Thursday, October 03, 2013

Holy Light Church Of Deliverance Bishops Utilize AKG Wireless For Inspiring Services

Holy Light Church Of Deliverence installs new Harman Professional sound reinforcement system for improved clarity.

Led by Bishop Samuel L Carruth, The Holy Light Church Of Deliverance has opened its doors for nearly 50 years for the St. Portsmouth, Virginia community. 

With an aging sound reinforcement system installed in the early ‘90s, Vicom was tasked with designing and supplying the church with a new system to ensure the upbeat services were clearly heard among the crowd of devout attendees.

Vicom installed a complete Harman system, including 10 sets of AKG WMS4500 wireless to improve live sound and recording ability. Handheld C5 microphones are on hand as presenters take advantage of the entire stage during services.

The Church now boasts a Soundcraft Si Performer 3 digital console, with a Compact Stage Box for up to 56 channels of mixing ability.

Dual clusters of four JBL VRX932 loudspeakers fill left and right sides of the church, while two VRX918 subwoofers were installed left and right on the ground for added low end.

The loudspeakers are powered by a total of eight Crown XTi 4002 and 6002 amplifiers.

“When we designed the Harman solution for the Holy Light Church, our main concern was clarifying the intelligibility of the vocals and band so any performance from a single speaker to a large ensemble would be completely audible; not bouncing off the walls,” stated Andrew Fantin, Sales Engineer, Vicom.

“The church is very lively in its performances and many times they reach levels of 100db. The AKG microphones are put to the test from the second the service starts to the end of the day. The entire Harman solution was a natural progression for the pastors and audience.”


Posted by Julie Clark on 10/03 at 11:17 AM
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Wednesday, October 02, 2013

The Northstar Session Uses Bose L1 Systems And B2 Modules For Acoustic Trio Sets

Rising California rock act finds the right blend and harmonies with their Bose L1 systems on stage

California-based rock band The Northstar Session has been thrilling audiences and gaining fans since its formation in 2007. The trio has a rigorous performance schedule with dates on both coasts, and they have earned a strong following for their literate songwriting and sunny harmonies.

A key element of their live performance schedule centers on their acoustic trio sets in smaller-to-medium-sized venues, and for these performances, the band uses L1 portable systems and B2 bass modules, along with T1 ToneMatch audio engines, from Bose Professional Systems Division

Matt Szlachetka, the band’s primary guitarist/vocalist, notes, “I think we found really quick that our harmony is our strength. To showcase that, we developed the acoustic trio sets. It’s a really good way for people to latch on to what you’re doing.”

Keyboardist/vocalist Dave Basaraba adds, “The easiest way to highlight that was to strip it down and make sure the harmonies can be heard. And there was the Bose L1 system.”

The compact nature of the L1 system simplifies the equipment setup the band has to haul with them.

Drummer/vocalist Kane McGee notes, “On the East Coast, we just travel in a pickup truck. We are able to fit everything into the bed of the truck.”

The smaller stages of these venues can also present certain challenges.

“A lot of these stages are really tiny. We don’t have a lot of space for amplifiers and stuff, let alone bringing in cumbersome speaker stands and monitors. We’ve got to be able to hear everything,” Basaraba notes. “It’s clubs and coffeehouses and bars. You’re trying to fit everything cohesively on stage, and at the same time make it look presentable,” Szlachetka adds. 

The band’s acoustic setup includes L1 systems directly behind McGee and Basaraba, along with a B2 bass module.

“We don’t have any feedback issues that way,” Szlachetka notes. “And the B2 bass modules, whether you’re doing a more stripped-down acoustic performance or a larger-scale electric thing, they’re incredibly powerful. They’re fantastic.”

“I’ve never heard anything like the L1 systems ever in any P.A. system I’ve ever used,” notes McGee. Basaraba says, “The mix and clarity of the instruments that you get with the Bose L1 system and using the ToneMatch mixers is incomparable.”

Szlachetka notes, “The reason that we’re able to soundcheck usually in about ten minutes is because of the presets on the ToneMatch engine, and then that gives us more time to relax and focus on the performance.” McGee adds, “You’re not thinking about the sound. It’s already there. It’s done. You’re just having fun.”

“Having the consistent sound all the time, it’s incredibly important,” Szlachetka continues. “People always comment, ‘You guys sound amazing!’ A lot of times they’ll look around and go, ‘Where’s your P.A. system?’” 

