. Genuine confidence is very powerful.
But be wary, because false confidence is equally as powerful and works against you. Everyone is on tight budgets, tight schedules, and very nervous about who handles their art. It’s extremely personal. People literally invest their lives into their music.
Artists live in a world full of insecurities, so real confidence is extremely important for them. The important thing is to deliver the goods when all is said and done! Then let it be known. Make sure you get your credits and remind the artists you work with to pass your name along.
Lastly, success relies on humility.
Reading up on what others are doing, listening carefully to artists’ needs and criticisms, listening analytically to songs you like and songs you hate, nodding to other people in the field who have their success are all very important.
The Artist Is Right
Mostly you have to keep in mind that the artist is right. If you ask an artist “do you want your music to sound bad” the answer will assuredly be “no.” But that doesn’t mean that artists won’t ask you to do crazy stuff: squash, distort, make something sound strangely disproportionate, dub in a chicken squawk, whatever.
The trick is to embrace that, and find a way to make it work in a sonically pleasing way. There’s a thin line between a bad idea and foreword thinking. So be humble.
Two common scenarios I face all the time are: trying to make an artist’s/band’s record sound like it was tracked in a million dollar studio even though they tracked at home in the bedroom; and, doing something with an effect or making an arrangement change that I really like that the artist hates.
The answer here is: get over it. That’s why it’s a job. If someone isn’t asking you to do the impossible then you’re not going to break new ground. If someone doesn’t hate a choice you’ve made then you’re not taking enough chances.
It’s a slow climb, but with experience, exposure, and humility you can turn a pursuit into a career.
Matthew Weiss engineers from his private facility in Philadelphia, PA. A list of clients and credits are available at Weiss-Sound.com. He’s also the author of the Mixing Rap Vocals tutorials, available here.
Be sure to visit The Pro Audio Files for more great recording content. To comment or ask questions about this article go here.
Sunday, August 18, 2013
Hands-On Versus By The Book: The Path To Learning How To Mix
Mixing is a special job and seems to require a certain kind of personality
How do we teach aspiring audio professionals to mix? How important is teaching? Can some people even be taught?
I’ve seen debates in the ProSoundWeb LAB forums as to whether mixing sound can/can’t be taught.
There are a couple of interesting books I’ve read that relate to this idea. The first one is “This Is Your Brain On Music” by Daniel Levitin.
Levitin addresses the age-old question of talent versus work effort, and essentially concludes that the real deciding factor of becoming an “expert” in any subject is how much time was put in. And the threshold of 10,000 hours seems to be the right number for someone to be a well-regarded expert in any field.
Levitin even brings up Mozart as an example, since the Austrian composer is often quoted as being someone who had “talent” and was a “genius.”
Interestingly, Levitin claims that it was likely Mozart had spent 10,000 hours learning music before his compositions were what we would still want to hear today. Of course by then, he was only 10 years old!
Additional research cited by the author followed violin students in the college environment. It was shown that the difference between the levels of the students towards the end of their education was not due to “talent” but due to how much they practiced.
So what does this mean?
To me, it’s still supportable that Mozart was a genius - in other words, something else was going on beyond just thousands of hours of toiling away learning and practicing musical composition.
If not, then why aren’t all the other composers who spent 10,000 hours considered to be just as good as Mozart? And maybe there are examples of genius in our profession - maybe a Phil Ramone or an AI Schmitt, perhaps?
But here’s where I think “This Is Your Brain On Music” gets interesting. In order to learn these new things, there has to be a willingness, or desire, to learn. The subject matter has to seem important to the pupil. And I distinctly remember the difference between subjects that I was interested in and those where I was not.
Math comes to mind. When math was all abstract ideas or things like, “This train left the station at 3 pm, while this train blah blah blah,” then it was as if the concept did not want to enter my brain.
Or, rather, my brain didn’t want the concept to enter.
But when math was applied to sound then I was interested. I learned how algebra applied to filter and power supply designs. I learned how to calculate dB relationships based on changes in voltage or power. Not complex math, mind you, but nevertheless in this scenario, the math was easy. And it was because I wanted to learn it.
Levitin mentions in his book that he often advised that his students would have to “care about the material if you want to do well on the test.” True enough - even though I struggled to get a B on a high school algebra test, I aced just about every one of my audio electronics theory tests, all with the same math.
THE CASE FOR HANDS ON
The second interesting book I’ve read on this topic is “Participative Learning” by Fredric Margolis and Bonnie Swan.
Sure, I have no doubt that we all know that hands-on learning, or OJT (on job training) as we used to call it in the Air Force, can be an excellent way to learn.
One of the main points that Margolis and Swan make is that adult learning is distinct from how children learn.
Here’s what they say on that subject: “The best training is training that gets into peoples’ heads through their thinking, not what gets into their heads through their eyes and ears.”
This somewhat flies in the face of the long-held belief that simply by showing someone how to do something that they should be able to do it. Instead, they have to understand why they are doing it before the learning really sinks in.
Hands-on training can, and certainly does, provide this type of understanding, but not always. Just saying, “Do this because that’s how we do things here,” is simply not enough.
What’s perhaps more important is that the student, intern, junior engineer or whatever, knows why, and the background to go with it.
This reminds me of talking with a four-year-old.
Kid: “Why is the sky blue?”
Me: “Because of the nitrogen in the atmosphere.”
Kid: “Why?” and so on.
People really do want to know why we use condenser mics for overheads and an SM57 on the guitar amp.
Once they know those things (and thousands of other things like that) they can choose to use those same tools or they may find that different tools work better for them.
Related to this, Margolis and Swan also point out that, “Adults have a deep need to be self-directing.” What this means is that someone in the role of the trainer or mentor must engage them in the process of inquiry, analysis and decision-making rather than simply saying, “This is the way to do it.”
This makes sense, and if you think back to your own roles as a trainer and trainee, you can probably think of the times when you experienced this process both ways.
SO, WHEN DO THOSE TWO TRAINS MEET?
Despite the obvious advantages of hands-on and participative learning, there is a lot to be said for book smarts. Certain ideas simply aren’t obvious or even are perhaps counterintuitive until the theory behind them is understood.
How about Ohm’s Law? How about the relationship between decibels and power? Many of the concepts that we deal with must be learned on a deep level before they can be tested and put into practice.
But I’ve also found that a long-term “give and take” process seems to be the most beneficial overall.
It’s simply this:
A) Learn some new idea and give it some thought. Possibly imagine how it might be put into practice.
B) Find an opportunity to put new ideals into practice, or stumble across a problem and realize that a new idea you’ve learned might be helpful here.
