Monday, September 30, 2013
In The Studio: The Power Of Subgroups (Includes Video)
A way of making the mix process more efficient
If you’ve been mixing for any length of time, you probably know how useful (and cool) subgroups can be.
But recently, Joe Gilder has come up with a different take on doing subgroups—a small change, but something that’s definitely had a positive impact in his workflow and has helped speed his mix process.
Previously, he only subgrouped things he felt should go together—guitars, or vocals, or drums—but that approach has changed. In this video, Joe explains his new strategy on subgroups, offers additional specifics, and explains the reasons why it’s making things more efficient.
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.
Blokhed Studios Invests In Soundcraft Si Expression 2 Digital Live Sound Console
A perfect fit for demands of local club gigs
Tom Stiegler, owner of Blokhed Studios (Deer Park, NY) is a drummer, recording, mastering and live sound engineer, and he’s also a self-described “sonic perfectionist” who didn’t want to make the move to live digital mixing until it met his standards. When the Soundcraft Si Expression 2 console hit the market, he was finally convinced to move to digital.
“I hated it,” was his reaction in finding earlier digital models thin-sounding and impractical to use because the faders had to be zeroed out whenever they needed to be reset. He shied away from trying a digital console again until recently, as prices came down and more manufacturers entered the market.
This time, Stiegler was more wary and spent a lot of time “obsessively researching my options” and making sonic comparisons. “Soundcraft was on my list. I always had faith in their mic preamps, and the new Si Expression Series looked like it had the features I needed. The Si Expression 2 was price-competitive with other comparable brands.”
After comparing various digital consoles in his price range, Stiegler chose the Si Expression 2. “I’m a mixing and mastering engineer,” he notes. “I hear everything. What seems like a little bit of a sonic difference to others is a big amount to me.”
Stiegler does a lot of live sound work for local Long Island and New York Metro area bands at outdoor and indoor gigs, where bands have come to rely on him for a good-sounding mix no matter where they’re playing—even in clubs with less-than-stellar acoustics and house sound systems. “I feel that local bands should have a live sound mix that’s as good as any national act,” he states. Easier said than done in some venues, which is why Stiegler often brings in extra PA gear and appreciates the Si Expression 2’s ability to adapt to any live sound situation.
In the hectic environment of club sound, Stiegler will often have to set up the Si Expression 2 in less than an hour—or even have only minutes to build a mix. He finds the console to be adept at getting him where he needs to be fast, thanks to its motorized faders, logical layout and features like Soundcraft’s simple, time-saving cue/snapshot system that lets Stiegler instantly recall any board settings from any club he’s ever worked in – “a beautiful thing.” Also useful is Soundcraft’s TOTEM (The One-Touch Easy Mix) system that reconfigures the console and allows users to press a single key to mix to an AUX, FX or Matrix bus.
Stiegler adds that all the compression, EQ, noise gate and other controls he needs the most are at his fingertips, with the “analog section” of the console right at the top and the touchscreen interface providing ready access to desired functions. Stiegler finds the FaderGlow color-coded illuminated faders a lifesaver in clubs where front of house is thrown into near-darkness when the house lights go down.
The Si Expression 2’s iPad compatibility gives Stiegler the ability to make adjustments to the console from anywhere in the room—an extremely useful feature that lets him adjust the mix from anywhere in the venue and set each musician’s monitor mix from the musician’s on-stage position.
In making the move from analog to digital, Stiegler found there was a learning curve, but not too big a one. “The Si Expression 2 is not that hard to learn. The biggest mistake you can make is walking into it thinking it’s going to be too complicated. If you’re thinking about buying a digital console in this price range, the Si Expression 2 is the one to get—it’s not even a question,” he concludes.
Church Sound: What Do You Do When Facing An Acoustical Nightmare?
If you go into it informed, you still stand a fighting chance
On a recent weekend I attended an event that was held in a very challenging space for sound reinforcement, and thought I’d detail it because it’s a good way to discuss some basic audio principles.
This particular event was held in a very, very nice pole barn. Really beautiful. It’s a large steel building, the walls covered by solid wood paneling, the floor a stained and sealed concrete, the ceiling made up of flat steel decking with no acoustical treatment. Dimensions were about 60 feet by 100 feet, forming a rectangle with parallel walls.
We were there for a formal ball, with dancing directed by a “caller” and CD tracks for music, both sources fed into the sound system. When I arrived, the person setting up sound was trying to get an omnidirectional lavalier microphone working. Not a good start. Even before hearing anything in the room, it was obvious there were going to be problems, and using an omni mic—particularly one that would be placed on the chest rather than at the mouth—is not a good way to capture strong, consistent vocal signal.
The system also included a simple 8-channel mixer, CD player, amplifier and a single 15-inch, 2 way loudspeaker (circa 1970s). In a lot of ways things were getting even tougher…
First, as predicted, the omnidirectional lav placed at mid-chest made gain before feedback tough. This became immediately obvious during the brief sound check. Second, the loudspeaker, with its “smile-shaped” horn, had very little effective pattern control.
Third, the mixer had minimal EQ, just some high/mid and low knobs usually associated with very inexpensive units. Fourth, the loudspeaker stood over three feet tall and probably weighed north of 100 pounds, so even if the sound person had a tripod stand, the loudspeaker would be too heavy for it. So instead, it was left sitting on the floor, and with associated equipment sitting on top of it.
Thankfully, he also had a wireless handheld on site, and switched to that instead. Once he got that working, he began testing the CD player. Right away I thought, “You’re playing everything way too loud. In this situation, less is definitely more.”
After some (very loud) feedback, caused when the caller for the dance went to turn off the CD player (one of those pieces of gear on top of the loudspeaker) while holding handheld mic by his side (and pointed right into the high-frequency horn), the event began.
Right away, it was obvious that the space was being excited with way too much energy, and unless you happened to be in the direct coverage pattern of the high-frequency horn, all you could hear was the mash being created by the reflected sound that was bouncing around in the room.
Besides the fact that this venue, while nice, wasn’t conducive to this type of event, the primary problem was direct versus reflected energy. Simply, this means to direct as much energy into the coverage area as feasible while minimizing energy that can reflect off of surfaces.
What would I have done?
1) Outfit the caller with a headset mic. It would sit consistently at the same distance from his mouth, thus providing a much more consistent vocal signal. Further, the caller would have been spared to problem of trying to explain/demonstrate a dance while holding a mic.
2) Add more loudspeakers, and these would be distributed around the edge of the dance floor. More sources of sound, more evenly distributed, helps achieve better direct energy while limiting reflected energy.
3) At a minimum, place the loudspeakers at ear height, or if possible, place them higher and then angle them downward. Again, this focuses the energy on to the coverage space and off the walls.
4) Turn it down! Always remember that in a situation like this, “less is more.”
Thankfully there was just the one mic and the CD player. If there had been a live band—wow it would have been really, really ugly!