Musically, the setup lets them find the right groove on stage.

“We’re virtually hearing what the audience is hearing, so that makes it really easy to harmonize, Basaraba notes. “With the L1 systems we’re all experiencing it together. You tend to feed off the audience response and give them a little more, and they give you a little more back. It’s a conversation.”

Szlachetka sums it up: “It’s been allowing us to connect with audiences on a higher level. For what we’re doing it’s like the best P.A. we could possibly use.”

Bose Professional

Posted by Julie Clark on 10/02 at 07:38 PM
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Mexican Stadium Juárez Vive Comes Alive With Danley

Juárez Vive is a stunning sports stadium that features sound reinforcement provided by Danley Sound Labs loudspeakers and subwoofers.

Located in the US border city of Ciudad Juárez, Mexico, the recently-completed Juárez Vive is a stunning sports stadium and a proud symbol of the recently-embattled city’s rejuvenation.

When the baseball season got underway in April, the newly-completed stadium accommodated 12,000 fans for a thoroughly-modern game-day experience.

An important component of that experience is now a highly-intelligible, pleasantly-musical sound reinforcement delivered by Danley Sound Labs loudspeakers and subwoofers, with support from Ashly amplifiers and Symetrix processing.

The Mexican government had great expectations for the sound system’s performance. North Carolina’s Clarity Incorporated designed the system which not only exceeded those expectations, it did so on the narrowest of budgets.

The new stadium thus bears a level of personal investment from city and government officials that have influenced its design and construction.

“The Governor of Chihuahua, César Duarte Jáquez, toured the construction site frequently to track and encourage the progress,” explained Rich Mason, president of Clarity Incorporated. “Our charge was to deliver ‘sound that was befitting the best of Mexico.’ They expected us to jump a very high bar.”

The construction company, Afirma, only involved Clarity Incorporated after first determining that the original sound system designer wouldn’t be able to deliver that kind of performance on budget.

As a result, Clarity Incorporated started the design three months late and operated on an emergency schedule.

Bill Weir, Clarity Incorporated’s director of technology designed the system with assistance from Ivan Beaver, Danley Sound Labs’ chief engineer. A small crew from Clarity Incorporated spent three weeks on site to assist Afirma with the installation.

“This is a value-engineered system,” asserted Weir. “Occasionally, you get a big-budget project in which issues can be overcome simply by throwing money at them. You don’t have to give them a lot of thought.

“But in this day and age, and especially for a government client, money is tight and you have to carefully balance the tradeoffs inherent in any decision, but in such a way that no one feels that it’s a compromise. That’s a value-engineered system.”

Loudspeaker and subwoofer placement at the new stadium looks deceptively simple. Nineteen Danley SM-80 full-range loudspeakers ring the lip of the roof that covers the stands, and every other SM-80 is joined by a Danley TH-118 subwoofer.

“We’ve been huge fans of Danley from the very first moment we heard their loudspeakers,” said Mason. “Their phase coherence and pattern control are unrivaled, and they present the most natural sound stage I’ve ever heard short of studio monitors.”

Weir, a loudspeaker designer himself, originally drew up the plans with a Danley SH-69 and a Danley SM-96 at each location.

“I was able to quite nearly create an equilateral triangle between the slant of the seats and the point at which the loudspeakers would hang,” he said. “I wasn’t aware of the new SM-80, and so the SH-69 and SM-96 would combine to give me the appropriate coverage.

“Ivan realized the SM-80 would be as effective, but at a fraction of the cost, and suggested them as a replacement. I’ve worked with him enough to trust his recommendation on an unproven product. Sure enough, he was right. I’m blown away by the SM-80.”

Weir observed that subwoofers are often omitted from stadium designs.

“With conventional subs, it’s hard to retain low end definition or clarity in a stadium situation,” he said. “It’s just mud. In contrast, Danley’s tapped-horn subwoofers have vastly lower group delay and a very definite focus that you can’t get from conventional designs.

“Put another way, it doesn’t matter how loud or low something goes, it’s the manner in which it does so that matters. And Tom Danley’s bass is not only loud and low, it’s musical and defined.” The stadium’s roof and appropriate spacing also contribute to exceptional low frequency definition.

