.(JavaScript must be enabled to view this email address)
Article
Friday, May 25, 2012
Solving A Conundrum: Deployment Of A New Subwoofer Configuration
Any reservations were quickly forgotten once we heard the rig...
I’ve spent the past several years experimenting with various types of subwoofer arrays and have arrived at the same conclusion each time: with each method, something is gained while something else is lost.
Left-right placement enhances coverage on the sides at the expense of drastic power alleys and cancellation zones.
Horizontal arrays produce even coverage in front of the array at the expense of a major drop-off on the sides.
With delayed arcing, side coverage is improved, at the expense of tightness.
With cardioid “front-back-front” setups, tightness is enhance at the expense of output SPL.
I could go on, but you get the idea.
Sub deployment is especially tricky in arenas, where the large open area and hard (usually concrete) surfaces makes for the addition of low-end “mud” in the diffused field, along with the need to cover a wide area in the horizontal plain – usually 180 degrees or more.
One of the more common practices I see in the field is flown left-right sub arrays, where hangs of usually eight or more boxes are flown either just outside of or behind the main PA hangs. While the throw is fantastic, comb filtering from left to right is a mathematical certainty.
My choice has been a horizontal array, where boxes are placed along the front of the deck and spaced evenly at a quarter-wavelength of the target frequency. The coverage on the floor is extremely solid and even, but it’s virtually non-existent elsewhere.
I wrestled with this problem last year with a system provided for an Easter Sunday worship service presented by Austin Stone Community Church at the Frank Erwin Center on the campus of the University of Texas in Austin.
While the primary function of the Erwin Center is hosting college basketball games, it’s also booked solid most of the year with A-list tours that draw 15,000-17,000 in attendance. The floor is elongated, stretching 110 feet from the deck to front of house, while the upper levels begin to widen, until the arena becomes almost perfectly round at the top of the mezzanine.
The scene on Easter Sunday at the Erwin Center—click to enlarge. (Credit: Scott Wade)
My typical horizontal array was less than ideal for this venue. While the floor sounded (and felt) fantastic, you only had to climb a few rows off to the sides before the low-end began to drop off quickly. While some of this can be corrected by delayed arcing (delaying the outside boxes to create a virtual arc), the added difference in arrival times causes the entire array to lose much of its tightness due to smearing in the time domain.
I left that gig determined to find a solution. There had to be a better way to get even coverage, both in front and on the sides, without power alleys or comb filtering. In an ideal situation, the low-end should come from a single source, equidistant from the main left-right hangs. But where would it be located?
NEW DIRECTION
This past Easter Sunday, when the church returned to the arena, I had the unique opportunity to try a different approach. Working with local audio providers Big House Sound, we designed and deployed a large-scale, flown, end-fire subwoofer array. This had yet to be tested, but once the rig was in the air, jaws began to drop.
Part of the idea came from fellow engineer/tech Sarah Butt, who had worked under Kenny Chesney system engineer John Mills for one of the “Goin’ Coastal” tour dates at Cowboys Stadium in Arlington. Mills’ design for a large Electro-Voice X-Line rig incorporated two monstrous hangs of subs (20 per hang) flown dead center of down stage, between the mains. The theory made sense – two hangs acting as a single source, dead center of the rig, with no cancellations or drop-offs anywhere in the venue. This got my brain spinning.
At left, a screen shot of the system modeling; at right, polar data of the “EF6”6 end-fire array, based on a ground stack two cabinets high—click to enlarge. (Credit: Adamson)
There were four primary issues to address in order for my approach to be a success. First, it had to be cost-effective. This is already an expensive gig, and I couldn’t afford to add a lot more boxes. Second, we needed to take advantage of the 90-degree off-axis null directly beneath a vertically flown sub array to reduce the back-pressure on stage. Third, the arrays needed to be short, since longer arrays reduce vertical coverage by making the hang more directional. They needed to behave as a point source.
Finally, we needed it to be cardioid. I’ve found that the boomy low-end in arenas typically results in me pushing the subs harder to get that tight “feel” I’m looking for. On the flip side, I tend to use less low-end in my mixes with cardioid subs, attaining a clean, tight feel at lower levels when I’m not dealing with a lot of late reflections from all of that backfiring energy.
PLAYING IT OUT
I decided on two arrays, each comprised of six Adamson Systems T21 subwoofers, one behind the other in an end-fire formation.
With significantly more output power than most dual-18-in subs on the market, it required less of them to get the desired SPL, and also kept the hangs short.
Four additional T21s were spread across the floor to help fill in some of the gap before the flown subs took over.
The main hangs consisted of 12 Adamson Y18s per side, with four Y10 for down fill and 12 Y10s for out hangs.
The rig was powered by 30 Lab.Gruppen PLM 10000Q amplifiers, with all system DSP done inside the amps.
Determining the exact end-fire spacing to optimize gain, spacing and delay time was a joint effort involving Adamson and it’s distributor in France, DV2. The result was a textbook cardioid pattern from 40 Hz to 80 Hz, where the subs were crossed over. The “EF66” preset, as Adamson called it, spaced the two hangs 66 inches apart around a center frequency of 51 Hz.
To prevent the rig from swinging (and hence changing the precise spacing), a piece of steel pipe was linked between the rear pick point of the downstage hang and the front point of the upstage hang. Cheeseborough clamps allowed us to adjust spacing once the rig was floating.
A look at the centrally located end-fire array, as well as some of the mains—click to enlarge. (Credit: Cody Hester)
Finally, a cable bridge constructed of minibeam spanned from the sub hang downstage center to the stage left cable pick, in order to reduce added tow on the hang.
MISSION ACCOMPLISHED
Any reservations were quickly forgotten once we heard the rig. I was afraid that the flown subs might lose some of their “punch,” but that certainly wasn’t the case here. The low-end was extremely tight, even and clean. There was no more than 7 dB of drop-off from the first row to the top of the mezzanine, and more importantly, the ratio of subs-to-mains remained constant, leaving the mix to sound full and clear, no matter where you were located.
Behind the rig, the cardioid pattern performed beautifully. The sections behind the stage were so quiet it almost felt eerie. In the past four years working with Big House, I’ve heard this rig deployed dozens of times, and became used to the large amounts of back pressure these subs typically produce – but not this time. It was practically silent, except for the late 1.5-second reflections coming back from behind front of house.