As a side note, after the event was over, about a dozen or so attendees got together on one end of the building and sang a song a capella in four-part harmony. It sounded absolutely beautiful. The direct-to-reflect energy was just about right, and the natural reverberation of the room added an incredible effect to the voices of the singers.
Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 30 years.
DiGiCo SD10 Console Keys System For Kansas 40-Year Milestone Concert
SD10 chosen for additional production inputs; SD Convert software makes console-to-console transition seamless
Legendary prog-rockers Kansas recently celebrated its 40-year anniversary with a one-off special concert at Benedum Center in Pittsburgh.
Accompanied by a 41-piece symphony orchestra in the first act, the next offered a traditional, rockin’ set of their classics. The event brought together original members for the first time in 30 years: the five touring members (original members Phil Ehart, Steve Walsh, Richard Williams, and long-time members Billy Greer and David Ragsdale), as well as guest spots by original members Kerry Livgren and Dave Hope.
One day prior, the production crew—including PM/FOH engineer Chad Singer, monitor engineer Derek Papp, audio tech Darian Cornish, Benedum A1 Chris Evans, guitar/violin/bass tech Jeremy Vig, drum tech Eric Holmquist, equipment relocation specialist Rusty Banks, and LD David Manion—embarked on a large-scale lighting and video load-in followed by backline and audio for the band in time for an afternoon symphony rehearsal.
The Benedum house PA was comprised of Meyer M’elody arrays and two 700HP subwoofers per side, supplemented by front fills, out fills, a center cluster, and balcony fills. Singer opted to go with a DiGiCo SD10 at FOH (running on an Optocore fiber snake) with a DiGiRack (on a MADI snake).
“We’ve been using an SD8 for FOH since 2012,” says Singer. “We typically do 60-70 shows per year so the console was out with me throughout the year that time. I rented the console from Rock N’ Road Audio in Atlanta, who have a wide selection of DiGiCo consoles available and have been big DiGiCo fans for a long time now.
“For this show, I needed more inputs than our usual band only inputs, so the natural choice was the SD10—especially since the new converter software was able to take my existing show and load it into the SD10. Immediately, I was able start to building the extra inputs. The transition was seamless. In general I have enjoyed using DiGiCo products because they have great preamps/ converters with plenty of headroom. The routing and layout of the consoles hardware and software has the most analog feel of any digital console out there.
“I don’t find myself ever having to dig though menus or pages especially during show time when all I want to think about is being creative. The consoles just make sense. I think my favorite part of the DiGiCo line is having 12 faders per bank. I had to do a fly-date show with a different console a few weeks ago and had only eight faders per bank.”
For the occasion, Singer managed a total of 92 FOH inputs and 10 FOH outputs (Left, Right, center fills, lip fills, subs to the Meyer Galileo Processer, TB, stereo symphony mix, and stereo band mix to monitor desk), with the SD Rack handling the 44 band inputs (14 drum channels, two bass, six guitars, two acoustics, six keyboard, two violin, four vocals, intro music, guest inputs for guitar, bass, keyboards, and vocals, plus four audio from video channels, and four emcee mic inputs.
The orchestra consisted of 41 members and the DiGiRack handled all 46 symphony input channels (20 DPA 4099 string mics, six woodwind channels, 10 brass channels, four tympani mics, three percussion mics, harp, and a stereo keyboard.
“Even with a pretty controlled stage volume,” notes Singer, “the only way to have a fighting chance to clearly hear the orchestra without a large phase spear from the drums is to avoid sectional area mics and close mic everything.”
The monitor desk only received the split inputs from the band inputs so Singer sent him a stereo string mix, a stereo band only mix (Derek combines it with his ambient mics for the band’s ear mixes), and a Talkback channel. “The only tricky part of the mix was creating a band-only mix and a symphony-only mix for returns to monitor world,” he says, “the desk was pretty quick making two more stereo auxes and dialing up post-fader sends of the appropriate channels.”
Singer says he found a lot to like about the SD10. “The desk can do so much. Even with all those inputs I did not need any extra gear to make audio life any easier. In addition to the 12 faders per bank, I’m a fan of the 12 CG’s. You get 24 on the SD10, so grouping can be real specific. Also, the dynamic section is very versatile and with some channels such as the tom-tom channels, I’ll use a normal comp followed by a gate.
“I’ll then send the 6 toms to a group where I can have a multi-band to catch low-mid frequencies and follow that with another comp set more like a limiter at the end of the chain and then blend that group in with the original tom channels that are already assigned to the master bus. All controlled with one CG. Then other channels like the symphony violins I use the multi band to catch upper mid frequencies and or the de-esser to knock down some of the shrill factor. I like using those to bring down cymbal and snare bleed without beating up an EQ and therefore altering the sound of the violins or cellos as well.
“The dynamic EQ is cool too. I use it on bassist Billy Greer’s vocals. He mostly sings in upper registers, so I find myself padding the upper mid-range and hi- frequencies while he sings with EQ and compression. But when he talks to the audience in between songs he speaks softer with a low voice. So instead of always cutting low-end and adding high-end to the EQ manually, I just set the dynamic EQ to do it for me.
Singer makes use of spill sets to handle all of his outs, pink noise and iPod music all in the same fader bank. “I have a ‘Dust in the Wind’ spill set for when they switch from loud guitars and drums to soft acoustics with only a few channels. It just makes organization of channels easy. Also, the DiGiTuBes on acoustic guitars and violin are pretty cool. I don’t know if it’s the right thing to do, but I slam the drive on the acoustics until the noise floor gets to be just too much. Then back it off a bit. It seems to make finger picking have some life and make it sound like it is just on the edge of taking off.”
He’s making use of all the onboard effects for reverbs, delays, and chorus effects, using mostly plate reverbs for drums, acoustic guitars, violin, vocals (and single tap delay and double effect for vocals), as well as a chorus on the acoustic guitars.
As for outboard gear, Singer uses the Waves SoundGrid bundle for instrument specifics. “I’m using Renaissance comp/exp on kick drums, Renaissance AXX comp on guitars, C4’s on snare, vocals, and violin, and an H-Comp on toms and keyboard groups where I can really clamp down on the compressor and then find the right blend between the compressed and original sound with the mix knob.
“On my master bus, I have a chain that’s the H-comp, C4, L1 limiter, then an analyzer. It’s nice to have two insert points on each channel so you can really put the out board processing where it needs to be placed exactly in the signal flow either pre or post EQ/dynamics. It’s also great to have parameter control of those through the SD 10. Since they all change from day to day depending on the venue, new strings, new drum heads, or even if the performer is playing stronger or softer than usual.”
Friday, September 27, 2013
Church Sound: Mixing Like A Pro, Part 2—Channel Layout
Grouping channels in a way to find them quickly and easily
Editor’s Note: Go here to read Part 1 of this series.