Heil microphones and a handful of other input sources feed a 16-channel Yamaha LS9 console, which in turn feeds a Symetrix 8x8 DSP with a Symetrix BreakOut 12 for additional outputs.

“Given the circumstances, we didn’t have a lot of design cycle time on this job,” said Weir. “And as well as one might plan things out, the system requirements are likely to change on site.

“Symetrix has a reputation for building solid algorithms that are supported by well-designed analog circuitry. Its flexible open-architecture topology allowed me to perfectly tune the system functionality while I was in Juárez.”

Nine Ashly pe3800 and four Ashly ne2400 amplifiers power the system. All of the Ashly amplifiers are networked to allow Ethernet control from a central location.

“Ashly is another company that puts sound and reliability first,” said Weir. “Their network amps are a great example of appropriate functionality. They sound great and maintain a robust low end even with a lot of speaker cable.

“Of course, that kind of sound quality is paramount. Beyond that, the network capabilities meet the client’s needs without adding any costly – but ultimately unnecessary – bells and whistles.”

He continued, “Clarity has no obligations to any manufacturer. I can use whatever I want in my designs. Given the design expectations and constraints at Juárez Vive, I’m certain that this is the only combination of gear that would have succeeded.

“It’s a very unique synergy, and I’ve never heard a better system for anything less than five times the price. From the client’s perspective, it’s simple: they have a far better audio system than even dared imagine possible, and they stayed on budget.”

Danley Sound Labs

Posted by Julie Clark on 10/02 at 10:53 AM
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Tuesday, October 01, 2013

MF Audio Chooses Adamson E15

Recently, MF Audio invested in a new Adamson System Energia E15 PA, which joins their other Adamson inventory items -- Spek-Trix, Y10, Y18 and Metrix – it was a likely pairing that has suited everyone extremely well.

MF Audio, founded in 1998 and based in the North of Paris, is a full service live event production company providing sound, lighting, and staging solutions.

The principle, Sebastian (Seb) Fleury has made a point to surround himself with accomplished production professionals, which has led to the success of the live event production company.

One of the other secrets to it’s success has been to ensure that all products in their inventory meet their four qualifications – quality, reliability, usability and versatility.

Recently, MF Audio invested in a new Adamson System Energia E15 PA, which meets their four qualifications in spades. As a long-time supporter of Adamson – with previous investments in Spek-Trix, Y10, Y18 and Metrix – it was a likely pairing that has suited everyone extremely well.

“We were looking for a line array system that would allow us to expand our PA offerings as business has grown,” explains Fleury. “Our goal is to always be at the edge of technology and provide ever more efficient solutions that combine innovation and performance. Purchasing the Adamson E15 system helps us achieve our goals.”

The original investment consists of 18 E15 enclosures powered by Lab.gruppen PLM20000Q amplifiers with LAKE signal processing and digital audio transport via Dante.

MF Audio christened the new PA on the main stage floating on the lake of Enghien Les Bains.  The system was also used in support of France distributor DV2 at Champs de Mars on July 14.

“With the E15, we sounded better than anyone else,” Fleury concludes. “The system is lightweight and easy to rig, amazingly consistent and has headroom that doesn’t quit. We are delighted and eagerly await the arrival of the new E12 line array so we can check that out as well.” 

Adamson System

Posted by Julie Clark on 10/01 at 12:34 PM
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Renkus-Heinz VARIA Delivers For Bethany Wesleyan Church

New system versatile enough to accommodate everything from worship services to concerts and even sporting events

From its 80-acre campus in Cherryville, PA, Bethany Wesleyan Church combines an inspirational, contemporary worship style with a broad mix of community programs including sports, education and family oriented activities.

With a dedicated and rapidly expanding congregation, Bethany Wesleyan recently took the wraps off of a new 1,600-seat sanctuary and multipurpose room, equipped with a custom designed sound system based around the versatile Renkus-Heinz VARIA modular point source line array.

“The church wanted a space that would be versatile enough to accommodate everything from worship services to concerts and even sporting events,” explains Allen “Doc” Nagle of Orwigsburg, PA-based Events Staging. “It’s essentially a gymnasium with a stage at one end and removable seating.”

As any systems designer can attest, multi-purpose rooms often present some of the most challenging audio environments, and Bethany Wesleyan was no exception. The room’s high, open ceilings and concrete and sheetrock walls could have been a potential nightmare. But as Nagle explains, “thanks to good communication with the architect during the planning phase, we were able to take steps to reduce standing waves and reflectivity.”