When all was said and done, we couldn’t have been more pleased with the outcome. It was hands-down the smoothest, tightest and most even low-end I’ve ever gotten out of an arena system. The comments about the superb low-end coverage haven’t stopped coming in, and one production manager told us that it sounded better than the rig a major concert tour brought in just a few weeks prior. Whether or not that statement is true, I don’t know, but I’ll take the compliment nonetheless!
The Easter Sunday sound crew at the Erwin Center, left to right: Mark May (system tech), Sarah Butt (A2), Todd Hartmann (front of house, designer), Fiona Cheung, (patch), Mateo Rodriguez (monitor engineer), and Cody Hester (A3)—click to enlarge. (Credit: Scott Wade)
I fully believe in giving credit where credit is due, especially in this industry. We’re fortunate to benefit from some pretty great minds, using their ideas to take things to a new level. I have tremendous respect for people like Dave Rat, who consistently put new concepts out there simply because they’re passionate about live audio, and want to make us all better as a whole.
We are, in a sense, standing on the shoulders of giants. I truly look forward to what future discoveries will be made and how they will continue to improve one of the coolest fields a person could find themselves employed in. Todd Hartmann is the audio engineering coordinator for the Austin Stone Community Church as well as A1 systems engineer for Big House Sound in Austin, TX.
CADAC Debuts CDC eight Production Platform To Theatre Sound Professionals In U.S.
CADAC demonstrated its new CDC eight digital production console to the U.S. theatre sound market as a sponsor of the Live Design Broadway Sound Master Classes.
The event also marked the first showing of the new 32-fader frame variant in the U.S.
CADAC U.S. theatre sound distributor RF Productions of NY showed the new console to theatre design professionals and students during manufacturers’ showcase sessions earlier this month in the intimate confines of the 99-seat Abe Burrows Theatre at NYU.
CADAC’s participation is part of the New Technology Presentation, with Tom Bensen of RF Production and CADAC International sales manager Ben Millson in attendance.
While the launch of the CDC eight has focused on the console’s critical role in spearheading CADAC’s move into the wider sound touring and installation sectors, the CDC eight’s extensive configurability and expansive feature set provides CADAC a uniquely competitive, high performance digital offering for theatre sound applications.
Critically for the theatre market, the CDC eight supports a uniquely high input count on a desk with such a compact footprint; and at a cost that makes it affordable.
Feature-wise and sonically, the CDC eight compares favorably with a CADAC J-Type. It has 40 configurable output buses which can be programmed as groups or matrixes, and additionally there are 16 internal VCA groups. The console offers up to 128 input channels, and using the input selection on each channel to sum two inputs together, that can be expanded to a potential 256.
Appropriate levels of operability is a big issue with digital consoles in theatre sound and the CDC eight offers an advance on that front, with its “high agility” user interface with two 24 inch 16:10 HD touch-screen control surfaces; a third smaller LCD touch-screen is dedicated to system control and automation.
Commenting on the ongoing CDC eight development program, CADAC senior development manager Peter Hearl states, “The CDC eight was devised as a console to address the widest possible range of applications to the highest standards of performance and with CADAC’s heritage, theatre sound was always going to be an important market for this console.
“The theatre development programme for the desk is addressing a number of issues and applications, including looking to the potential compatibility of the CDC eight with the J-Type SAM theatre automation software, enabling the recall of all SAM settings on the new console. The program objective is to make the CDC eight a great CADAC theatre sound desk.”
The Live Design Broadway Sound Master Classes provided CADAC the opportunity to address an audience of dedicated theatre sound professionals, but most importantly it offered them quality time with the industry’s top designers in a relaxed, intimate setting. This year’s group of speakers included such Broadway vets as Mick Potter Nevin Steinberg, Kai Harada, Ken Travis, Jill DuBoff and Abe Jacob.
Two Midas PRO6 Live Audio Systems Deployed At Premier Casino du Liban
Two Midas PRO6 live audio systems supplied by distributor of the year, Dubai-based NMK Electronics, have been installed in one of the Middle East’s premier casinos, Jounieh’s Casino du Liban, bringing the venue’s total PRO6 count to three.
Tony Khoury, the casino’s chief sound engineer since 1996, took delivery of his first PRO6 system in 2009 for the 1,800-capacity theatre space.
“I chose the PRO6 then because I couldn’t find the right warmth of sound in any other digital desk,” says Khoury. “The MIDAS pre amps in the PRO6 provide really effective power and quality. I also found the PRO6 so easy to use; you can get to any zone really fast – it feels like I am working on one of the MIDAS analogue consoles, like a Heritage 3000.”
Khoury was so pleased with the performance of his first PRO6 that when it came to upgrading the sound system in the casino’s dining and entertainment space, he chose it again for the 600-capacity La Salle Des Ambassadeurs, with one PRO6 for front of house and another for monitors.
A DL431 mic splitter has been added to the monitor system while both monitor and FOH consoles have DSP engines, DL252 digital snakes to optimize mic audio and a DL451 audio system modular I/O with both analog and digital cards to maximize flexibility.
“Having had the PRO6 in the theatre for three years, I insisted on MIDAS again,” says Khoury. “I am very satisfied with the PRO6 and I am happy to be the only sound engineer in Lebanon to have three in one place.”
Khoury has now mixed hundreds of shows on his first PRO6, from live bands to musicals and dance performances to stand up comedy, a variety which has showcased the system’s flexibility. “I’ve used almost all the effects and dynamics, I like the channel inputs and the EQ is very precise and sensitive,” he notes. “The choice of four compressors on each input channel provides nice gain reduction and the gate is also very helpful where there are many instruments miked up. I also like the Klark Teknik graphic EQ and delay effects.”
Says Chicco Hiranandani, NMK business development manager, “I had a great time working together with Tony on the recent upgrade of the Salle des Ambassadors at the casino. It is always enjoyable working with individuals who share the same kind of intensity or passion for what they do. I am sure that patrons of the casino will experience a spectacular show or event with the new equipment at the venue.”
Midas brand development manager Richard Ferriday adds, “We are delighted that Casino Du Liban are continuing their investment in MIDAS, and our new competitive pricing now makes that even easier. Another important consideration is that the PR06 is a modular, upgradable system, which means that the casino has the option to upgrade to full 88 input PR09 capacity if their requirements increase in the future.”
Soundcheck, one of the Philippines’ top audio contractors, has handled live sound for artists such as Rihanna, Chris Brown, Incubus, Beyoncé, Linkin Park, Christina Aguilera and many others.