Experienced audio people have the ability to mix while keeping their eyes on the stage and the crowd. They know that they need to continually watch their musicians for visual cues in order to catch any trouble early and to verify that the people on stage are able to connect with each other.
Just as important, you should be regularly watching the audience to make sure they are engaging with the music and message. The surest way to verify that the mix is full and engaging is to watch your audience and see if they are into it.
I regularly teach new audio people that if they don’t see at least a few heads bobbing or toes tapping in the audience, they need to listen critically to their mix for changes that need to be made.
Know Right Where To Find It!
So if you’re going to mix like a pro and keep your eyes scanning the stage and the audience, having your eyes locked on your mixer is not an option. You need to be comfortable enough on your mixer that you know where everything is so you can move quickly to any given input without searching for it.
I can’t tell you how often I see churches make the mistake of having a poor input channel layout, which frankly contributes to struggling with the mix. Inputs should be arranged in a way that makes sense to anyone who walks up to the console, and should be clearly labeled so there’s no guessing or hunting for inputs. It seems like such a simple concept, yet so many churches don’t practice it.
Create a Standard Channel Layout
There are two common ways to layout the inputs of the console. The first one is what is very typical on concert tour riders and is the old school way of laying out your console. It’s the method that I tend to prefer, and it’s the layout I’m very comfortable mixing with—meaning that regardless of what console I’m on, I can find channels pretty fast without having to search for them.
Starting with channel 1 and working across, this method looks like:
Rhythm (electric and acoustic guitars, pianos and keyboards)
Playbacks (like CD, DVD, computer, etc)
Drum Mic Channel Layout
If you’re miking the drums, chances are that you’re taking up several channels on your mixer to control the sound of the kit. Having an order for the drum mic channels is also necessary for you to make quick, accurate adjustments without having to think too much about which fader controls which mic.
Here is a common order for average drum set ups based roughly on mic priority:
Kick | Snare | Tom1 | Tom2 | Tom3 | Overhead L | Overhead R
Alternate Channel Layout
Another accepted method for stages that stay set up the same week after week is to lay the console out in the order that you see things on stage. The idea behind this layout is that the inputs are left to right as your stage appears from left to right, so as the physical matches the visual.
This method can make a lot of sense for churches that leave their setup the same all of the time, though for most of us that just isn’t reality. As an example of this setup, if the stage from left to right included a singing bass guitarist, lead singer/acoustic, drums, singer and an electric, your inputs would look like:
Everything In It’s Place!
Regardless of what method you choose, the key is to choose some kind of organization that will group your channels in a way that you can find them quickly and easily, without searching. A console that has mic assignments spread out all over the place is tough to operate without focusing on the console.
With a consistent and logical channel layout, you’ll be able to mix quickly while keeping your ears and eyes up and active.
Duke DeJong has more than 12 years of experience as a technical artist, trainer and collaborator for ministries. CCI Solutions is a leading source for AV and lighting equipment, also providing system design and contracting as well as acoustic consulting. Find out more here.
Thursday, September 26, 2013
In The Studio: Misunderstood Points Of Threshold Based Effects
How they work, what the jargon really means
In the world of recording there are numerous kinds of effects. However, often there are more terms and details specific to each device than the average engineer would care to learn before jumping in and using the new equipment.
Details are very important, though, and are critical to understanding the basic opperation of all equipment. So let’s take a look at threshold based effects and make sure we all have a good understand of how they work and just what everything means.
Some effects are triggered when a sound volume passes a specific point called a threshold. A good example of a threshold based effect is a gate. A gate will mute a sound (stop it from playing) until the sound reaches a loud enough volume to reach the threshold.
Then the gate will open and the sound will play. Quiet sounds that are below the threshold will not trigger the gate to open and loud sounds that are at or higher than the threshold will trigger the gate to open and the sound to be heard.
This makes a gate very handy if you wanted to be able to hear important loud sounds and automatically mute quiet sounds that are not supposed to be heard, such as low background sounds or noises.
Some effects will become more extreme when the sound passes beyond the threshold and keeps getting louder.
A ratio of 1:1.
For example a distortion effect that starts to sound dirty when a sound reaches a minimal threshold volume will get dirtier as the incoming sound gets louder.
Some effects use a ratio to set the amount of processing that happens once the threshold is passed. For example, an effect used to control volume (a compressor / limiter) can be set to a ratio of 1:1 (no change past the threshold), a low ratio such as 2:1 (for every 2 dB of volume increase the compressor only allows 1dB of change), a heavier ratio such as 8:1 (for every 8 dB of volume increase the compressor only allows 1 dB of change) or a very heavy compression ratio (called limiting) such as 10:1.
You can even limit sound using a ratio of infinity : 1, which means that no matter how much louder the incoming sound gets beyond the threshold everything will be squeezed into only 1 dB of change.
Although usually the sound that is being processed in the effect is used to judge if the threshold is reached and the effect is triggered, it is possible to use an external trigger (called an external key) to trigger the effect.
Imagine a long sustaining vocal note that is being processed with a gate that is using an external key from a drum beat. The gate would open when the drum beat reached the threshold and the vocal note would be heard with the timing of the drum beat.
It is possible to use a side chain to trigger the threshold, which often involves using the original sound for the trigger, but processed. A typical example of this would be if you have a snare drum recording with leakage from the kick that you want to remove. The snare will be much brighter sounding than the kick, which will sound more like a thud.
Multiple ratios demonstrating compression.
If you used a gate on the track, but instead of just using the track from the trigger you first used an EQ to remove low thud parts of the sound and accentuate the bright parts, then the trigger will have more snare sound than kick and more easily hit the threshold only when the snare is playing but not the kick.
Another example is to use a compressor on a vocal to reduce how loud sounds get, but to trigger that compressor using an EQ’d sidechain that has all the low sounds removed so you can only hear the SSS of the vocal.
Such a sidechain would compress only when the sss sound is heard, which will created something called a de-esser which is used to control sibilance in vocal performances.
Threshold effects sometimes use attack, hold and release settings. Attack refers to the speed with which the effect will begin to work after the threshold is crossed.
Release refers to the speed with which the effect settings will return to the original setting if the trigger goes down below the threshold level.
Hold (also called sustain) refers to a time that passes between when the trigger drops below the threshold level and when the process starts to release.
Bruce A. Miller is an acclaimed recording engineer who operates an independent recording studio and the BAM Audio School website.
Church Sound: A Primer On Phantom Power For Microphones
We use it all of the time, but what's it really doing?
Condenser microphones need phantom power to operate their internal circuitry. The power is supplied to the mic through its 2-conductor shielded cable, and can be provided either from a stand-alone device or from a mixing console (at each mic connector).
The microphone receives power from, and sends audio to, the mixer along the same cable conductors. It’s called “phantom” because the power does not need a separate cable; it’s “hiding” in the signal conductors.