For the sound system, versatility was key. The system needed to keep the sound tightly focused on the room’s key listening areas, which could change via a variety of room configurations. It also had to accommodate a wide range of program material, from spoken word to different types of musical content.

“VARIA was perfect because it provides the best options for placing sound exactly where you need it in the room, as opposed to spilling it all over the place,” says Nagle. “VARIA’s transitional waveguide allows you to vary the horizontal coverage from the top of the array to the bottom, which was perfect for shaping the coverage pattern off the reflective surfaces.”

The system comprises left and right hangs, each with five VARIA VAX101 series cabinets. Horizontal dispersion varies from 60 degrees at the top of the array, increasing incrementally to 120 degrees at the bottom. A center fill of two VARIA 90 degree cabinets adds a bit of psychoacoustical enhancement.

Eight PNX212-SUB subwoofers are bunkered, four per side, into the face of the stage. An Avid VENUE Profile at the front of house position handles the mix, while four CFX121M wedges provide onstage monitoring.

“The VARIA made all the difference in this installation,” adds Nagle. “It allowed us to easily pinpoint exactly where the sound should be in the room while avoid any unwanted interactions. It sounds amazing.”

An ancillary chapel and cafe incorporate the Renkus-Heinz IC7-II steerable array, with a pair of PNX212 series subwoofers providing a solid bottom end. Ashly Audio NE2424m matrix processors handle speaker management for the entire facility while providing a selection of easily recalled presets for the church’s all volunteer crew. 

Visual impact is equally impressive, with three 16,000-lumen projectors firing at three, 20-foot screens positioned left-to-right across the back of the stage.

Finally, multiple HD PTZ and manned cameras and a Pro Tools system capture weekly services for distribution on DVD, with plans already in place to expand into Internet streaming and live broadcast.

With the system up and running, Nagle is pleased that his team was able to strike such an effective balance between usability and performance, while placing the church on a solid foundation for future growth.

“This is a really dynamic and active congregation,” he concludes. “They’re looking to entice national worship tours, as well as setting the stage for broadcast and streaming. The combination of the VARIA and Avid systems ensures them a level of sound quality and technical capability they’ve never had before. I’m confident they’re set for many years to come.” 



Posted by Keith Clark on 10/01 at 03:14 AM
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Field Museum Of Natural History In Chicago Goes Modern With L-Acoustics

Frost Chicago and Clearwing Productions install ARCS WIFO and XT Series systems at James Simpson Theater

With its majestic Doric-style columns, intricately carved molding and beautifully detailed 20-foot recessed ceiling, the 700-seat James Simpson Theater at Chicago’s Field Museum of Natural History is popular among Fortune 500 companies and other organizations as a prime venue for corporate events.

Recently, production design firm Frost Chicago enhanced the nearly 100-year-old space with a very modern addition—a full surround sound system headed by L-Acoustics components to complement a new 3D/HD digital cinema projector installed by D3D Cinema.

For audio design and installation support, Frost Chicago turned to Milwaukee-based Clearwing Productions, which specified a system built around L-Acoustics ARCS WIFO and XT Series product ranges.

According to Clearwing Systems project manager Megan Henninger, three pairs of ARCS WIDE loudspeaker enclosures were installed at the front of the room for LCR audio reinforcement: two cabinets horizontally flown above both the far left and right sides of the stage with a third pair positioned behind the newly-installed retractable, perforated screen.

Four tiny 5XT coaxial systems spread out and hidden behind a dark grille on the face of the stage provide supplementary front-fill coverage to the first several rows of seating while two SB18i subs centrally stacked upstage deliver the low end.

For surround, the theater now benefits from 10 coaxial 8XTi enclosures all standard-finished in white to discretely blend in with the classically elegant setting. Eight of these are ceiling-mounted between the columns on either side of the room, while two more are positioned at the back of the space for rear left and right channel audio. The entire system is driven by a total of six LA4 amplified controllers.

“Preserving architectural integrity was one of the biggest concerns on this project,” notes Henninger. “With the historic Field Museum being nearly a century old, most of the surfaces in the Simpson Theater are plaster, but the client didn’t want to disturb the look of the room by altering structures or adding acoustical treatment.