C) Take your experiences in the real world back to the books and search for new ideas to give you an edge.
Rinse and repeat.
To a large extent, I think many of us do this automatically.
But I have always encouraged audio people to “hit the books” or perhaps attend a seminar or workshop, or even sign up for a class.
More knowledge can always help you in general, and help in providing greater value to your clients. Instead of knowing “just enough to be dangerous,” the goal is to know enough to be dangerous to your competitors.
And who knows? The state of the art of our industry might just get pushed forward in the process.
Let’s get back to mixing—I’m not convinced that anyone can actually be taught the art of mixing, meaning the fine nuances that go together with a great soundscape out in the audience.
Nevertheless, the nuts and bolts can be taught. Signal routing is the most basic of tools in this craft. Next comes gain structure. But despite this being such a basic thing, I’ve found that it’s not as widely understood and employed as we would all like to believe.
From there comes the more subtle aspects of individual channel EQ and then overall system EQ. All these things are really basics and for the most part can be taught and learned.
Like the art of musical arrangement in the first place, the fine art of mixing appears to be a talent that not all of us possess. Sure, most of us learn to pull together a decent mix that gets the job done. But what about those shows where you hear a great artist mixed by someone who obviously knows what they’re doing, and you go “wow!”
OK, maybe that doesn’t happen too often, and indeed, many of us probably hear more shows that sound terrible than any other. But believe me—awesome sounding shows do exist.
So how do we teach these skills? How do we learn them?
In my view, the first step is the understanding of “what is music” for your particular style. The second part is to think about how you want to present this music to your audience, i.e., imaging, SPL, coverage, etc.
Then think about the technical requirements to make that happen. Getting familiar with the actual music is paramount - understanding the musical arrangements, who will take solos in what songs, how the vocals are supposed to fit into the mix, etc.
This, in turn, requires an understanding along with the musicians themselves. They should have some input, but it’s up to you to actually mix.
And finally, as we’re doing our job, a fine mixture of humility and thick skin will come into play. We must be open to taking suggestions and putting them into practice.
Mixing is a special job and seems to require a certain kind of personality. Someone who cares about what they’re doing and wants to learn more, extend their skills and their craft, and take the heat when things get challenging.
So it’s time to get out the books, sign up for the seminars, find a mentor and go out there and make things happen. And if you’re an experienced road dog, consider taking some of the “young whippersnappers” under your wing and help them understand why things are the way they are.
They may not thank you now, but they’ll certainly come to know that you were right as they achieve success.
Karl Winkler is director of business development at Lectrosonics and has worked in professional audio for more than 15 years.
Saturday, August 17, 2013
What’s The Measurement? Understanding And Properly Using RTA & FFT
The difference between sound that’s “OK" as opposed to spectacular
What’s the best way to equalize a sound system—by ear or by measurement?
The short answer is both. Each method compliments the other. The ultimate qualification for sound quality is the ear. If it doesn’t sound right, nothing else matters.
I once spent two-plus hours tuning a church sound system using the same methods I’ve employed on more than 100 systems, and the resulting sound was terrible. Rather than trying to convince my ears that the very good curve on the laptop screen sounded just wonderful, I had to trust my ears that something was wrong.
It turned out that my calibrated microphone was damaged. Once the process was repeated with an undamaged mic, the sound quality matched the curve.
The ear is the final judge. No matter how enamoring the technology, common sense must prevail. However, due to things like illness, drug effects, fatigue, poor acoustic memory (from which we all suffer) and hearing deterioration due to age, tuning a system purely by ear will not produce consistent sound quality nor the absolute best a system can do.
Tuning by measurement will uncover problems that the human ear just isn’t very good at detecting, but they are problems that make the difference between sound that’s “O.K.” as opposed to spectacular (and consistent).
I challenge anyone to find the one out-of-polarity transducer in a large system by ear. You might be able to tell something is wrong at a particular listening position, but you just can’t tell exactly what. Measurement pinpoints that type of problem exactly.
Don’t get me wrong - the ear is ideal for certain things. For example, balancing levels between separately amplified multi-way loudspeakers. Level matching two jagged response curves of a woofer and high-frequency section on a computer screen is much more difficult to get right than by ear. This is particularly true when changing just 1 dB of relative level can completely change the character of the speaker system.
And there are many systems that have been “TEF’ed , SIM’ed and Smaart’ed” that plain just don’t sound good. Could this be a big reason some sound folks think that measurement systems just don’t work? I think so. But the issue lies not with measurement systems, which are improving steadily. No, the largest problem, by far, is operator error.
Rather than continue the debate, let’s get busy with addressing the errors that commonly plague us as we tune by measurement.
Operator errors fall into three general categories:
1) The flat RTA response misnomer
2) Improper measurement-mic placement, and
3) Attempted equalization of multiple-source or multiple-reflection-contaminated, FFT measurements
It’s been my experience that 95 percent of all sound systems are equalized improperly due to these three errors, and this is why some “road dog types” thoroughly mistrust measurement geeks.
An RTA (Real-Time Analyzer) is a two-dimensional measurement system that displays energy in dB SPL or volts versus frequency in hertz. Meanwhile, TEF, Smaart, SIM and the like are all 3-dimensional (3-D) measurement systems that display energy vs. frequency vs. time.
Therefore an RTA, unlike FFT (Fast Fourier Transform) based 3-D measurement systems, is time blind and lumps all energy occurring within a fraction of a second together. A fraction of a second is an eternity to a 3-D measurement system.
When measuring an electrical voltage signal, like pink noise at the output of an analog mixing console, the mixer’s electronics have very little propagation delay.
Electrical signals on an RTA display will very closely match what 3-D FFTs display. This is because electronics do not time-smear the original signal. Therefore, if an electrical pink-noise signal is flat on an RTA, it will also be flat on a 3-D measurement system as well.
However, once an electrical signal is converted by a loudspeaker to an acoustical one and reflected around a room, the time smear is substantial. All the energy is not present at the same point in space at a single point in time, nor is it all dispersed from the speaker uniformly with respect to frequency.
The direct sound signal that travels straight from a loudspeaker to a measurement mic will be the shortest path and travel time between the two. Energy that first reflects off a side wall, then a back wall, then the floor, then to the mic, will take many milliseconds more. This later energy appears to arrive simultaneously with the direct sound on an RTA display and will be summed with it.
However, this reflected energy can be ignored by the display of a 3-D measurement system, and it is this characteristic that makes it a superior measurement system.
If you equalize a loudspeaker to be flat on an RTA display with the measurement mic in the middle of the listening area, you’ll be unpleasantly surprised by the resulting bad sound quality. This is not a similar measurement to the electrical one by any means.