According to DIN standard 45596, phantom powering is a positive voltage (12-48V DC) on XLR pins 2 and 3 with respect to pin 1. The cable shield is the supply return. There is no voltage between pins 2 and 3. Pin 1 is ground; pin 2 is audio in-polarity, and pin 3 is audio opposite-polarity. Also, pin 1 has 0 volts; pin 2 has a positive voltage, and pin 3 has the same positive voltage as pin 2.
The phantom on/off switch in many consoles is labeled “P48” to signify “phantom power 48 volts”.
Figure 1, top, shows a microphone plugged into a stand-alone phantom power supply. Inside the phantom supply (Figure 1, middle) are two equal resistors R.
Figure 1. Phantom power equivalent circuits.
They supply equal voltage to pins 2 and 3 with respect to the pin 1 ground. Inside the mic, phantom power is tapped off two equal resistors (or a center-tapped transformer).
Figure 1, bottom, shows how the phantom current travels through the mic cable from right to left:
1. The current leaves the DC power-supply positive terminal and goes through two equal resistors.
2. The current travels along the mic cable to the mic.
3. The current is recovered inside the mic and goes through the mic circuitry.
4. The current returns to the DC power-supply negative terminal via the cable shield.
Some microphones or mic capsules work on DC bias rather than phantom power. A separate wire supplies B+ to the mic capsule. You’ll see this arrangement in lavalier mics or choir mics between the mic capsule and its XLR connector.
The mic capsule itself runs off DC bias, while the XLR connector houses a circuit that runs off phantom power. That circuit converts phantom power to DC bias for the mic capsule, and balances the signal.
Why Condenser Mics Need Powering
Let’s explain why condenser mics need power in order to operate.
In a condenser microphone transducer (Figure 2), a conductive diaphragm and an adjacent metallic disk (backplate) are charged with static electricity to form two plates of a capacitor.
When sound waves strike the diaphragm, they vary the spacing between the plates. This varies the capacitance and generates an electrical signal similar to the incoming sound wave.
The diaphragm and backplate can be charged in two ways:
1. By an externally applied voltage (from phantom power). This arrangement is called “external bias” or “true condenser”.
2. By a permanently charged electret material in the diaphragm or on the backplate. This is called “internal bias” or “electret condenser”.
The output of the condenser mic capsule is extremely high impedance so it is very hum-sensitive.
To bring that impedance down to a usable value, an impedance-converter circuit is connected to the capsule output.
Figure 2. A condenser transducer (mic capsule).
This circuit is necessary whether the capsule is electret or non-electret. The converter needs a DC voltage to power it, and this voltage is supplied by the phantom power supply. Sometimes other transistors are added to give the mic a balanced output, and these components work off phantom power too.
In contrast, a dynamic microphone needs no power because it has no active electronics. It generates its own electricity like a loudspeaker in reverse. In a moving-coil dynamic microphone (Figure 3), a coil of wire is attached to a diaphragm.
Figure 3. A dynamic (moving coil) transducer or mic capsule.
This voice coil is suspended in a magnetic field. When sound waves vibrate the diaphragm and its attached coil, the coil vibrates in the magnetic field and generates an electrical signal similar to the incoming sound wave. The voice-coil leads are soldered to XLR pins 2 and 3 inside the microphone.
In a ribbon microphone (Figure 4), the diaphragm is a thin metal foil or ribbon. Sound waves vibrate the ribbon in a magnetic field and generate corresponding electrical signals. The ribbon leads are soldered to XLR pins 2 and 3 inside the microphone.
Figure 4. A ribbon transducer (mic capsule).
You can plug a dynamic or ribbon microphone into a phantom supply without damaging the mic. That’s because the voice-coil or ribbon leads are not connected to pin 1, so no current from the phantom supply can flow through them.
However, if there’s any imbalance in the phantom voltage applied to pins 2 and 3, a current will flow through the microphone voice coil or ribbon (which is connected to pins 2 and 3).
Or if one terminal of the coil or ribbon is accidentally shorted to the grounded mic housing, a current will flow through the coil or ribbon. For this reason, it’s best to switch off phantom power for dynamic and ribbon mics.
Using a Stand-Alone Supply
You can buy a phantom power supply from your microphone dealer or online. Some supplies are AC powered; some are battery powered; some are both. Some can power a single microphone; others can power several at once.
In any case, you plug the supply in series with the mic cable. The supply has XLR-type input and output connectors, one pair per channel.
Connect a mic cable between your microphone and the supply’s input connector. Plug another mic cable between the supply’s output connector and your mixer mic input.
If you need to convert a low-Z condenser mic to high-Z unbalanced using a transformer/adapter, run the mic through a stand-alone phantom supply before converting it to high-Z unbalanced.
Cautions for Use
Don’t plug a mic into an input with phantom already switched on, or you’ll hear a loud pop. If you have no choice (as during a live concert), mute the mic channel when you plug the mic in.
Make sure your phantom voltage is adequate for your microphones. Some mics start to distort or lose level if the phantom voltage drops significantly below 48 volts.
To measure the phantom voltage at your mixer’s mic input, get a DC voltmeter and measure between XLR pins 1 and 2. Do the same between pins 1 and 3.
Be aware of phantom-voltage sag. Microphones draw current through the phantom-supply’s resistors, and that current causes a voltage drop E = IR across each resistor. The higher the current drain of a microphone, the more it drops the phantom voltage at the mic connector. Current drain is usually specified in the mic’s data sheet.
Power supplies are rated in the total number of milliamps they can supply. Make sure that the total current drain of all the mics plugged into the supply doesn’t exceed the supply’s current rating.
Avoid having phantom in a patch bay because someone is likely to patch in and cause a pop. If you must patch into a jack with phantom on it, mute the input module that the mic is connected to, or turn down its fader. Mic-level patches should be avoided anyway.
Some phantom supplies cause a hum when you plug in a connector that ties the shell to ground. Float the shell. This also helps to prevent ground loops.
Since the cable shield carries the DC return, be sure the shield and its solder connections are secure. Otherwise you can expect crackling noises—especially when the cable is moved.
Some microphones work on either internal batteries or external phantom power. In most designs, connecting the mic to phantom automatically removes the battery from the circuit. Otherwise, the battery would severely load down the phantom supply. It this appears to be happening, remove the battery.
If a condenser microphone doesn’t work due to low phantom-supply voltage after the mic is plugged in, try these suggestions:
1. Supply phantom from a better-regulated console.
2. Use a mic with less current drain or with lower phantom-voltage requirements.
3. Add a voltage regulator to the supply voltage.
AES and SynAudCon member Bruce Bartlett is a recording engineer, audio journalist, and microphone engineer. His latest books are “Practical Recording Techniques, 6th Edition” and “Recording Music On Location.”