The company has utilized JBL Professional VerTec line arrays since their introduction, and just upgraded to new JBL V5 VerTec preset tunings.
“We installed the V5 DSP presets in all of the VerTec VT4888 and VT4888DP line arrays in our inventory, in the speakers’ DrivePack DPDA input modules,” says Jaime Godinez, owner and chief engineer of Soundcheck. “After installing the V5 upgrade and testing the speakers by running pre-recorded music for them for hours, we sent them out for a major local show right away. We went through the show without a hitch—and have since done more than 60 shows with them including a few large open-air concerts.”
“When JBL introduced the V4 presets a few years ago I felt that they had given new life to the VerTec system. But the improvement offered by the V5 is even more significant,” Godinez continues. “The improvement in clarity, balance and coverage have made me fall in love with our audio systems all over again. But most importantly, my clients are noticing the difference between our sound and that of the competition.”
“No comments I can give can adequately convey the experience of actually hearing the sonic improvements that V5 provides, especially in an open-air environment,” Godinez adds. “I’ll try by giving one example. Last March I was mixing the Coca-Cola Concert ng Bayan at the SM Mall of Asia, which celebrated 100 years of Coca-Cola in the Philippines. The concert featured around 20 performers and I brought 32 VerTec VT4888’s for the main front of house.”
“The sound was so transparent and ‘in your face’ that many times during the night I had to double-check to see whether my near-field monitors were off or on, as the sound from the VerTec arrays seemed to be coming from that close.”
Godinez pointed out that he has to add more gain to the VT4888’s using V5 to keep some mixing consoles from being driven too hot. “It’s just something you have to pay attention to,” he says. “With V5 you might have to make an adjustment to the gain of the DriveRack modules compared to the settings you’d use with the V4, to make sure the gain between the board, speakers and the rest of the system is optimized.”
“Another big benefit we found is that using the V5 presets enables us to set the systems up faster,” Godinez concludes.
Turbosound Flex Array For Evangelistic Temple In Nassau, Bahamas
In the summer of 2010, Jeff Cameron of Encore Broadcast Equipment Sales began communications between STS, Atlantic Professional Audio’s installation division, and the Evangelistic Temple in Nassau, Bahamas.
Encore and STS have collaborated on numerous occasions where a performance audio installation is needed.
Later that year, Cameron brought Craig Beyrooti, CEO of Atlantic Professional Audio, to Nassau for a site visit with lead pastor Gary Curry. The temple would be finishing construction of a new building over the next year and was in need of performance audio design-build services.
After determining and understanding the temple’s vision, STS presented a complete solution including pictorial schematics, fully explaining the install proposal and after a couple of visits the design was complete.
Beyrooti saw the temple’s 8,500-plus square-foot sanctuary as an good fit for a Turbosound Flex Array loudspeaker system based on the architecture of the room and style of worship.
“In a modern church with a praise & worship band and a lot of spoken word Flex Array is an excellent choice because of its smooth vocal projection and response. The sound character remains constant irrespective of level. Loud or soft, the rig sounds fantastic,” Beyrooti says.
Head of STS Michael Ramey and his crew flew to Nassau to install the sizable sound system.
The final agreed design consisted of six Flex Array TFA-600H mid/highs per side, two NuQ-12s for outfills, six NuQ-6s across the lip of the stage, and seven NuQ-8s for under balcony fills hung 50 feet from the PA. Four TSW-218 subwoofers completed the system, which was powered by twelve Turbosound RACKDP-50 amplifiers.
While all the flown speakers were finished in white to compliment the sanctuary walls and ceiling, Turbosound pulled its legendary TurboBlue speaker paint out of storage so that the front fills and monitors would match the church’s blue carpeted platform. Special thanks to Florida’s Turbosound representatives Clinton Muntean and Michael Palmer for their assistance.
In a further customization of the system, STS installed a 94-loudspeaker distro system which provided audio to surrounding hallways, bathrooms, support offices and nursery.
By December, the install had been completed and Beyrooti, together with Darryl Phillips, joined Ramey in Nassau a final time in order to tune and the room. Phillips, who mixes in excess of 200 shows per year for APA’s event division, stayed on a few extra days and helped Media Director Charles Burrows II with sound checks.
Finally, all the work paid off and ET members got to utilize their facility at the first service in early December 2011. Ramey and Phillips stayed to enjoy it and ensured everything went smoothly. Afterwards, STS was invited to sit with the church board and elders for lunch. Everyone was extremely thankful and pleased.
“Atlantic Pro Audio has exceeded my expectations. From the start of the project, there was a high level of professionalism,” says Burrows. “Atlantic Pro Audio went above and beyond to accommodate us; their service was exceptional. Once the project started, I felt secure that it would be handled properly. The audio system is excellent; its clarity and intelligibility are very high. It sounds natural, fills the hall evenly and a full sounding low end rounds it off nicely.”
Meyer Sound has announced the company’s first live webinars on AVB (audio video bridging), the non-proprietary IEEE AV networking standards built upon the ubiquitous Ethernet protocols.
The newest addition to the Meyer Sound worldwide education program, these webinars will be led by John McMahon, executive director, digital products.
In these online sessions, McMahon will give a beginner’s introduction to the AVB standards, show how AVB can simplify AV network design and implementation, and explain the role of the AVnu Alliance certification program.
Meyer Sound is one of more than 40 members in the AVnu Alliance, an industry consortium formed by silicon, automotive, and pro AV manufacturers working together to define AVB and certify AVB devices for interoperability. To date, two AVB-capable products have been announced in the Meyer Sound product line, including the CAL™ column array loudspeaker and the D-Mitri digital audio platform.
Attendees of InfoComm 2012 in Las Vegas interested in AVB are also invited to participate in the education programs at the AVnu Alliance Pavilion. The presentations will feature an interoperability demo with devices from Meyer Sound and 14 other member manufacturers including Avid, Biamp, Harman, Marvell, Sennheiser, and Yamaha. More info is at http://www.avnu.org/news_and_events/events/infocomm_2012.
Rane Introducing HAL2 And HAL3 DSP Devices At 2012 InfoComm
At the upcoming 2012 InfoComm show in Las Vegas, Rane is introducing the HAL2 and HAL3 DSP devices, which expand HAL hardware applications, along with new RADs (wall-mount remote audio devices performing A/D and D/A using Cat-5 cable), DRs (wall-mount digital remotes controlling audio but not passing audio) and Halogen software features.