Figure 1: The ideal-room curve, or preferred-listening curve with its range of high-frequency variation in gray. Note that the lower limit is the original standard for cinema sound systems with beaming, radial-derivative, high frequency horns. The upper limit fits better for more-recent, constant coverage horns. (click to enlarge)
Types Of Curves
Probably the first person to recognize this difference was Dr. Charles Boner, the godfather of audio consultants who was one the first to practice equalization. He developed what has been called the ideal-room curve or preferred-listening curve. It’s largely the acoustical power response of the loudspeaker system, as modified by air and surface absorption within the room (Figure 1).
What is the power response of a loudspeaker system? Other than being one of the most over-used and least-understood terms used today in audio, it is the sum total acoustic power that a loudspeaker produces.
For example, let’s measure a two-way loudspeaker system in a large room, with the measurement mic in front of the speaker in the middle of the listening area, in the reverberant field beyond critical distance, where the direct sound is lower in level than the reverberant field energy.
Most of the high-frequency energy is aimed in the general direction of the mic due to the directional effect of a horn on the high-frequency (HF) driver.
However, most of the low-frequency energy is not aimed at the mic because the low-frequency (LF) driver is omni-directional for most of its passband.
Therefore, if a flat anechoic or direct sound is desired, much more energy must be generated into the room by the LF driver to equal the sound pressure level (SPL) of the HF driver at the mic’s position.
On an RTA display, this will look like the LF is a big haystack, and the HF gradually rolls off toward the higher frequencies where its horn exhibits better dispersion control.
Obviously, this is Boner’s curve. Where the exact hinge-point of the HF roll-off begins—and just how steep the roll-off is—depends on the dispersion of the HF horn, the number of devices, and whether the LF section has any directional control or not.
In a movie theater, where the size and absorption characteristics of the room, number, location, and specs of the loudspeakers are all fixed, a tightly defined curve can be used. For most other sound reinforcement applications, where every room and loudspeaker system is different, the modified power-response curve that produces a flat direct response can vary a lot.
In the days before 3-D measurement systems, one had to vary the hinge-point and roll-off characteristics of the curve until everything sounded right.
Figure 2: Various measurement-resolutions of a comb-filter caused by a 3-inch signal delay. Even the 1/3-octave resolution cannot clearly show that the frequency-response problem is non-minimum phase, time-oriented, and therefore cannot be equalized. (click to enlarge)
This took a lot of time to get a satisfactory result. It had to be done with each individual system until manufactured one or two-box systems (mid-high packs and subs usually) came along.
Time Oriented Events
Let’s shift focus to “non-minimum-phase anomalies.” This $10 phrase describes time-oriented events that cannot be equalized. Examples of this are other delayed sources, like reflections or more distant loudspeakers, which are delayed enough in time to cancel the direct-sound energy from a loudspeaker at particular frequencies (Figure 2) .
Another is the notch at the crossover frequency of a speaker system when the drivers are not time-synchronized (Figure 3). Neither of these frequency-response problems can be remedied by equalization. These are non-minimum-phase events and cannot be fixed with EQ.
This also applies for reverberation or echoes. Even a change in the reverberant nature of a room, due to a change in its acoustical absorption characteristics, is not an equalizable situation.
Figure 3: Non-synchronized LF and HF drivers cause the notch at 2000 Hz. Since this problem is due to the later arrival of the signal from the system woofer, it is a time-related, non-minimum-phase event, and is also not equalizable. Viewed by an RTA in the reverberant field, this notch would not be revealed. (click to enlarge)
Yes, I can already hear the protests to this statement: “But I’ve had to change the ‘room EQ’ numerous times when it was equalized empty early in the day, and then didn’t sound right when the room filled with people because it was less reverberant.”
Yes, the EQ had to be changed, but it was not due to reduced reverberation in the room. What had to be accommodated was the effect of temperature and humidity changes on the direct sound from the loudspeakers, not the reverberation. These effects on the direct sound are not time-oriented; rather, they are minimum-phase and can be equalized. Why else would a BSS Omnidrive include a meteorology probe?
Congratulations - you say you’ve gotten a new 3-D, FFT-based measurement system and you’re going to tune a system? Next time, I’ll talk about how to use it correctly and optimally.
John Murray is a 30-year industry veteran who has worked for several leading manufacturers, and has also presented two published AES papers as well as chaired four SynAudCon workshops.
Friday, August 16, 2013
Fresh Start: Deploying A New Rig For Kenny Chesney’s “No Shoes Nation”
A different approach for one of the summer's largest tours
Kenny Chesney has established himself as one of the top touring artists in North America, consistently filling stadiums, arenas, and sheds year after year.
His ongoing “No Shoes Nation” tour with his superb band is no exception, keeping a busy schedule at venues such as Dallas Cowboys Stadium and Ford Field in Detroit while also featuring opening performances by Eric Church, the Eli Young Band and emerging artist Kacey Musgraves.
We detailed the sound reinforcement approach for Chesney’s tour last year (here), but this tour is quite notable due to several significant changes. Chief among them is Morris Light and Sound (based in Nashville) deploying a completely new house sound system headed by NEXO STM modular line arrays, as well as Chris Rabold now providing the front of house mix.
I caught up with the tour in late June for a show at Crew Stadium in Columbus, OH, an outdoor venue with a capacity of more than 20,000. Arriving at the back gate that morning, I was greeted by audio crew chief/system engineer John Mills, who’s also the VP of Morris Light and Sound, where he works closely with president David Haskell in managing the company.
We wandered out to the large stage (interestingly, a permanent structure at this facility), where a busy team – PA techs Phil Spina, Justin Meeks, Kyle Fletcher, and Tanner Freese joined by stage patch Jamison “Jamo” Beck – was assembling and flying the main system.
Audio crew chief John Mills at “laptop central” at front of house. (click to enlarge)
As Mills departed to join assistant system engineer Preston Gray in handling some things at front of house, I took a seat in the front row to observe the construction of the arrays, which were up in the air within an hour.
Later, I spoke with Mills about the new system, including specific technical aspects as well as more general topics. He and Haskell began contemplating a change quite some time ago, with an eye on serving the particular needs of the Chesney’s annual touring schedule while also strongly considering the countless events, shows and productions the company serves on a yearly basis.
Local stage hands in Columbus assist the sound team’s array assembly. (click to enlarge)
“We took a look at the market and identified three systems of interest, with one of them being NEXO STM,” he notes. “Our previous large-format system, while still quite good, was aging in terms of technology, and it lacked flexibility. So we determined that an upgrade was in order from both quality and business standpoints.”