Mackie Adds Apple Lightning Connector To DL1608 & DL806 Digital Mixers
Enables direct docking of the iPad (4th generation) and iPad mini (using the new iPad mini tray kit)
Mackie has introduced two new versions of the DL1608 and DL806 digital mixers that include the Apple Lightning connector, enabling direct docking of the iPad (4th generation) and iPad mini (using the new iPad mini tray kit).
While both of these iPad devices currently work with DL mixers wirelessly via Mackie’s Master Fader app, physical docking offers advantages. In addition to device charging, users will have access to 2-channel recording straight to the iPad and the ability to playback audio from nearly any iPad app straight to the DL mixer.
The new DL1608 and DL806 featuring the Apple Lightning connector are currently in production and are shipping to dealers and distributors worldwide.
The Master Fader Control App is available for free from the App Store on iPad or at www.AppStore.com/MackieMasterFaderControl, while the My Fader Control App for iPhone and iPod touch is available for free download from the App Store at www.AppStore.com/MackieMyFaderControl.
Wednesday, September 25, 2013
New SSL Live Console Now Shipping
Three to Britannia Row and two more to SGroup in France
The new Solid State Logic (SSL) Live console, which debuted at Prolight + Sound in April, 2013, is now shipping.
The first three consoles all shipped to U.K.-based global tour production company Britannia Row for use on Peter Gabriel’s forthcoming European ‘Back to Front’ tour, and another two shipped to SGroup in France for the imminent Amel Bent tour.
Console manufacturing production for 2013 has been sold out since July, and details of the new commercial partner network for SSL Live are available in the “Where to Buy” section of the SSL website (here).
SSL CEO Antony David states: “The on-schedule completion of the new Live console is an important milestone for SSL. This has been one of the biggest developments we have undertaken for some time and marks the first application of our new Tempest digital platform. We have been very encouraged by the response from mix engineers, rental companies and our channel partners since we presented the console in April this year. Demand has substantially outstripped our initial production plans, but we will return to reasonable lead times by early 2014.”
Since April, SSL has expanded its dedicated Live product team with key hires including Jason Kelly as Live Consoles product manager based in the U.K. office, and Jay Easley as vice president – Live Consoles to lead SSL’s live sector sales operation in North America.
Certified training courses have also commenced, with focus on commercial partners and initial purchasers. A training program for the wider operator community is scheduled to commence from January, 2014.
The SSL Live will be exhibited at next month’s 135th International AES Convention in New York and at ISE in Amsterdam in February 2014. With its latest offerings, SSL will relocate to Hall 8 at Prolight + Sound in Frankfurt in March 2014 and the company will exhibit at InfoComm for the first time, in Las Vegas in June, 2014.
Solid State Logic
Church Sound: Are You Blinding The Congregation With Your Mix?
Escaping personal mixing bias
My eyes felt like they were as big as dinner plates and any sunlight was blinding. Welcome to having an eye exam. During the exam, your optometrist will insert eyes drops that widen each pupil to the size of a grapefruit. This enables him to check the back of your eyes for signs of diseases and medical conditions.
The problem with the eye drops is they keep your pupils in that WIDE OPEN state for about, let’s see, it’s 3:42 now, so….about six hours. During this time, sunlight is your enemy. It’s your kryptonite. It’s blinding.
Mixing audio, you have a natural bias in your mix style. You might mix bass-heavy. You might mix with a strong tendency towards boosting high frequencies. You might push an instrument a bit too loud a result of your personal preferences. To you, it seems natural but to others, it’s BLINDING!
This isn’t to say all mix biases are bad. I’ve heard song covers that were better than the original. I’ve heard re-mixed songs that brought new energy or a different emotion to a song.
However, it takes a lot of talent to be able to do that and you need an audience accepting of “something different.“Today let’s consider if your mix bias benefits the music or blinds your congregation.
Two Signs Of A Blinding Mix
1. You get regular complaints from different people about the same problem.It’s one thing to get an occasional complaint about volume (overall or instrument-specific) but if you are getting a weekly dose of complaints from different people about the same thing, then you are blinding them with your mix. You must reconsider your mix and remember your purpose in mixing.
2. The congregation doesn’t engage in worship as when someone else mixes. You might think the other tech’s mix is sub-par to yours, but if the congregation is more engaged in worship whenever they’re mixing, then it’s your mix that’s sub-par.
Please know that comparing mixes isn’t a competition. It’s not about who is better, it’s about who is doing what’s best for the congregation, the room, the music, etc.
Are there other signs of a blinding mix? Probably. But these are the two which should give you immediate cause for consideration.
What Can You Do?
1. Take it personal. I mean this in a bad way.
You could think things like, “the congregation doesn’t appreciate the quality of my work” or “I don’t care what the congregation thinks.” Ummm, too bad for you, if you do. You have missed the point of church audio production. If you are at this place, please read this article.
2. Take it personal. I mean this in a good way.
You might mix on weekends for a country band and create a phenomenal mix. But that same mix might not be best for your church congregation. It’s different people, it’s different music, and it’s a different purpose. Accept that you have a mix bias and learn to adjust your mix style for the congregation.
For a month:
—Listen to how the other church techs mix.
—Listen to worship music via iTunes or Spotify (or whatever works for you).
—Listen to how volumes are different.
—Listen to how individual instruments are mixed.
—Listen to how the overall mix is shaped.
—Listen for similarities in how you mix and listen for the differences.
The Take Away
Working behind the mixer, you can be as creative as you wish to be. The problem is you can be blind to your personal mix bias. That, in turn, can be blinding to your congregation. Be aware, mix smart, and let the Holy Spirit be the only thing that shines on your congregation.
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.
Church Sound: A Step-By-Step Process For Optimizing Stage Monitors
“Oh it sounds much better. Very musical and pretty much exactly what I expect to hear.”
A few weeks ago I was having a discussion about the sound quality of our monitors with our sax player.
He’s a very discriminating guy who really knows sound. If he says he needs 185 cut 3 dB, he needs 185 cut 3 dB.
As we listened to his wedge, it was clear that the sound he was putting out of his horn was not being faithfully reproduced by the wedge.
I knew it would take a while to remedy this, but I set aside a day to go through and fix the wedges.
First, here is our signal chain: Each mic gets plugged into a splitter (it’s passive, not transformer isolated, which was a bad choice, but not mine).
One split goes to an M7 in monitor world. The M7 mixes up to seven wedge mixes.
The omni outs of the M7 go to two Klark Teknik 9848 4x8 processors set up in 4x-biamp mode. The KT 9848s feed a rack of QSC amplifiers, which in turn feed EAW SM12 monitors.
Aside from the split, it’s a decent system. It took me a little bit, but I finally got my Mac talking to the 9848s (using Parallels, XP and a RS232-USB converter). I should mention that KT’s tech support was very helpful and quick in getting this running.
Once I had that going, I had a nice, graphical interface with which to adjust the settings. I positioned a wedge in the middle of the stage, and placed our Earthworks M-30 measurement mic right about where a musician would stand.