The new Halogen software release adds Ethernet control support for third-party control systems. Example programs for AMX, Crestron and Stardraw Control makes programming easy.
HAL and Halogen are designed specifically for the installation market due to being tailored for room combining and paging/distributed audio systems.
Halogen software simplifies programming tasks, making it easier to set up multizone paging systems or complex room combining.
Halogen teams with HAL to automatically check the status, location, Cat-5 wiring integrity, and checks that audio is flowing in all peripheral devices. RAD I/O devices and DR remotes extend HAL’s reach well beyond the equipment room and provide for ultimate flexibility as well as providing isolation from grounding problems and electromagnetic interference.
Rane will be at booth C-11316 at the InfoComm show.
Microfiles: Univox DF-14B, A Smooth (Looking) Operator
“Compare its quality…its good looks…”
If you were a musician in the 1970s or are a fan of vintage gear, the name Univox should be familiar.
Merson Musical Products, a musical instrument division of Unicord Incorporated, made and marketed a wide range of products with the Univox brand, including guitars, keyboards and cool-looking blue Tolex-covered guitar and bass amps.
In addition, Merson Musical Products was the U.S. importer of Marshall amps, Korg keyboards and other lines including Tempro brand drums (my first kit).
Some big acts of the day, including The Doors, Led Zeppelin and Billy Preston, endorsed Univox gear, but it never really caught on with professionals.
The company was purchased by Korg in the mid 1980s, and the brand was retired.
The Univox line also included some PA gear, such as the LEM-2 PA system, which consisted of the LEM-XI matrix mixer (tape echo unit, 6-channel mixer and 150-watt power amplifier all in one package) and two horn-loaded column loudspeakers loaded with four 12-inch transducers. Other offerings included the Band-Aid 7-band graphic EQ, and of course, a few microphones.
My first exposure to a Univox mic was through a friend in high school who owned a DF-56, dubbed “The Cheep” in company advertising. That entry-level handheld mic served him well during his garage band days. I didn’t see another Univox mic for many years until, one day at an estate sale, I spotted this beauty, a DF-14B dynamic microphone still with its box.
The DF-14B and cable in the original box. (click to enlarge)
Sporting smooth lines reminiscent of “Art Deco” styling, the DF-14B body and casing is made of polished die cast aluminum. It’s outfitted with a recessed multi-impedance switch on the rear allowing a choice of 50K ohms or 10K ohms high impedance, or 600 ohms low impedance. It also includes an on/off switch.
There were at least two versions offered over the years, the earlier version DF-14 and the later DF-14B. I don’t have specs on the DF-14, but I’ve seen photos, and it looks the same as the “B” version except it didn’t include an on/off switch.
One catalog photo that I have of the DF-14B shows a different logo on the front, leading me to believe that this mic, like many others that were manufactured in Japan at the time, was simply “re-badged” for Univox by changing the name on the faceplate.
The recessed multi-impedance switch and 3-pin connector. (click to enlarge)
A dealer memo dated December 23, 1965 states: “Compare its quality…its good looks…its performance with other mikes selling for $85.00. You’ll find the DF-14B as good. Yet, we’ve priced it at $45.00 – with extra profit for the dealer.” While the list price may have been $45, a catalog ad from 1969 shows the microphone was selling for $32.50 complete with a “heavy-duty reinforced shielded cable with phone plug.”
The console end of the cable terminates in a standard connector, but the mic end does not. It uses a 3-pin connector similar to the one found on a few Aiwa microphones from the period, yet while the pin layout and spacing is the same, the pin diameter on the Univox is thicker, making the plugs non-compatible.
Even though it’s not an expensive or particularly rare model, the DF-14B is definitely one of the favorites in my collection because of its unique boxy look and the heritage of the Univox name.
The slim profile of the DF-14B as seen from the side. (click to enlarge)
Univox DF-14B Specs
Transducer Type: Non-metallic diaphragm dynamic
Polar Pattern: Omnidirectional
Frequency Response: 50 Hz – 15 kHz =/- 5 dB
Sensitivity: -45 dB @ 50K or 10K ohms, -55 dB @ 600 ohms
Nominal Impedance: Switchable 600, 10K or 50K ohms
Size: 5 X 2.75 inches
Net Weight: 29 ounces
Price for a new one in 1969: $32.50
Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb.
Top Concerns Faced By Churches When Purchasing New Sound Systems & Equipment
A load of salient advice...
The Church Sound Forum here on ProSoundWeb is a free, on-line resource of information and dialog for individuals working with sound at their church that is moderated by Tom Young, a highly respected A/V consultant with decades of experience. To participate, go here.
Here, we offer an interesting discussion thread from the forum packed with useful advise from the community.
Question Posted By Kevin
What would you say is the number one concern that a church may go through when purchasing new equipment?
Reply By Dan
People who don’t know what they’re doing and have some misguided notion about what a system should do, cost and/or look like. If you take care of the people problems, everything else will fall into place.
Reply By Dave
Sad, but true. Another main concern is trusting the company who is advising them, as many churches have little expertise in the field of sound & lighting.
I know my church has been bitten once or twice by people who really know their stuff but have recommended things that are cheap and easy to install, which didn’t do the job or solve the problem.
These mistakes can be expensive.
Reply By Bob
Clearly defining what they want the system to do now, six months from now, and two to three years later. Then comes BUDGET. For some reason, budget seems to be set first, before the system requirements are spelled out.
Usually it’s better to wait until funds can be raised to meet the system requirements than to buy what is in the budget now.
Reply By Mac
Thank you. This is the heart of the matter. It’s so frustrating to see all the posts (on the forum) saying we have “X” amount of money, so what should we buy?
Another issue that is almost always ignored is the cost of design and installation. Spending all your money on a bunch of gear - no matter how good the gear - does not make a good system.
Reply By Clark
I can attest to that. We’re in the middle of a renovation. The first budgetary numbers is what they always come back to, forgetting that they added $30,000 worth of stuff we “can’t do without.”
Reply By Ivan
I would agree to some point, however, spending a lot on an improperly designed (over-designed in some areas and totally missing the most important issues) is something I see quite often. In many cases, the customer could have spent less and ended up with a better performing system.
This is not to say that you can do it cheap, I just see way too often sound systems that (as designed) do not have proper coverage, enough gain before feedback and etc., but are full of almost useless “fluff” that drives the cost way up.
People can’t hear it properly, but they have lots of “cool expensive” toys to play with.