He and Haskell arranged for direct comparisons of their list’s top three systems with their existing rig, all conducted in the same controlled environment. They also evaluated each of the three in terms of musicality, throw, and headroom, and significantly, modularity.
Out of this exhaustive process, NEXO STM emerged as their preference. “All of these systems were excellent, one a bit better at a certain aspect, the next a bit better at another aspect, and so on,” Mills says. “But in terms of overall sonic quality combined with extremely flexible modularity, STM won the day.
One of the completed STM array sets with M46 and and B112 modules forming main and side arrays, backed by an S118 subbass array. (click to enlarge)
“The truly modular nature is a big attraction,” he continues. “This is our biggest tour, by far, and then we also work a wide range of smaller tours and events of varying scale. This system allows us to provide a PA that is ideally scaled to the needs of the particular coverage requirements and program material of every one of those events, large down to small.”
“The bottom line is that we wanted to move beyond a ‘stadium box’ approach to something that’s going to be out of the shop, working constantly and delivering consistent revenue.”
The STM (Scale Through Modularity) concept enables line arrays to scale up or down depending on audience size, ranging from 1,000 to 100,000 people.
Arrays can be configured from three loudspeaker modules with the same cabinet width: M46 main, B112 bass, and S118 subbass. (An M28 “Omni” module for dedicated down fill is also coming soon.) Configurations can comprise arrays of main cabinets only, or main plus bass, or bass plus main plus bass.
The M46 main cabinet includes 4 x 6.5-inch LF/MF drivers and 4 x 2.5-inch voice coil neodymium HF compression drivers. The drivers have flat membranes that help produce even full-range coverage over the entire 90-degree horizontal dispersion. Frequency response is 85 Hz to 20 kHz.
The cabinet measures 13.8 (h) x 22.6 (w) x 28.2 (d) inches, sharing the exact same dimensions as the B112 bass cabinet, which is outfitted with a neodymium high-excursion 12-inch bass driver in a hybrid horn-loaded design that helps maximize the efficiency of the driver. Frequency response is 63 Hz to 200 Hz.
M46 and B112 boxes can be arrayed together in a variety of configurations, the most common being paired side by side to form a full-range loudspeakers that are then linked in a vertical array. “It’s this flexibility that really appeals to us,” Mills says. “Want more mid-low? Add some boxes. Want less? Take some away. You can tailor it by both the number and type of boxes in the arrays, as well as the array structures.”
The S118 subbass cabinet, extending response down to 25 Hz, has a neodymium high-excursion 3,000-watt 18-inch driver, with a bandpass load incorporated to help provide output equivalent to some dual 18-inch units. These cabinets can be stacked, flown with other elements, or flown as their own arrays.
The proprietary PistonRig system built into each cabinet offers streamlined compression-mode rigging and allows pre-setting of inter-cabinet angle values, while a “REDLock” handle locks front rigging points from rear of cabinet. All rigging adjustments are made from one position at the rear of an array.
STM cabinets are paired with the NEXO Universal Amp Rack (NUAR), which contains plug-and-play digital patches, real-time system monitoring, and control network functionality in addition to two NXAMP4x4s, which together can power up to 12 loudspeakers in groups of three.
A Morris Light and Sound tech climbing “Mount Line Array” to check a connection. (click to enlarge)
NUAR includes a dual-voltage version of the NXAMP and works in conjunction with the Digital Meter Unit (DMU), an intelligent input patch panel providing digital communication with the NXAMP, and the Digital Patch Unit (DPU), an intelligent output patch panel.
“These drive racks are another aspect of the advantage of modularity. You only carry what you need, and it’s all very efficient,” Mills explains. “We needed 17 drive racks per side for the old system, and on this tour, with a much more powerful system, we’re down to an average of only 11 racks per side, which saves a lot of space, not to mention effort.
“The racks are also flyable, which we’ve chosen not to do on this tour, but it would be great for things like corporate gigs where the client usually wants a very clean stage,” he adds. “For companies of our scope, and even smaller, this package can be a great investment. You can easily handle the occasional big show, or multiple smaller gigs, with the same system, and without having to cross-rent.”
NEXO Universal Amp Racks (NUAR) that drive the arrays. (click to enlarge)
Mills models each venue prior to a show on the Chesney tour, using the NEXO NS1 program that he’s found to be “very consistent,” with the array structures remaining relatively constant from show to show.
Main left and right flown arrays usually run 22 deep (22 M46 mated with 22 B112), with flown side fill arrays of 15 M46 and 15 B112. The flown S118 sub arrays, located several feet behind the mains, are 2 boxes wide, 12 deep.
“Something I’ve been considering for the next tour is going to triple-wide main arrays, with two B112s flanking an M46, and then a single M46 line for each side,” he notes. “We’ll evaluate that fully after this tour is complete.
A closer look at the rigging on the back side of an M46/B112 array. (click to enlarge)
“But I’ve been doing some preliminary modeling and it looks like it will work quite well. Of course, we’ll also need to see what happens with it in actual practice.”
The arrays are supplemented by four NEXO S12s on each end for the stage for near fill, with five to six PS10s spaced on the stage for front fill. Eight RS18 “Ray Sub” subs on the ground provide bolstered low-end presence right up front, and they’re run in their cardioid setting to move stray energy up and away from the stage, and in particular Chesney’s runway “T” stage that runs well out into the audience area. Delay towers are also deployed on stadium shows a couple hundred feet from the stage, with their number and location varying by venue.
As noted earlier, the tour also welcomes Chris Rabold as front of house engineer, and he’s melded nicely with the veteran sound team. He’s mixing on a Midas PRO9 console, his first experience with that particular digital platform.
“It really sounds good,” he notes. “The EQ isn’t like any other digital EQ I’ve used, very smooth and very musical. This desk has just enough color to have some personality, which is nice. I’m happy to say it’s a personality that I get along very well with.”
Rabold got his start at the age of 19 when he hooked up with Widespread Panic, working a few tours with no specific role. “I did everything from sell T-shirts to load the truck. I was in heaven,” he notes, and it spurred him to focus on a career in audio. He attended Middle Tennessee State University and also took any mixing gig available, served as a stagehand, and worked at Soundcheck rehearsal studios in Nashville. His path eventually led back to Widespread Panic, who he mixed for more than a decade.
“I spent 11 incredible years with them, and also mixed whatever else I could during their downtime,” he says. “At a certain point I found myself in the pop world and have been doing mainly that for the past few years.” This includes stints with Beyonce and Lady Gaga, and when the latter’s latest tour was unexpectedly cut short, he got a call from Chesney production manager Ed Wannebo.