I took the following approach: When EQ’ing monitors, you really aren’t worried much about the room as it’s really a near-field monitor.
The only real boundary is the stage itself, and my goal was a pretty linear system; that is, flat and set up so that what goes into the board comes out of the monitors.
In the past, I would have started running pink noise through the system and looking at the response on an RTA. But that amount of noise (I measured at 94 dB SPL-A) gets annoying really fast.
I’ve been learning more about more modern forms of measurement including swept tone and FFT, so that’s how I went about this process. I’ve found swept tone gets me a lot closer a lot faster than pink noise, without the grating noise.
I started off with a great little program I found called FuzzMeasure Pro 3 (from SuperMegaUltraGroovy Software, the best software company name ever). FuzzMeasure uses swept sine wave deconvolution to report frequency response.
If that sounds complicated, don’t worry. All you need to do is hook the mic up to a USB interface and press measure.
The software emits a quick click that is used for impulse measurements (great for setting delay, but that’s another post), then a swept sine wave from 20 Hz to 20 kHz (or in my case, 50-17 kHz; it’s user-adjustable).
The resultant frequency response is shown on a graph. Here’s where we started:
(click to enlarge)
Keep in mind that each light grid line is 1 dB. So we started off with the low end some 13 dB below the mid- and upper-range; not so good. I made a 12 dB adjustment on the gain for the low channel (actually I cut 6 off the top and added 6 to the low).
(click to enlarge)
Now we start getting a little closer. But it’s still way off. After spending some time with the parametric EQs built into the KTs, I ended up with a sweep that looked like this:
(click to enlarge)
It might look a bit wonky, but realize that this time, the heavy grid lines are 1 dB. So the response could be called flat, ±1 dB from 80 Hz-17 kHz (the right edge of the graph is 17 kHz, as set in my prefs)—that’s not too shabby.
Now, because I was getting more complaints from some musicians than others, I decided to drag another wedge over and take a measurement.
I was surprised (well, not that surprised) to see a significantly different trace, even with the same EQ settings as the first one.
So I decided to tune each monitor individually. In my new setup, each monitor has a number, and it will always be used with the matching monitor send.
Thus, Speaker 1 will be plugged into Monitor 1 on the patch panel. That ensures that all monitor mixes are basically the same, even though production variations give each wedge a slightly different response curve. I’ve applied custom EQ to each one.
The final step was to tweak it a little closer using another cool program called Spectre from Audiofile Engineering.
Spectre has a a compare trace FFT function which allows you to look at the signal coming out of the board and the signal coming back from the measurement mic at the same time.
With some gentle pushing and pulling of the EQ curves, I was able to get that line almost completely flat. It’s a little easier to see in this view, where the purple is the output of the M7 pink noise generator and the green is what’s coming back from the measurement mic (which is nearly completely flat from 20-20 kHz).
(click to enlarge)
What’s fun about this view is that if you add say, a 8 dB bump at 1 kHz on the output EQ of the monitor mix, both curves show exactly 8 dB of bump, in a bell curve that looks just like the graphic on the EQ display.
I had now reached the point of it being a linear system; what goes in comes out. The next weekend, I asked our sax player how his wedge sounded.
Without telling him what I had done, he commented, “Oh it sounds much better. Very musical and pretty much exactly what I expect to hear.”
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
Wednesday, September 18, 2013
Properly Cleaning Mixing Console Faders
Cleaning a fader is not brain surgery, but it takes practice and a lot of care. Here's how to go about it - successfully.
Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.
I’ve cleaned a lot of faders over the years and suppose I’ve gotten a little bold when it comes to tearing a fader apart and giving it a good bath.
And I’ve learned the hard way just how much punishment a fader can take before it breaks.
In some cases, a certain amount of brute force is required to crack open a fader, but then a certain amount of gentle finesse is needed to clean its individual parts.
I recommend practicing with old junk faders – without experience, it’s all too easy to ruin a good one.
Cleaning a fader is not brain surgery, but it takes practice and a lot of care.
Before we getting into a total fader rebuild, let’s talk about quick cleaning. Much of the time, if the gear has been well cared for and the faders are not too dirty, then a little routine maintenance is all that is likely required.
Besides, keeping faders clean is always a good idea, preventing dirt from becoming embedded deeper inside where it can cause more wear and tear.
Keep in mind: with relatively new faders in particular, do as little as possible in order not to undo the original lubrication. And overall, don’t go any further with this process than you feel you need to.
The first step is to use compressed air to blow as much dirt as possible out of the fader.
Figure 1: Start by blowing one end, and then the other. (All photos by Alex Welti)
There is usually “dust bunnies” in the fader that will come out easily, and this might be all that needs to be achieved in terms of cleaning.
Move the fader carriage to one end and blow air into the slot aiming away from the carriage so that dust can escape through the slot. Then move the carriage to the opposite end and blow air aiming the opposite way.
Skip this step and compound the laziness by spraying some off-the-shelf cleaner-lube into the fader, and it’s likely that the dust bunnies will be matted down and stick in the corners.
Laziness can lead to temporary improvement but later, the dreaded “dust bunnies” in the corner syndrome.
A fader might seem to work better for a while, but this won’t last and might lead to the need for a more substantial (and time consuming) cleaning effort.
Note that the compressed air must be clean and dry.
I do a lot of cleaning, so I’ve invested in a $100 air compressor and then added an air filter / dryer unit for about $40. To this I’ve added a dryer cartridge that contains silica beads for about $5.
If an air compressor isn’t available, cans of aero-duster will work, but they don’t last long.
If the plan is to clean a couple/few consoles, an air compressor is a worthwhile investment, and it helps do the job right because you don’t need to be worried about running out of air.
In addition, the compressor will offer higher pressure.
Most canned air provides about 60 psi, with this dropping as the can is used.
With the compressor, I’m able to set pressure at a consistent 80 psi, which works very well. (And I found out the hard way that 100 psi will blow some faders and switches apart!)
If the initial “blowing out” process didn’t offer the desired results, it’s time to move on to use of chemical contact cleaner.
Figure 2: Contact cleaner outfitted with a nozzle that adds precision and cuts waste.
Some faders have lubricating grease applied by the manufacturer, while others employ a self-lubricating Teflon-type of plastic.
If used sparingly, chemical contact cleaner shouldn’t impact the self-lubricating type, but it will invariably wash away lubricating grease.
The goal is to avoid adding any more lubrication than is absolutely necessary - dust tends to fall away from dry surfaces, but it sticks to oily surfaces.
Figure 3: “Snap together, snap apart.”
After spraying contact cleaner, exercise the fader and then quickly blow out the excess cleaner.
This helps to spread the cleaner over the entire fader surface, while the excess cleaner carries away additional loosened dirt.
I’ve tried several types of contact cleaner since canned Freon was banned from the market.