Reply By Pacman
I think the number one concern is proper planning (long term - think five-plus years out) and deciding before purchasing exactly what you want the system/room to do.
In our case, we originally designed the system (11 years old now) for a typical service, which meant a choir, a couple of singers, a piano, perhaps an organ/keyboard for music, and a pastor speaking.
This allowed our contractor to determine what we needed, and the architect also designed the platform to “amplify” acoustic sound with a baffled ceiling (to help the choir sound bigger).
Now we’ve changed in 11 years (as many churches do), running a full band with a small choir - totally different than designed, and we’re paying a price.
We are looking into what we can do to redesign, but if we had looked further out than a couple of years when planning, we would have installed a different system and saved ourselves the issues we have now.
Reply By Willy
If possible, visit other local churches that have the same type of worship environment. Maybe even call ahead of time and speak to someone about the satisfaction level with their current system.
While there, take a good look at the shape of the space. Listen intently as you sit, and if possible, move to different locations or have some other people with you to sit in different places.
After the service is over, make arrangements to talk to the person who dealt with the sound contractor responsible for the system. If you liked what you saw and heard, you’ll have a good head start in the right direction.
I know quite a bit about sound and connections and how things work, but I would not even want to attempt designing something that works in a big, complicated space. It truly takes an expert.
The acoustics are probably the number factor for getting good sound. If this is overlooked, the chances are slim of getting something that functions well.
Reply By CR
We had a set of consultants that we let go, not due to technical knowledge but due to their inability to understand what we were trying to achieve. They were very nice people, they knew their stuff for sure, but they just couldn’t get the church leaders here to “buy into” their vision for our equipment needs.
The second group of consultants spoke our leaders’ “language” and embraced our church’s vision better. We immediately hired them and moved forward.
From my standpoint as the tech director, both companies would give us a quality system, the difference is in the paradigm used.
As a customer, my job was to thoroughly examine what and how we do things here, and communicate that as precisely as possible to the design people.
A gear list isn’t going to do that, you need to try to understand your church’s “culture” at a very deep level. I spent much more time trying to do this than obsessing over which loudspeakers/mixers/mics and etc. that we would need.
The root purpose drives everything else tied to it.
Click here to go directly to the Church Sound Forum.
One of the comments I received asked about more information on just how to do that, so here it is.
Timing the effects to the track means that all of the delays and reverb parameters are timed so that they match the tempo of the song so they pulse with it.
As a result, the effect isn’t noticed as much (sometimes not at all), but make the track sound bigger with a bit of an ambient sheen that doesn’t push the track too far back in the mix.
First of all, the absolute easiest way determine the timing is with my Delay Genie iPhone app (that’s its icon above/left) which will easily determine the tempo of the song if you don’t know it, and give you all of the possible combinations that you need. If you don’t have an iPhone or just want a bit more in-depth explanation, here’s some info taken from The Mixing Engineer’s Handbook and the upcoming Audio Mixing Bootcamp.
Delays are measured tempo-wise using musical notes in relation to the tempo of the track. In other words, if the song has a tempo of 120 beats per minute (bpm), then the length of time it takes a quarter note to play would be 1/2 second (60 seconds ÷ 120 bpm = .5 seconds).
Therefore a quarter note delay should be .5 seconds or 500 milliseconds (.5 X 1000 ms per second) which is how almost all delay devices are calibrated.
But 500 ms might be too long and just sound confusing in the mix.
Divide that in half for an 1/8th note delay (500 ms ÷ 2 = 250 ms). Divide in half again for a 1/16th note delay (250 ms ÷ 2 = 125 ms).
Divide again for a 1/32nd note delay (125 ÷ 2 = 62.5 ms or rounded down to 62 to keep it even).
That still might not be short enough for you so divide again for 1/64th note (62 ÷ 2 = 31).
Again this might not be short enough, so divide again for a 1/128th note (31 ms ÷ 2 = 15.625 rounded up to 16 ms).
And yet this still might not be short enough so divide again for a 1/256th note if there is such a thing (16 ms ÷ 2 = 8 ms).
Now such small increments like 8 and 16 ms might not seem like much, but they’re used all the time to make a sound bigger and wider. It’s something that you might not exactly hear, but you can perceive it since it acts as the critical “first reflection,” which is the loudest and most important echo of a sound in any environment. Even a short delay like this will fit much more smoothly into the track if it’s timed.
Another way to determine the delay time is to use the following formula:
60,000 (the number of milliseconds in a minute) ÷ Song Tempo in bpm = Quarter Note Delay In Milliseconds
Example 60,000 ÷ 128bpm = 468.75 milliseconds (rounded down to 468 to keep it an even number).
All the other values can be determined from this by either:
—Dividing by 2 for lower denominations (i.e 468 ÷ 2 = 234 ms for 8th note delay, 234ms ÷ 2 = 117 16th note delay, 58.5 32nd note delay, 29ms 64th note delay)
—Multiplying any of the above by 1.5 for dotted values (i.e. 234 ms x 1.5 = 351ms for dotted 8th note)
—Multiplying any of the above by .667 for triplet values. (i.e. 234ms x .667 = 156ms 8th note triplet)
Dotted and triplet values are very effective delay settings and many times take precedence over straight note delays since they have an interesting feel, providing movement to the part in a subtle way. Plus they fall in between the “can be heard” and “can’t be heard” crack. In other words, they’re noticeable without sticking out like an untimed delay.
It’s also an interesting effect to sometimes use a stereo delay with a straight delay of a 1/4, 1/8th, or 1/16th note on one side and a dotted note or triplet on the other. If the delays are under 100 ms or so, it simulates the sound of a room. During the early 1980s and 90s, a delay of around 25 ms on one side and around 50 ms on the other was used to enhance the sound of a clean electric guitar, for instance.
While we’re mostly talking about delays, timing also applies to things like predelay and decay time on reverbs. Use the above formulas for the predelay, but to time the decay time, use the snare drum to set off the reverb, then set the decay so it lasts just until the next snare drum hit or the one after that.
Remember that you don’t always want an effect timed to the track. Sometimes a delay that’s not timed will stick out, but it might be appropriate for the song. As in all cases with mixing, the right effect or effect parameter is what works for the song.
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 blog.
Allen & Heath To Host iLive/GLD Digital Mixing Education Center At InfoComm
Allen & Heath will be featuring free hands-on live training sessions at the upcoming InfoComm 2012 show in Las Vegas.