Chris Rabold at his Midas PRO9 console at front of house. (click to enlarge)
“I was told coming into this gig that Kenny wanted a big sound, a rockin’ sound with big guitars and what-not,” Rabold says. “Even if I hadn’t been told that, the musical design of the show just begs for it. There’s a whole lot more of a rock feel then I would have expected, as well as some down-tempo stuff and a few songs that flirt with traditional country arrangements, but for the most parts its just a fairly sizeable band going at it with Kenny running around right there with them all night.
“I just went whichever way I thought the music wanted to go, and it worked out really well,” he continues. “Kenny gives me all the vocal I need too, so fitting his vocal on top was pretty easy from the beginning. I really believe that whatever the gig, whatever the genre, if you just focus and listen to what you’re being given, the music will let you know how it needs to be mixed.”
His outboard rack is relatively sparse, chosen for select purposes to augment what he’s able to do with the console’s onboard dynamics and effects. Mainstays are an Empirical Labs Distressor, Fatso, and Derresser, as well as API 2500 stereo compressors, and an SPL Transient Designer. “By far, the coolest piece of gear I’ve stumbled upon lately is the Sonic Farm Creamliner,” he adds. “It’s a tube-based processor that gives some extra weight and muscle to my stereo bus. It’s one of those ‘you have to use it to understand’ pieces.”
A percpective of the stage and arrays in Columbus. Photo by Steve Jennings
Also in the rack is a Neve Designs 5045 primary source enhancer from Yamaha that’s impressed both Mills and Rabold. Essentially, the unit is designed to reduce stray noise that can compromise vocals. “
It stays parked on Wyatt’s vocal, Kenny’s primary backing vocal, and I’ll use it when needed on Kenny himself,” Rabold states. “It’s just a useful problem-solver of a unit. It takes the input signal and splits it into parallel paths. Those paths sum at the output. The trick is that the two paths are out of phase with one another until the signal level crosses a user determined threshold. Think of it like a gate with a really cool way of achieving the gating!
“A vocal mic can sit there amidst all kinds of stage volume and crowd noise and you’ll never hear it open until the vocalists give you the input you want, assuming, of course, their singing creates a level great enough. It’s a threshold dependant process, and it’s not a magic box by any means, but it’s a damn witty way of achieving what it sets out to do.”
Solid & Consistent
On stage, monitor engineers Phill “Sidephill” Robinson (Chesney) and Brain “Opie” Baxley (the band) both utilize Midas PRO9 consoles. Robinson delivers his mix to Chesney’s Shure PSM1000 personal monitoring system, while Baxley serves the band’s Sennheiser in-ear mixes along with a few NEXO 45-N12 line monitors.
Chesney’s distinct vocal is captured by a Shure SKM9HS capsule on an Axient AXT200 wireless transmitter. The upgrade to the Axient system came about on last year’s tour, where Chesney started each show on a second stage out in the audience, roughly 200 feet in front of the PA, and then was transported to the main stage on a flying metal chair.
At front of house, signal from both PRO9s there goes digitally to two Yamaha DME64 digital mix engines outfitted with 16 channels of Lake Mesa EQ for zone control.
Four inputs of Lake are set up for left, right, sub and fill, with 12 outputs running to house left and right, out fill, ground subs, and so on.
Audinate Dante networking takes signal to the NUAR drive racks on stage. The DME64s are also used to switch the two PRO9 consoles as well as the Avid VENUE Profile favored by Eric Church front of house engineer Brent Sparks.
Chesney performing with Shure Axient wireless mic and PSM personal monitoring systems. (click to enlarge)
The sound team performs meticulous system tuning prior to each show, working a zone grid to capture input that’s interpreted with an assist from Rational Acoustics Smaart.
Mills runs the NEXO NeMo remote monitoring app on his iPad that provides control over the NXAMP network, providing him with access to key system factors before and during a show from anywhere in the venue. Rabold also does tuning via a tablet equipped with the Lake app.
“Once John and his crew have verified the rig and its components, he hands it over to me,” Rabold says. “I get his input on why things are the way they are that day, and if there’s anything out of the ordinary. I have a very specific curve I want to achieve every day, and I can honestly say we get that. I don’t want to have to recreate the wheel every day in terms of my mix.
“I like to tune my own system. At the end of the day I’m being paid to deliver a certain level of sound quality to the audience,” he adds. “The better acquainted I am with the sonic characteristics of the room and the system, the better and faster I can confidently react to what’s given to me on a daily basis.
Mills checking things via the NeMo remote monitoring app, with Chris Rabold doing some tuning with the Lake app. (click to enlarge)
“I’ve been blessed with incredible system engineers for most of my career, and we’ve got a great one on this tour.”
For his part, Mills and his team have been quite pleased with the performance of the new rig. “It’s beautiful – musical, solid, and consistent throughout the coverage area, night after night,” he concludes. “The front row and the back row do really sound the same. The vocal is gorgeous everywhere, and there’s great separation of instruments. And Chris does such a great mix, bringing out and then blending just the right elements.”
Keith Clark is editor in chief of ProSoundWeb and Live Sound International.
Elusive Quests: What’s All The (System) Buzz About?
If all else fails, don a bee bonnet
If you’ve been in the live sound business for any time at all, or working in any field related to sound, you’re quite likely to agree that the most maddening technical anomaly of all is the elusive “hum” or “buzz.” These two terms can be related or separate, but they’re both nasty business just the same.
We’ve all set systems - large and small - only to power up and discover the beast within. Is it 60 Hz? 120? 240? A combination? Is it mains? Monitors? Backline? Lights?
First, do no harm! (Yep, this applies to us too.) Don’t start undoing your stage work in a panic. Try to discern what frequency(ies) are the problem. A Real-Time Analyzer can come in quite handy - even a rudimentary online RTA app is worth the investment.
Let’s focus on the common 60 Hz hum. Hopefully you’ve taken the time to verify the integrity of the house power. This is a must. A simple line checker from the hardware store is a cheap way to make sure the house is “keeping it clean” and feeding the proper line voltage – and with the hot, cold and ground in the right order.
Make sure (especially in small/medium house applications) to connect all PA power feeds, as well as backline, to the same source. Don’t mix and match! It’s a sure-fire ground loop headache waiting to happen.
Invest (wisely) in proper power conditioning and sequencing devices. These protect and catch big voltage issues long before they reach your gear. Also be sure to reduce any capacitive coupling by keeping parallel runs of audio and AC power separated as far as possible from one another. Do a little homework on parasitic coupling. It’s a fascinating subject and very important to us sound people who are prone to running lots of wire.