There are a lot of good choices – my preference is Contact Cleaner II made by Techspray. It’s about $30 per can and worth the price. Note that I also invested another $30 for a screw-on trigger nozzle so that I can be precise and cut waste.
The fader is still feeling a little rough? Time to try a little lubrication. The key word is “little” – use as little as possible.
Did I mention not to use too much lubrication? Third time’s the charm – lubrication collects dust, so don’t overdo it!
Depending on the type of fader, I use a precision dropper to place just a few drops of lubrication in the fader, or give it just a quick squirt.
Exercise the fader and then blow away the excess with compressed air. Again with the compressed air?
Figure 4: The basic parts of a typical fader, and where they’re located.
Seriously, this helps spread the lubrication into a thin film and gets rid of any excess.
I’ve had good results with a spray lubrication called Tefrawn, made by Rawn. It’s Teflon-based and beneficial to the self-lubricating type plastics noted earlier.
Also, it smells like bananas, not that it matters!) Caig also offers products of this type.
Lesson learned the hard way: some oils react with plastic, causing it to break down. If there’s any doubt, test it out on a spare fader first before applying.
Figure 5: Be careful not to damage the wiper, which can ruin the fader.
Also, certain faders use thicker grease that results in a “smoother” feel, and these may actually feel too loose after lubrication.
If this proves bothersome, use silicon or petroleum grease (but not bacon grease!). I’ve found this step to be more trouble than it’s worth - if “feel” is that important, buy new faders.
Time to reiterate: “Air > Cleaner > Air > Lubrication > Air” About 30 seconds of effort for each fader.
Figure 6: Under and around the rails, but don’t touch the carriage.
Some of the more expensive faders are designed to be easily taken apart for cleaning.
If less expensive faders can’t be cleaned using the steps already outlined, it may not be cost-effective to go any further.
Consider replacement, but if it’s an emergency, keep in mind that you’ll be dealing with tiny parts that are easy to break and lose.
A total fader rebuild should take only about 5 to 10 minutes, after going to the trouble of taking apart the console to get to the fader.
If I take a module out for repair, I go ahead and clean its fader at the same time. Otherwise, I do fader rebuilding as part of a larger console-cleaning project.
There are several different types of fader construction. Higher-cost faders are literally a “snap” to take apart; that is, they have a “snap together” design.
A much more pleasant use of a dental pick than usual.
The main parts of a typical fader include the element that carries audio on conductive tracks, the carriage that holds wipers against the tracks, and the rails that guide the carriage.
Be extremely careful with the wipers - they’re easy to damage, and once bent, the fader is toast.
After opening the fader, first blow away the loose dust. There might be dirt wedged in at the point where the carriage and rails meet, so use a dental pick to loosen this up, and then blow it out.
Again, blow the loose dirt out.
Use a strip of clean cloth dipped in isopropyl alcohol to clean the rails, pass the strip under and around each rail.
Gently wipe the surface of the conductive element with a clean cloth dipped in alcohol or contact cleaner. Be gentle, and do NOT go under the carriage with the cloth. This can damage the wipers!
Top it off with just a dab of lubricant. Caution: a little goes a long way!
Apply just a few drops of lubrication to the rails and exercise the fader. Blow away any excess lubrication with and reassemble the fader
And that’s it. With a little practice and patience, anyone can make old faders feel like new again!
Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.
Alex Welti is vice president of research for Creation Audio Labs, a service facility in the southeastern U.S. He served for a decade as service manager of Soundcraft, and prior to that, worked as a technical supervisor for Westlake Audio.
Friday, September 13, 2013
Symetrix Tunes The Audio For Chadstone Shopping Centre
Corporate Initiatives designed and installed a sound reinforcement system centered on a Symetrix SymNet Radius 12x8 Dante network audio DSP, which tunes the system perfectly.
With over five hundred stores, the Chadstone Shopping Centre in Melbourne, Australia is the largest shopping center in the southern hemisphere.
In the run up to the recent holidays, Chadstone’s owners hired Corporate Initiatives of Nunawading, Victoria to install a video wall and sound system.
The video wall would run advertisements for select stores in the shopping center, and the owners wanted audio support to make the advertisements as compelling as possible. The only catch was that the sound had to be focused. It had to be vibrant and punchy right in front of the video wall but it couldn’t interfere with the ambiances of the nearby stores.
Corporate Initiatives designed and installed a sound reinforcement system centered on a Symetrix SymNet Radius 12x8 Dante network audio DSP, which tunes the system perfectly.
The video wall is an NEC 6mm Pixel Pitch LED array. Videro Digital Signage provides content via a redundant pair of MAC Video Playback devices, each with Dante Virtual Soundcards.
Because the Symetrix SymNet Radius 12x8 has two Gigabit Switch ports, both playback devices are connected to the system via the Dante network, and the backup is always on standby.
Additional analog inputs include two Shure wireless microphones and a flexible XLR input plate for use at events in the large public space in front of the video wall.
Robust automatic gain control is a critical component in the success of the installation. The volume of foot traffic – and hence the volume of background noise – varies tremendously depending on the day and time.
An appropriate volume on a Wednesday morning would be completely drowned out on a Saturday afternoon, and an appropriate volume on a Saturday afternoon would be overwhelming on a Wednesday morning.
Two ambient noise-sensing microphones complete the input count, to provide information to the Radius 12x8’s automatic gain control algorithm.
“The Symetrix SymNet Radius 12x8 was the perfect processor for Chadstone’s video wall,” said Michael O’Connor from PAVT. “For an affordable price, the Radius 12x8 provides Symetrix’ broadcast-quality audio path and algorithms, including automatic gain control, and Dante audio capabilities.
“Moreover, our experience has been that Symetrix builds the most reliable DSPs on the market. We have been distributing Symetrix for about ten years now, and in that time we have sold hundreds of Symetrix units for corporate installations, stadiums, broadcast facilities, and nightclubs.”
Because the audio content is delivered from a variety of sources and because the level is not always consistent between those sources, the Symetrix SymNet Radius 12x8 also delivers uniform content level using a number of dynamic processing modules.
Two dual-channel Powersoft K2 DSP+AESOP amplifiers provide biamped power to a pair of EAW QX full-range loudspeakers.
Because the system runs ten hours a day, seven days a week on average, the K2’s Power Factor Correction maximizes the system’s efficiency by drawing only as much power as is actually needed.
The QXs provide precise directivity so as to keep energy focused on potential viewers and away from the nearby shops. An AMX Netlinx Processor and user interface allows intuitive source selection and volume control.
“This difficult audio environment in a large shopping center was a first for us,” said Chris Gauci, director at Corporate Initiatives. “But with the assistance of Production Audio Video Technology, we were able to design and install the perfect audio system within this acoustically complex environment.
“We had to have the right audio level covering just the right public space, without impeding on any of the shop spaces very close by. The carefully-calibrated ambient noise sensing is perfect, and the customer is ecstatic with the final result.”