Sessions will alternate between the full featured iLive digital mixing systems and the new, compact and scalable GLD system, along with daily product giveaways.
Classroom sessions will cover topics including basic digital mixing fundamentals, integration of the iLive digital mixing system, Waves Multi-Rack and Waves Sound Grid, Dante and the iLive digital mixing system, GLD overview, and hands-on mixing and virtual sound check.
The company will have a hands on approach to these classes including multiple Allen & Heath digital consoles networked with multi-track source media for demonstration of virtual sound check.
Allen & Heath will be located at InfoComm booth # C10123. Go here for more info.
As a founding member of the newly formed Event Safety Alliance, TOMCAT continues to be heavily invested in the advancement of safety initiatives in the entertainment industry.
TOMCAT’s Keith Bohn recently attended the Tour Link and Pollstar Conferences and participated in the unveiling of this initiative to the Entertainment Industry community. The group contains a unique cross section of consultants, equipment providers, free lance labor, artist management, production management and insurance professionals as members, who all have an interest in improving live event safety.
The potential reach and impact of this group is significant and holds great promise to provide information and guidance to all stakeholders in the live event industry.
In April, the founding members of the Event Safety Alliance, including Bohn, were invited to meet with Indiana state officials regarding their legislative efforts to prevent future outdoor event injuries. Indiana has recognized the importance of engaging the experts on the topic prior to passing sweeping legislation and have asked for additional assistance in the coming months in drafting more permanent legislation.
In addition to the ANSI standards that are developed by PLASA, a guideline of best practices is in the works by the ESA. The vision of the Event Safety Alliance, as noted on its website is “To unite the live event industry for the purpose of assembling key technical and production information which can be used to establish formal guidelines to improve safety at live events across the United States.”
The STK restaurant concept has opened haute steakhouses in major cities around the country, and now its second Manhattan location occupies the ground floor of the borough’s famed W.R. Grace Building.
Music plays a vital role in STK’s contemporary vibe, with a custom playlist drawing on classic rock, dance, and everything in between gives way to a live DJ on most evenings.
New York City’s EL Media Group designed and installed STK’s sound system around two Symetrix Jupiter 8 turnkey processors with Symetrix ARC-2e wall panel remotes.
“Rather than deal with the hassle of orchestrating multiple contractors, EL Media Group provides a one-stop shop,” explains Andrew Mitchel, A/V technician with the company. “Without compromising our depth, we deliver a breadth of services that greatly simplifies the A/V needs of our clients.”
In addition to audio and video design/install/support, EL Media Group can provide clients with custom-branded CDs, iPods, and USB drives; mobile app development; and A/V production.
“After comparing the overall fidelity and musical impact of multiple well-known pro audio manufacturers, it became clear that Symetrix built processors of noticeably higher quality than the rest of the field,” says Mitchel. “The Symetrix Jupiter series maintains that high audio quality and pairs it with easy deployment at a remarkably affordable price.” Inspired by the app-based paradigm of smartphone technology, integrators can configure the Jupiter hardware to suit the needs of widely divergent situations by selecting situation-appropriate “apps.”
In the case of STK, Mitchel used the “Sound Reinforcement #6” app on a pair of Symetrix Jupiter 8s. Each Jupiter 8 provides the restaurant with eight inputs and eight outputs, and the selected app provides each input with filters, graphic equalizers, split-frequency compressors, and feedback fighters; and each output with comprehensive speaker management, gain, and liming. A matrix mixer allows any input to be routed to any output.
The design for STK required multiple zones and, within most of those zones, separate outputs for full-range loudspeakers and subwoofers. The zones include the bar area, the dining area, each of two service bars, each of two private dining rooms, and the restrooms. The restaurant’s aesthetically-compelling ceiling, which is composed of individual bent slats separated by airspace that extends up to the true ceiling, proved a challenging impediment to easy loudspeaker placement.
EL Media Group located the full-range loudspeakers (from Tannoy’s Di-, CMS-, and V-series) at strategic locations between the slats. Apart from a few Tannoy V-Series subwoofers hidden near the bar, the remainder are made by Community and chosen because that have the unique ability to nestle within the booth seats. Lab.gruppen C-Series amplifiers power the system.
In contrast to the large output list, the input list at STK is modest. Most of the time, STK uses the custom music service provided by EL Media Group. They provide clients with a proprietary music player and maintain its own server that delivers user-specific content to that player.
The player allows manual track and playlist changes over and above the regularly scheduled changes programmed into the server. The other input is for a DJ rig, which is frequently used in the evening and on into the night. The currently unused inputs may be used in the future if, for example, STK adds screens or presentation capacity in the private dining rooms.
Restaurant staff can select the input source and adjust the output volume across multiple zones using the four Symetrix ARC-2e wall panel remotes. Each private room contains its own ARC-2e, and the third and fourth units are placed at the back bar and on the main restaurant floor.
“The Symetrix wall panel remotes are great,” concludes Mitchel. “They offer me tremendous flexibility in what I present to the user and allow me to concisely and intuitively present controls that would require multiple remotes from other manufacturers.”
The Eight Prime Factors That Determine The Sonic Properties Of Loudspeakers
The many individual elements that must work together in order for a loudspeaker to provide outstanding sound quality
There are eight prime factors that determine what we hear from a loudspeaker, each governed by a wide – and interactive - range of electrical and mechanical parameters that come into play.
These include the following:
—The materials used to construct cones and diaphragms (paper, aluminum, titanium, carbon fiber, composites, etc).
—Low-frequency and mid-frequency cone geometry.
—The power, linearity, and type of magnetic circuit used in drivers (Alnico, ceramic, neodymium), as well as the voice coil wire (copper, aluminum, round, hexagonal, single layer, double layer, etc.), voice coil former material, acoustical reflections from the spider; the phase plug (in high-frequency drivers), and the concentricity of the voice coil in the magnetic gap.
—Horn material and flare rate.
—Enclosure material and construction quality (rigid is always better than poorly fitted joinery), as well as enclosure volume, bass port area, duct length, and port material (hint: wooden or PVC ports are better than cardboard and soft plastic).
—Composition of interior dampening material.
—Edge diffraction from the enclosure shape; the design of the crossover circuitry.
—Driver protection circuitry (if any).
—And much more!
This list makes it clear that a loudspeaker is the sum of its parts. Less apparent is the complex interrelationship of the many individual elements that must work together synergistically in order for a loudspeaker system to provide outstanding sound quality.