I know, I know, some of you are saying “go digital and clear up a lot of these issues.” True, some of this applies more to us old (and young) “analog dogs,” but digital offers its own sets of problems. I also don’t think I’m speaking out of turn in either case when I say that you’re only as good as what you’re being fed from the house.
Maybe you’ll get lucky and be able to isolate the problem to one or two input channels. And don’t forget the backline. Sometimes a bad filter capacitor in a backline amp can wreak havoc on your system and the stage audio as well. Use those DI ground lifts as needed. Keep XLR cables in mint condition. A few hours of work in the shop can save lots of headaches in the field.
There’s nothing wrong with older power amplifiers, but you might want to have a look inside. Change those old filter caps, or if you’re not budget-challenged, get some new amps. (They’re lighter weight too!)
Watch out for older small-venue fluorescent lighting or newer compact fluorescents. They and their respective ballasts have a way of getting into your wired (and wireless) systems. Most of the time the problem is a poorly grounded “main,” but nothing substitutes for a clean, stable and well-grounded source. (Not always possible, I well know.)
If all else fails, don a bee bonnet. (I always keep one handy in the gear box.) Convince your audience that the sound they’re hearing is a recent insect infestation, smile, and (try) to have a great show anyway!
Greg Stone has worked in live sound since 1976 and is the owner of Hill Country Ears Sound Company in South Texas.
Thursday, August 15, 2013
History Files: The Crown DC300 Amplifier Leads The Solid-State Revolution
A marvel of power density that revolutionized pro audio in 1967...
Not so long ago – the mid 1960s to be more specific – the most reliable large (for that era) power amplifiers were vacuum tube designs, with a unit providing 50 watts per channel being considered “hefty” and one offering 100 watts per channel downright “industrial.”
Solid-state amplifiers were long on promise of being solid, but short on substance. Bad enough that solid-state was unreliable, but it didn’t even put on a good show when it catastrophically failed! A tube could at least put on a good “Roman Candle” display as it’s grids and plates arced explosively. (In high school, I worked on tube amps with multi-loop, high-feedback designs that, on occasion, put on dynamic fireworks shows.)
Things changed in 1967, when Crown produced the DC300, the first reliable, solid-state, high-power amplifier. Rated conservatively at 150 watts per channel, it also offered low distortion and noise.
Two generations of “large” Crown amplifiers preceded the DC300, and were made only in prototype quantities. The first generation (1964) was known as the SA-60-60 and produced an unreliable 60 watts per channel at 8 ohms.
As an undergraduate at Michigan State University at the time, I had developed this design during summer break, a time when the nature of transistor failure mechanisms (and solutions) weren’t known.
My return to Crown the next summer saw further work on an amplifier that was reliable, but when taken to a hi-fi show in New York (1966) was criticized as being too “small” in relation to other 75 watts-per-channel models appearing at the time. (This unit was called the D150, not to be confused with the later model of the same name.)
Front panel view of the DC300. (Click to enlarge)
In the spring of 1966, with a Master’s degree in hand, I went to work on the size problem. The electronic protection methods to be used (VI limiters) were now adequate, and with a newly forming knowledge base on semiconductor failure mechanisms, it was possible to deploy paralleled single-diffused power transistors in a circuit (Class-AB+B) that had ample speed and previously unattained reliability.
Large amounts of feedback allowed the DC300 to set new standards for fidelity at the same time it was making solid-state more than hype. Subsequently, U.S. patent number 3,493,879 was issued for the design.
The naming of the DC300 derived from it being Direct Coupled for Direct Current (DC) operation and having 300 total watts of stereo 8-ohm power. Also at the time, the DC3 was a popular airplane, and it just made sense that this amplifier should be called the DC300 if one were flying higher.
Back view of the DC300. (Click to enlarge)
Overall, the DC300 could produce 500 watts, was just 7 inches tall and weighed 45 pounds. This is very interesting when compared to a 500-watt tube amp of the time, which would weigh at least 200 pounds and perhaps be 35 inches tall - the gains were conspicuous. (Today, a Crown I-Tech I-T8000 produces 8,000 watts, is 3.5 inches high and weighs 28 pounds. That’s really flying higher!)
By 1968, the product was shipping in quantity and finding new markets for DC coupled power. Some of the early adopters were makers of jet engines (fatigue testing) and makers of sonar transducers for the military. With all other models either smaller or unreliable, we had the market pretty much to ourselves for a time.
Gerald Stanley, then and now. (Click to enlarge)
This is an exciting time to be an amplifier designer, as we’re making a paradigm change today that is every bit as profound as the change from vacuum tubes to solid-state. The DC300’s generation of solid-state was a dissipative design approach, where the output signal was controlled by modulating the dissipation in the semiconductors.
In the new generation, the control approach is to use switching statistics. High-speed switches now enable much higher efficiencies and more compact designs than were ever possible for the DC300, which was a marvel of power-density in its day.
Gerald Stanley has been affiliated with Crown Audio for more than 45 years. Read more about him here.
Monday, August 12, 2013
Church Sound: Mixing For The Whole Audience
The live sound experience is different in every seat in your worship space
Have you ever heard a comment from a worshipper, whether positive or negative, regarding the live sound experience that totally differs from what you thought you just heard and mixed?
Large room acoustics (particularly room modes), loudspeaker selection/orientation/optimization, audience size and participation, and several other factors all contribute to the fact that the live sound experience is different in every seat in your worship space.
If it’s a great room with proper system design and installation, those variations may be minor. In many instances, they are not minor.
Either way, they do exist, and the front of house mixer must realize that he or she is only listening to (and mixing to) one position’s perspective when standing behind the mixing console.
During worship, only one of all those factors is under his control: the mix. The best the mixer can do is understand the other factors and learn to mix within that particular environment.
There are some worship facilities where consistency has been achieved across most of the audience area through excellent design and integration. But for the vast majority of venues, it’s one thing to create a brilliant mix for the mix position and another thing to translate that across the whole house.
So it is critical to walk the audience area whenever possible to hear the perspectives of the audience areas (especially if there is a trustworthy A2 to drive the console for a few minutes at a time).
Tonality may be noticeably different in some locations. For instance, it may be discovered that the majority of the house hears a little more bass thump than the mix position does. The mixer that notices this can take it into account in the mixing process. That would never be noticed, and compensated for, without walking away from the mix position.
In addition to tonal variations, it is not uncommon that loudness changes with position as well. If the loudest locations are in the front rows, that may be OK. Wouldn’t even the least technical worshipper expect a bit of a louder experience when choosing a front row seat?
Consider that the overall worship level should be mixed for the loudest location in the house. If that’s not the mix position, then periodic walks are necessary to ensure excessive loudness does not occur at any seat (or the complaints that follow).