“Thankfully, the 8XTi surrounds pack a lot of punch in a very low-profile design, and their standard white-finish option allowed them to seamlessly blend into the space. And although the hang points for the main PA were hardly ideal, the precise coverage pattern and sound quality of the ARCS WIDE enclosures allowed them to perform exceptionally well.”

“As the exclusive vendor for production at the Field Museum as a special events venue, the client came to us wanting to increase the amenities and marketability of the Simpson Theater for rental use,” says Frost Chicago principal David Kelly. “Additionally, with the success of the 150-seat 3D cinema on their upper floor, the museum recognized the opportunity to generate additional visitor revenue by creating a second, larger 3D space on the ground floor.

“With that in mind, our task was to create a system that could handle a wide variety of potential uses—from shareholder meetings and product launches to concerts and truly immersive 3D experiences—and we knew that L-Acoustics would provide a top-notch solution. We frequently deploy our own KARA rental system for events at the Field Museum’s Great Hall and the equipment’s integrity and fidelity have always delivered a first-rate experience. Since the Simpson Theater’s new system first debuted this past Memorial Day weekend, both D3D Cinema and the museum have been absolutely thrilled with the results.”

In addition to the sound system, Frost Chicago spearheaded the installation of the theater’s screen and soft goods (e.g. draperies) as well as coordinated the related electrical work.

Frost Chicago
Clearwing Productions

Posted by Keith Clark on 10/01 at 03:03 AM
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Friday, September 27, 2013

The Right Balance: Amplifying Acoustic Instruments In The Real World

Observations on stuff that works as well as stuff that does not

I started out wanting to write about the design task of coming up with hybrid microphone and pickup rigs for Punch Brothers (a progressive acoustic band that I work with) and building a completely self-contained stage and mix system. But that initial focus has changed, where I now feel the need to share some of what I’ve learned by plugging it in to many different sound systems, as well as offer some observations on stuff that works and stuff that does not.

Punch Brothers had been touring for several years with a carefully conceived “microphone only” system that worked very well in controlled concert situations, but we found ourselves seriously limited in many louder venues and made the decision to add pickups to it. The goal was to put together a “bulletproof” package that would work in every venue, everywhere.

After much research and discussion, we booked time in a rehearsal room with a sound system and a pile of gear to build our pickup systems (despite our serious dislike of pickups) with a highly developed sense of what we thought the instruments should sound like. We tried various combinations of pickup, preamp, and effects with all of us sitting as jury, until everyone was satisfied with all sounds and our house mix output sounded as much like the record as possible.

This self-contained stage system included all stage wiring, monitor mixer with passive split, in-ear monitoring, pedal boards, front-of-house preamps, processing, and mixer all packaged to be checkable and able to fly. It was carefully designed to make setup and strike easy. and so that the only things that would change technically from show to show were the snake to FOH, the amps, the processing, and the loudspeakers.

Absolute Control
Amplifying acoustic instruments is always difficult at best, and having an entirely repeatable input scheme gave me an efficient way to judge the performance of sound systems and to avoid making corrections at the console that should be made on the system. It tended to make any kind of system problem readily apparent, and I became more aware of issues related to acoustic amplification that tend to be problematic.

First in line is a very general comment on amplifiers, loudspeakers and their associated processing. Most of the systems I encountered were technically adequate, but there were all too many that were optimized for loudness and then locked in the system processor beyond all discussion.

In terms of amplifying acoustic instruments I would argue that any equalization that is done in setting up a sound system, whether by ear or even by technical measurement, is completely subjective and does not relate to how my inputs will react on “this stage with this system.” I would go so far as to say that good amplified acoustic sound does not require fidelity in a sound system as much as it demands absolute
control of it.

All acoustic instruments are completely subject to their sonic environment. They require the application of energy to make mechanically produced sound. When the sound of the instrument is amplified with a mic or a pickup, the sound from the loudspeakers becomes additional energy applied to the instrument and microphone, creating a passive feedback loop between the main loudspeakers and the instrument, with a resonant peak consistent with that of the instrument itself.

The fact that this resonant peak occurs in the same frequency range where most sound systems have been beefed up to get loud makes for a double handful of energy to keep under control. This makes the term “flat” have little or no meaning in live sound and may be the main difference between recording to a reference loudspeaker and amplifying sound in a room.