Factor 1: Frequency Response
There are two aspects to frequency response. First is the overall bandwidth of the response range. A wide bandwidth, say 40 Hz – 19 kHz, provides an immediate sensation of “high fidelity.” Conversely a narrow response range such as 200 Hz – 6 kHz will be perceived as “low fidelity,” though within that range the overall performance might be quite good indeed, such as a mid-range device in a 3-way system that was not designed to reproduce the full spectrum.
The second aspect is how even, or uneven, the response may be within the intended range. An even response equates to a flat loudspeaker…which usually is a good thing. When the response is uneven, the loudspeaker is not flat and can’t be relied upon for important judgments such as balancing input channels and setting EQ. While the overall bandwidth is a function of the loudspeaker’s design, an uneven response – if the magnitude is not excessive - can usually be corrected with precision parametric equalization, though this will require the use of a high-resolution spectrum analyzer.
Factor 2: Phase Response
Intimately linked with frequency response, phase response is quickly identifiable by using an FFT (Fast Fourier Transform) analyzer to characterize a loudspeaker. Every deviation in the frequency domain will yield a corresponding deviation in the phase domain.
Though we do not hear variations in phase response as readily as frequency response, such deviations are nonetheless present in virtually all real-world loudspeakers. When the other parameters have been optimized by careful design work, variations in phase response become quite audible.
What is the difference between phase, time smear, and group delay? They are three different terms that describe the same acoustical condition, that of time variations across a loudspeaker’s frequency range at a given point in space.
When drivers are not mechanically arranged so their acoustic centers are perfectly aligned throughout crossover, where they are both providing equal energy, one energy source will lag or lead the other in time. This can be partially corrected by incremental delay, but broadband delay may not solve the problem.
Each driver, as it approaches the extent of its response range, typically exhibits a deviation from a flat phase response all on its own – not just in relationship to the other driver. Fortunately, with modern DSP technology, phase filters and/or all-pass filters can be used to minimize phase versus frequency deviations.
What does “all-pass” mean? An all-pass filter alters time in relation to frequency, rather than altering frequency response like a parametric EQ. It provides delay, but as a function of frequency.
What’s the difference between time delay and phase delay?
While the underlying mechanism is the same, when we speak of time delay, we’re usually referring to long periods of time, such as the differential between a main array and a delay tower.
When we speak of phase, we’re either talking about a 180-degree reversal (more accurately termed polarity), or we’re speaking of the relationship of arrival time in respect to frequency.
There are almost no individual drivers that exhibit uniform phase versus frequency response when measured alone, and deviations are normally much greater when one driver interacts with another driver in a multi-driver system.
Where does group delay fit into all this? When phase response is linear (flat), group delay and phase delay are identical and are the same as time delay. In a non-linear system, group delay is the slope of the phase response at a given frequency. Variations in group delay cause signal distortion (not to be confused with harmonic distortion), just as deviations from linear phase also cause signal distortion.
Factor 3: Harmonic Distortion
This is extremely important because it determines much of what we perceive when we decide that we like one loudspeaker over another. All loudspeakers produce distortion, with most being three decimal points higher than any other device in the signal path – amplifiers included. The question is how much distortion, as well as how it varies as power levels vary, along with the nature of the distortion.
Let’s discuss how harmonic distortion is measured with an FFT. Typically, a low-distortion sine wave is applied to the loudspeaker. The acoustical response is then captured with a measurement grade microphone and viewed on an FFT. Ideally, the driver should produce only the fundamental frequency of the applied sine wave.
However, in the real world, the driver will inevitably produce second, third, fourth (and higher) harmonics that are easily seen on the FFT. The combined magnitude of all the harmonics is the THD, or Total Harmonic Distortion.
But there’s more. To fully understand a driver’s distortion characteristics, one has to change the frequency of the sine wave and look at the harmonics over a large range of frequencies and power levels, a time consuming effort. What you’ll see is that the distortion products of most LF and HF drivers will increase as the frequency is lowered.
You’ll also see an increase as the power level is raised. In a high-grade driver this should be a linear function; i.e., 10 dB greater amplitude of the fundamental equals 10 dB greater amplitude of the harmonics.
At some point, however, as the driver is pushed hard enough, the harmonics will no longer maintain a linear relationship to the fundamental. It’s actually possible to measure a higher level of second or third harmonic distortion than that of the fundamental. In such case the driver is producing more than 100 percent distortion, and the sonic result is truly awful.
Knowing the frequency range and levels in which distortion starts to radically increase will greatly help when deriving low frequency port alignment, as well as determining optimal crossover points.
Factor 4: Non-Harmonic Distortion
This is even worse than the harmonic variety. When quality drivers are operated below their power limit, the distortion they produce is harmonically related to the fundamental. Apply a 100 Hz sine wave to a cone driver and the distortion “product” will consist of a 200 Hz component (second harmonic), a 300 Hz component (third harmonic) a 400 Hz component (fourth harmonic). and so on.
Though distortion is not desirable, harmonic distortion is, at least, related to music. The pure beauty of a fine piano might be compromised, but at least it will still sound like a piano. Not so with non-harmonic distortion.
When a loudspeaker’s distortion products are not related to the harmonic scale, the effect is a radical alteration in tonality. A piano might sound almost nothing like a piano, if the non-harmonic distortion products are high enough. Usually (but not always), non-harmonic distortion is the result of a mechanical problem, not a design issue, and can therefore be fixed.
Incidentally, when we say distortion “products,” we’re referring to the contribution of harmonic and non-harmonic energy that’s the product of a flawed transfer-function of electrical power being inaccurately converted to acoustical power.
We don’t want all of this extra energy coming out of our drivers, but it’s going to be there anyway.
Driver designers minimize distortion by choosing optimal materials, while mix engineers can utilize the system well below its peak output power to keep distortion at very low levels, where it belongs.
Distortion is not just related to output power, but is a function of output power, at least at this time of technological development.
Factor 5: Linearity
This is not as rigorously defined as frequency response. One manufacturer touting “linearity” might mean something quite different from that of another.
I define linearity as the ability of a loudspeaker to maintain its performance characteristics over a range of operating levels. Each time the input power jumps from 100 watts to 1,000 watts, such as during the impact of a snare or kick drum, if the loudspeaker increases its distortion, alters its frequency and phase response, or does not respond with precisely 10 dB greater acoustic output, then it will be exhibiting one, or more, non-linear characteristics.