Posted by Julie Clark on 09/13 at 02:20 PM
Thursday, September 12, 2013
St. Jude The Apostle Catholic Church Thrives With Lectrosonics
Aspen processing and Digital Hybrid Wireless technology combine to deliver exceptional audio performance.
In the fall of 2009, St. Jude the Apostle Catholic Church, located in Baton Rouge, LA, embarked on an ambitious campaign “To Build a Future of Hope!”
Among its various directives was the complete renovation of the church including the installation of a new sound reinforcement system. At the heart of that system is Lectrosonics audio processing and wireless microphone technology.
Technical Services Group (TSG), a design/build firm also located in Baton Rouge, was brought into the project by David Hebert of GHA Architects to install the new equipment.
After meeting with church management to ascertain their expectations, the TSG team ultimately deployed a wireless microphone setup consisting of a Lectrosonics Venue modular receiver mainframe stocked with five VRS receiver modules, along with two HH handheld transmitters, five LMa beltpack transmitters, plus two M152/5P lavaliere microphones and two SNA600 adjustable dipole antennas.
To ensure optimized signal management, the TSG team also equipped the church with a Lectrosonics Aspen SPN1624 (16 input / 24 output) audio processor that was augmented by an SPN16i 16-input expansion unit.
TSG’s Scott Richard, the firm’s AV designer, discussed the nature of the project.
“Services at St. Jude the Apostle Catholic Church are quite traditional,” Scott Richard, TSG AV Designer, explained. “As you can appreciate, the ability of the roughly 2,500 strong congregation to understand clearly what’s being said during a sermon is absolutely essential.
“If the audience can’t clearly understand, it becomes impossible to relate to the subject matter. Further, the audio system’s ability to function largely unattended was equally important and, for this reason, the deployment of the Lectrosonics Aspen processing system—with its automixing and feedback elimination parameters would help ensure trouble-free operation.”
In addition to its automixing and feedback elimination capabilities, the Aspen processing system is also managing the monitor mixing for both the choir and the piano / keyboard station. Overall, the system is managing 32 input channels, all of which was programmed by Jason Martin using the Aspen control software.
Of particular note, the Aspen system offers the ability to be controlled via an Apple iPad tablet.
In this particular application, the iPad provides the ability to manage volume level control for the main loudspeakers as well as for the transept loudspeakers, with each microphone being configured for independent level control.
As for the selection of wireless microphone equipment, Richard notes that Lectrosonics Digital Hybrid Wireless technology was selected because of its superior sonic quality, dropout-free performance, RF agility, robust build quality, and ease of operation.
“We’ve had a great experience with Lectrosonics over the years,” says Richard, “so when it came time to deploy a wireless mic system on this project, there really was no reason to look elsewhere.
“Combined with the company’s excellent customer and technical support services, the selection of Lectrosonics was an easy decision.”
St. Jude the Apostle Catholic Church’s audio system was deployed throughout from April through June with the first service occurring on June 19th. The church dedication service took place later that month on June 30th.
Since that time, Patrick Meek, TSG’s Vice President of Sales and Marketing reports that the entire system has been meeting and exceeding expectations.
“Everything’s been working really well,” he said. “The Pastor, Father Trey, is ecstatic over the positive reaction he’s received from the parishioners as well as his own personal satisfaction with the system.
“St. Jude’s audio system is a first-rate setup and much of its performance attributes are directly related to the quality of the Lectosonics’ audio processing and wireless mic equipment. We are extremely pleased with the final results.”
Technical Services Group
Wednesday, September 11, 2013
CCI Provides Life Church With 360-Degrees Of Mackie
CCI Solutions installs a 360-degree Mackie audio system in multi-purpose your facility.
CCI Solutions has a well-deserved reputation for creative solutions to challenging spaces. So it’s not surprising that the Olympia, WA-based AV design firm is behind the unique installation at Life Church in nearby Oak Harbor, featuring a 360-degree all-Mackie
With multiple weekly services hosting a growing membership approaching 900, Life Church recently completed an expansion project that includes a new multi-purpose youth facility.
Housed in a futuristic looking geodesic dome, the youth center is designed as a circular worship space with integrated audiovisual system and seating for 150.
"This was an unusual space that presented some unique acoustical challenges," explains Rick Boring, Sr. Systems Consultant for CCI Solutions. "The structure had been built several years ago but was never completed, and was uninhabitable.
"The room itself is circular, so every single interior wall is angled. Configuring it with traditional theater seating would have presented some coverage problems, and the need to go with a curved video screen, which would be expensive."
CCI's simple and effective solution was to design a space with 360-degree seating. A small, slightly elevated stage at the room's hub is built around a single center truss that holds four Mackie HD series loudspeakers and four LCD displays.
"The youth pastor conducts the service from the center stage, while the rest of the band is on another small stage an alcove off to one side," says Boring. "And depending on how the room is configured, the band alcove can sometimes wind up behind the seating area."
"Placing the speakers in the center was the easiest and most cost-effective way to ensure good, consistent coverage," Boring adds. "Plus, everyone is watching the worship leader on the center stage, and reading the text on the screens above him. Having the sound emanate from the middle of the room simply makes the most sense."
The main loudspeaker cluster consists of four Mackie HD1531 powered 15-inch 3-way systems. The speakers are mounted vertically above the LCD displays and angled down toward the seating areas.
"The Mackie HD1531 offers great sound quality and versatility for the money," Boring remarked. "With a challenging job like this, where space and budget are important, the HD1531 delivers a tremendous amount of clean output and bandwidth.
"So when the kids want to rock they can do it no problem.They're fully powered, so we didn't have to worry about outboard amps, and they've got more than enough low end for this space, so they don't need a sub."
For mixing duties Boring specified a Mackie DL1608 digital live mixer with iPad control. In addition to the band, inputs include a wireless microphone system for the youth pastor, as well as a Denon recording and playback system.
The Mackie DL1608 provides outputs for mains as well as monitors, and handles 100% of the system DSP.
"The DL1608 is absolutely perfect for a job like this," says Boring. "It's really simple to use, has excellent processing power already onboard, and packs plenty of I/O for its size."
The "keep it simple" mindset extends beyond all things audio to the video system as well. Beneath the four Mackie HD series loudspeakers are four 65-inch LCD displays.
Signal is provided by a Mac Mini computer running ProPresenter software. Four separate Xbox 360 consoles are each coupled to the four screens for rowdy, eye-to-eye multi-player sessions.
While technology may be their passion, the team at CCI Solutions will ultimately cite customer satisfaction as their highest priority. For Life Church, Boring is especially pleased with the finished product.
"We care about these churches and we want to help them communicate their message," concludes Boring. "I worked very closely with Life Church's Sr. Pastor from the very beginning of this project. He's very pleased with the final result and that's exactly what we strive for."