If the mixer can only walk the house during sound check or review, OK. If he can walk the house discreetly during the live service, even better. Not only does the presence of the audience acoustically affect the result, but an audience participating in corporate worship (singing) markedly affects the overall sonic experience.
For this author, nothing replaces the value of briefly walk-checking the house during the live worship mixing experience. The varying parameters discussed above, within which we must operate, are mostly results of room design and system design or optimization missing their marks. But rather than blame those factors, learn them, and mix around them. If they are to be addressed and improved, that is for another time (and is off topic here).
So, next time you receive a comment regarding the sound experience in worship, whether positive or negative, make sure you ask where the person was sitting. That can help greatly in understanding and interpreting various perspectives.
Kent Margraves began with a B.S. in Music Business and soon migrated to the other end of the spectrum with a serious passion for audio engineering. Over the past 25 years he has spent time as a staff audio director at two mega churches, worked as worship applications specialist at Sennheiser and Digidesign, and toured the world as a concert front of house engineer. Margraves currently serves the worship technology market at WAVE (wave.us) and continues to mix heavily in several notable worship environments including his home church, Elevation Church, in Charlotte, NC. His mission is simply to lead ministries in achieving their best and most un-distracted worship experience through technical excellence. His specialties are mixing techniques, teaching, and RF system optimization.
In The Studio: The Power Of Mono (Includes Video)
A relatively straightforward process that can make a big difference
Time for a bit of the old school.
Mixing in mono can actually provide a lot of advantages in helping to improve a stereo mix. It seems counter-intuitive, but the mono process puts things more “up front” so that it’s more focused and can be better analyzed.
Joe provides a look at ways to utilize mono effectively, ranging from very simple to more involved methods. The key is to get the focus in the center as opposed to spread-out as it is in stereo. He also supplies some samples of the process of moving between stereo and mono.
It’s a relatively straightforward methodology but can make a big difference.
Enjoy the video.
Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner. Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.
Saturday, August 10, 2013
Variable-D And Beyond: Classic EV Microphone Design & Evolution
Lou Burroughs would demonstrate the 664’s ruggedness by smacking it on a two-by-four...
The Electro-Voice (EV) model 664 microphone, introduced in the mid-1950s, was designed for typical sound reinforcement applications of that era.
EV employees with the company at the time recall that one of the reasons for the 664’s development was to answer the considerable success of the Shure model 55 (Unidyne).
Yet the 664 was hardly an imitation. It’s the first microphone to employ the company’s renowned Variable-D design principle, which is still at the heart of some EV mics popular to this day.
Patented by Alpha (Alphie) M. Wiggins in 1963, Variable-D mics use frequency selective rear ports to achieve a cardioid pattern. This results in considerably less proximity effect in comparison to single-D designs.
Whether this is a positive or a negative is in the ears of each beholder, but suffice to say that Variable-D has earned its place in the “microphone hall of fame.” (Well, if one actually existed.)
The 664 also earned the nickname “Buchanan Hammer,” a moniker paying homage to the company’s then headquarters in Buchanan, Michigan as well as some serious durability.
Word has it that EV co-founder and mic guru Lou Burroughs would demonstrate the 664’s ruggedness by smacking it on a two-by-four board. A later (unconfirmed) demo reportedly had Lou using the mic to actually drive a nail into the board.
The model 635A later assumed the “hammer” nickname, but it pales in comparison: think tack hammer versus framing hammer. However, the 635A did carry on the tradition of being able to withstand brutal punishment. (Editor’s Note: while working at EV, I indeed drove large nails into boards with the head of a 635A, and it still performed just fine - in addition to providing a chuckle.)
A short time after the 664 debut, EV introduced the models 665 and 666 for broadcast. The 665 looks and feels like a 664, but its finish is non-reflecting gray rather than chrome.
Left to right: EV microphone models 664, 666, RE15 and RE20 - an interesting evolution. Photo by Rick Chinn. (click to enlarge)
And while the 664 was capable of high- and low-impedance operation, the 665 and 666 were low-impedance only.
The 665’s connector is an XLR instead of the dreaded 91-series Amphenol four-pin used on the 664. Meanwhile, the 666 was the premium broadcast model, outfitted with a Cannon UA series connector (which looks vaguely like an XLR, but is larger and “D shaped”).
Where the 664 and 665 could attach directly to a mic stand, the 666 required a specialized clip.
Although the 666 was discontinued by the late 1960s, it still commands a premium price on eBay, and many live sound engineers still prefer it for kick drum and bass.
Straight from the source: How EV explained Variable-D in its marketing materials (click to enlarge)
It’s also an excellent horn microphone, and I happen to like the 666 (and its newer incarnations) for electric guitar amps.
The subsequent model 667 combined the 666 design with a transistorized preamplifier. This preamp could supply extra gain if needed, and offered equalization switches for the low and high ends of the spectrum.
A separate on/off switch could be used to add in a presence peak, if desired. The preamp used a mercury battery; it predates phantom powering by many years.
Later, the preamp was abandoned on the models 667A and 668, replaced with internal equalization settings that allowed frequency response to be tailored with use of several pins that “programmed” the equalizer.
The 667A and 668 were primarily intended as boom mics - and -they were the first mics to make use of the Continuously Variable-D principle, with that patent credited to Harold S. Mawby.
The model RE15 came along to replace the 666, but those who knew still preferred the 666, establishing its beginnings as a cult object of present day.
A popular myth goes that the 666 was discontinued because of the satanic implications of the model number, but the people who were there at the time say this just isn’t so. Competition, not the devil, was the end of this microphone.
The RE15 also offered a Continuously Variable-D design, meaning that it had even less proximity effect than the 666, and its polar patterns were very consistent with frequency.
Although the RE15 never attained the cult status of its older brother, it was a favorite with broadcasters because of its smaller size.
Laugh if you want, but The Lawrence Welk Show used a bunch of RE15s to replay its 666s. The more uniform polars contributed to less acoustic phase interference in the finished mix, and the resulting cleaner sound was not lost on the ABC television network’s technical crew or on Welk’s people.
The 667 mic and its companion preamamplifier/equalizer. Note the “curve-plotting” capability on the preamp—very cool. Photo By Rick Chinn. (click to enlarge)
The final chapter in our story is the model RE20—ever have heard of it? The late Tom Lininger was the principal designer, and it was originally conceived as a “condenser killer.” It was quickly adopted it for a variety of tasks in the studio, mostly relating to things that were either loud or low.
Broadcasters also found it to be a very good announce mic, and it’s still popular in that application today, as is the RE27N/D, which incorporates a neodymium element.