Conversely, if none of its response parameters alter at all – other than an increase in output level – then the loudspeaker exhibits linearity.
No loudspeaker is truly linear throughout its full power and frequency range, though some come close. Most cone and compression drivers exhibit significant non-linearity as they approach the upper extent of their power handling, and also in respect to the program material.
A given loudspeaker might be good at accurately reproducing a single 100 Hz sine wave with low distortion, for example, but may “fall apart” when trying to reproduce the complexity of multiple musical tones that all occur together. Therefore, detecting distortion by stimulating the driver with only a single sine wave does not tell the whole story.
Some acoustic analyzers provide multi-tone sources for distortion measurement, as well as frequency sweeps and automatic power level increments. Both are excellent tools for approximating real-world musical passages.
Factor 6: Transient Response
This is the time that it takes a loudspeaker to respond to the input stimulus, and how quickly it stops producing energy after the stimulus ceases. As with the other parameters in this article, the answer will always be a function of the frequency of the stimulus.
Some analyzers can display a 3D waterfall plot, which depicts the variations in the start and stop time versus frequency, as well as magnitude-versus-frequency of the steady-state period in which the loudspeaker has settled after initial acceleration, and before the stimulus has ended.
Obviously, the faster that a given loudspeaker responds, the more accurate it will sound. However, a very fast loudspeaker may not sound as “warm” or desirable as a less accurate one. That’s because we’re schooled by a lifetime of listening to loudspeakers that exhibit a relatively slow transient response, especially in the very low frequencies.
In listening tests, many people prefer a slow subwoofer to a fast one, because it sounds like it’s “filling out” the bottom end. Moreover, most musical instruments do not exhibit uniform transient response. The 9-foot bass strings of a grand piano do not start and stop anywhere near as quickly as the 6-inch strings in the upper register – by several orders of magnitude. Nor does a tympani exhibit the same transient response as a pair of claves.
It’s therefore a common human response to desire a slower transient response in the low end, while preferring a faster transient response in the high frequencies, particularly in respect to naturally occurring acoustical events. This works out well because a heavy 21-inch woofer cone is never going to exhibit the same transient response as a 1-inch soft-dome tweeter.
If you wish to experience music reproduced with extremely low distortion, highly uniform and precise transient response, and near-perfect phase/frequency response, listen to electrostatic headphones, such as the STAX line. With a diaphragm that’s only 3 microns thick (3 microns = 0.000118 of an inch) and weighs almost nothing, electrostatic headphones are a great way to train your hearing skills.
The clarity and evenness of response will probably never be matched by a PA loudspeaker, because it must provide far greater output power in order to be useful. And that brings us to the last two factors.
Factors 7-8: Power Output & Dispersion
These two are closely related, because one is partially the function of the other. High power systems typically exhibit narrow, or at least controlled, dispersion in one or both axes. Examples are line arrays and long-throw horns.
When acoustic energy is concentrated, it increases in intensity, though often at the expense of higher distortion and less response uniformity. Couple this with drivers that are engineered to be powerful rather than uniform and linear, and the sonic quality can suffer.
Conversely, a smaller loudspeaker might exhibit nearly perfect response in all other categories, but only be capable of providing enough power to function as a nearfield monitor with no dispersion control, a poor candidate for sound reinforcement in large, reverberant spaces.
Power output capability and dispersion play one of the largest roles in how useful a loudspeaker might be – hence the first spec on many contract riders is often system wattage, or SPL at a certain location, usually the front of house console. Although neither will give you a clue as to how the system might actually sound, and whether it’s properly covering the seating plan, it’s still a prevailing demand made by production managers and sound engineers.
Conclusion
What happens when a loudspeaker falls short of achieving reasonable performance in one or more of these factors?
—It may sound cloudy and unclear.
—It may favor one musical register (or even one note) over others.
—It may hurt the ears with excessive distortion.
—It may perform well at low levels but poorly at high levels.
—It may not cover the audience well, especially on the fringes.
—It may cover the audience too well, sending too much energy toward sides, ceilings, and rear walls, causing undue room excitation.
—It may simply not get loud enough to handle the show’s requirements.
—And, it may do much…or even all of the above!
Ken DeLoria is senior technical editor for Live Sound International and ProSoundWeb, has had a diverse career in pro audio over more than 30 years, including being the founder and owner of Apogee Sound.
Thunder Audio Provides Meyer Sound MILO For Movement Electronic Music Festival
The Movement Electronic Music Festival, taking place May 26-28 at Detroit’s Hart Plaza, will feature ab array of Meyer Sound equipment provided by Livonia, Mich.-based Thunder Audio.
The annual festival, which drew some 100,000 attendees in 2011, has grown and evolved considerably from its inception as the Detroit Electronic Music Festival in 2000. This year, three of five stages will feature Meyer Sound components.
“Our company was unaware of Meyer in a practical sense until we teamed up with Thunder Audio on the Movement Festival a few years ago,” recalls Sam Fotias, operations manager of the Movement Electronic Music Festival and Detroit-based Paxahau Event Productions and Management. “We had tried other loudspeaker products, but nothing really gave us the raw power and, more importantly, clarity at high SPL that we were looking for until we were exposed to MILO.”
The festival’s Main Stage will boast a system comprised of 48 MILO and 16 M’elodie line array loudspeakers; 36 700-HP subwoofers; two UPA-1P and UPA-2P loudspeakers each; and a Galileo loudspeaker management system. The monitoring system includes 12 Meyer Sound MJF-212A stage monitors.
The Beatport Stage will feature 24 MICA, 12 MILO, and eight M’elodie line array loudspeakers; 18 700-HP subwoofers; three UPJ-1P VariO loudspeakers; and a Galileo loudspeaker management system. The Made in Detroit Stage will feature 24 MILO and eight M1D line array loudspeakers, and Galileo.
“We got involved in this event six or seven years ago, and there was all manner of different configurations on each stage,” recalls Thunder Audio’s Paul Owen. “We came in and did one stage, and then they saw how good the Meyer product was—the powered line array, without hundreds of amp racks everywhere.”
Fotias adds: “Having Tony [Villarreal] and Paul and their team from Thunder so close is amazing. They have been such an instrumental part of allowing this event to grow and being able to facilitate all of the big crowds we get.”
Quote meon an estimate et non interruptus stadium. Sic tempus fugit esperanto hiccup estrogen. Glorious baklava ex librus hup hey ad infinitum. Non sequitur condominium facile et geranium incognito.