Oh—and let’s not forget that the RE20 (sometimes also branded as the PL20) is still one of the most popular kick drum mics in sound reinforcement some 35-plus years after its introduction.
Frequency response and polar response of the RE20. (click to enlarge)
Take a look at the RE20 response curve, and you can see that it indeed offers the high end of a good condenser.
While history says that it didn’t really “kill” the condenser genre, the RE20 has nonetheless more than earned its place in the “mic lockers” of many. (There’s an interesting marketing lesson to be found here.)
Rick Chinn is a long-time audio professional and history buff. He heads up Uneeda Audio. Find out more about Rick and the company at www.uneeda-audio.com.
Telex Communications and Electro-Voice
Allied Radio Catalogs: 1954, 1955, 1956, 1957
US Patents: 3,115,207, 3,378,649
And “Those Who Were There”—the author’s heartfelt thanks to all of the following EV folks for helping with 50-plus-year-old memories:
Jim Long, senior sales support engineer
Bill Raventos, product manager, professional products
George Riley, marketing manager
Don Kirkendall, manager of advertising and promotion
Frank Spain, national service manager
Lloyd Loring, sales promotion manager
Editor’s Note: EV continues to sucessfully utilize Variable-D technology, as evidenced in the RE320 microphone introduced just a couple of years ago. Read about it here.
Friday, August 09, 2013
The Difference Between Replaceable And Invaluable
Learning to take care of someone else’s business is a great way to end up with your own
During my time with the last AV crew, I watched people come and go. Most who stayed for any length of time fit into one of a few categories:
—People who worked cheap.
—People who the owner felt obligated to keep.
—People who were absolutely invaluable.
That’s pretty much it.
We had our roadies. They set up shows, broke them down, hauled gear, loaded and unloaded trucks, cleaned the shop and other random jobs that involved a lot of sweat. The roadies were mostly good guys. Hard workers who did their job. But easily replaced.
Most of them were guys who had minimal technical knowledge and people skills. They were the guys we kept away from the clients and audience as much as possible. Like Igor and Dr. Frankenstein. Igor probably knew his way around the lab as well as the doctor did, but would never get to run the show. Igor was the grunt. He kept the doctor from getting his hands dirty.
Now, I’m not bashing these people. There were some really good folks who worked their butts off to make things happen. But, there were a lot who were determined to be useless.
Some had been around so long, the owner just tolerated whatever stupidity they produced each day. Wrecking trucks, losing gear, delivering the rig to the wrong venue, falling asleep in the truck 200 miles away while we waited to set up a show, pulling away from a dock without strapping gear down or closing the rear doors. Yeah. We saw a lot of that.
Those guys made less in a week of hard days than most techs made in a day. Seriously. But they kept showing up even though a job at Burger King would have paid more each week. Why?
The job gave them the flexibility to maintain their lifestyle. Staying out drinking all night, showing up looking like a dead cat, smelling like old liquor, talking trash and arguing with everyone would get you fired from most jobs. Not these guys—it seemed to improve their resume.
They were either family, old friends or just guys who worked so cheap it was worth tolerating. Some of them had potential to make good money as techs. A few actually worked their way up and became techs. But most are still doing the exact same thing as they were years ago.
So, who gets paid well and keeps their gig year after year? Mr. Invaluable.
The owner was very skeptical of any new guy. It was easier to get hired with the Secret Service than to get in with him as a tech. He had been burned so many times by hacks that he was paranoid about placing a new guy with a client. He just expected everyone to cause problems.
I got in, again, because of someone I knew.
His girlfriend was a bartender at a place one of my friends liked to shoot pool. They got talking one night about what they did for a living. When my friend heard what he did, he threw my name into the conversation. The timing was perfect.
We were days away from a week-long local festival. Shows every night and during the day. Three or four stages around town and he was short on techs. I got a call, met with him and worked that weekend. He put me side by side with one of his main techs for that first day. We hit it off great. I ended up building a stage that morning and mixing monitors for the show that night. Everything seemed to work perfect.
I found out a few days later that the owner grilled that guy about me. He wanted to know how I handled everything and if I knew what I was doing. He admitted to telling the owner that I probably knew my stuff better than he did.
That gig lasted seven years. Never really quit, just moved away. Still might work with them again. We left on good terms. Just needed to get the family to a better place.
During those years, I had the opportunity to make myself more valuable each day. Every problem that came up gave the roadies something to gripe about and gave me a chance to shine. I am a problem-solver by nature. I like to figure things out. I’m always looking for a better and more efficient way to do everything. Apparently those are very valuable job skills.
The owner had lots of powered monitors. They were decent speakers, but nobody knew anything about them. When one quit working or made weird noises, it went to a better place. It went to the monitor graveyard where it spent the rest of its lonely life resting in a corner. It had to. Nobody there knew how to fix them.
After a few weeks working with them, it finally happened. A monitor died during soundcheck. It was unplugged and moved off stage. That’s all. We were short on monitors, so I broke out my tool kit and opened it up. Found a loose wire or something simple. Plugged it back in and it was fine.
I heard the owner backstage, 10 minutes later, chewing out the other techs. He wanted to know why the new guy was willing to fix stuff but they wouldn’t. They were making excuses about not knowing what was wrong.
He came out and asked me how I knew what the problem was. I told him I didn’t. I just opened it up and looked for a problem. That was exactly what he wanted to hear.
I ended up getting work even when nobody else did. Within a year, he was sending me all over the country running shows. Alone. He would fly me to a venue and have a truck meet me there. My show.
I had become invaluable. I was a tech who could mix and manage a show. I could run the crew for load in and load out. I could fix problems on site. I was willing to tackle problems and find solutions. I treated his gear like it was mine.
I’m not telling you this to brag about me. I want you to understand how to get and keep a job. I’ve already talked about attitude a lot. Going to keep talking about it. Skills will get you a job, but attitude keeps it.
Someone once complained to me that if they were only getting paid $8 an hour, they were only going to give $8 worth of work. That is completely wrong. If you want to make $50 an hour, you better produce $50-an-hour work.
You make the company look good, make them profitable, make their clients happy, build confidence with the owner, and the money will catch up.
Very few people start out on their own. Most of us have to start out working for someone else. Learning to take care of someone else’s business is a great way to end up with your own.
It’s also a great way to develop that reputation of excellence that will carry you from now on.
M. Erik Matlock is a 20-plus-year veteran of pro audio, working in live sound, install, and studios over the course of his career, as well as owning Soundmind Production Services. Erik provides advice for younger folks working (or aspiring to work) in professional audio at The Art Of The Soundcheck—Random Stories and Wisdom from an Old Soundguy. Check it out here. style=