Correcting these resonant peaks and feedback loops with input channel EQ is always the right place to start, but the cut made should be done equally in the main loudspeaker system as well, in order to avoid thinning (or reducing gain on) what is coming through the console too much. Keep in mind that finding that balance point between console and system EQ can be difficult with multiple loudspeaker systems and
multiple inputs, but if you know that input sounds right at the console, you can lean more and more on system EQ – and even reduction of gain to the amps.

Sound systems that are optimized for loudness (which is most of them) are basically the opposite of what is needed for acoustic instruments. The fact that systems do have a lot of power in low mid and low end, and are often placed too close to the stage always makes it difficult to get a musical balance with an acoustic instrument.

By musical balance, I mean when the amplified instrument can play all notes in its range with the same perceived loudness in as many places as possible in the room. It’s always the interaction of the instrument with the system that dictates the degree of control needed. This contrasts sharply to a rock band that has very loud sound sources and directly synthesized sounds with little or no interaction with the main loudspeakers.

Key Interaction
Another issue that was encountered over and over in many systems was that of the efficiency of high-frequency response. The problem that kept appearing was that high-frequency levels on a system would sound perfectly fine until there was a dynamic peak with a lot of high-end content (a fairly common occurrence with acoustic instruments), and then it became very efficient on this peak, making the system far too bright.

In fact, I found this so often that I started calling it a syndrome, and in extreme cases, actually resorted to de-essing the entire system. Theories about why this might be happening include:

1) Perhaps the amplifier is too powerful and I’m barely driving it (over-powered system). 

2) It could be as a result of individual system setups and their locked processors. Perhaps it’s also due to poor gain stage at some point in the chain, with some poor little op-amp at the edge of distortion. Or it could be a combination of all these things. To be fair, it did not happen with the best of the systems, and particularly, the lovingly maintained older systems.

Being blessed with no monitors on stage, and after having determined that all loudspeakers were working, my first order of business was always to check the interaction between the mains and the stage. This interaction is the single largest limiting factor in amplifying acoustic instruments. I deal with it by tuning the system like a giant stage monitor, and on problem stages, sometimes resorting to altering the phase relationship of individual inputs. In really bad situations where the system involves the entire stage,  the only recourse is to turn it down.

Loudspeaker placement is always dictated by space and sight lines, but unfortunately also by things like the length of the lighting truss and even worse, by someone’s sense of aesthetic: subwoofers and stacks on stage, or loudspeakers coupled in any way to the stage, are the “worst case scenario” of loudspeaker interaction, and it
means you’re going to have a bad day.

Many older loudspeakers radiate large amounts of low and low-mid behind them, and there are venues where the loudspeakers are hanging just a few feet over the stage. Poor placement presents the biggest challenges, with a huge effect on acoustic instruments, and it’s found regularly in venues around the world. In defense of those who decide where to put loudspeakers, while there might be other options, these are often extremely expensive to achieve. I’ve found that loudspeaker placement relates directly to my level of perspiration during a show and degree of satisfaction at the end.

Placement of mains often leaves an area of poor coverage along the front of the stage and sometimes out into the center of the room, requiring front fill. These situations are the real wild card in amplifying acoustic instruments, with the potential of being the most interactive and likely to change their relationship to microphones on stage when a wall of people line up in front of them.

A front fill system should always have its own EQ separate from mains at FOH. just like a monitor mix. Wedge mixes for acoustic instruments should always be from post EQ mixes from FOH and not from a monitor desk on stage. If all loudspeaker systems, including monitor wedges, are balanced properly, then the input channel EQ becomes the master, allowing you to compensate for loudspeaker interaction in the right place.

Separating stage and house sound with acoustic instruments can work in some places, but I would argue that it’s better to treat it as one system. No stage monitors or IEM systems provide maximum control at FOH, and any additional loudspeakers that do not follow input EQ from FOH will degrade overall quality in the room with the addition of its energy to the feedback loop. This can easily lead to overcompensating at the console and system EQ.

I’ve seen many different configurations of front fill, but by far the most effective placed the loudspeakers just off stage, mounted as high as possible above the heads of the audience, and angled in and down towards down stage center. I do not agree with delays on front fills on most stages, and usually push them as far upstage as possible and closer to the band to get less delay.

Control of front fill can often be the key to having good sound, particularly in very ambient rooms. Even a slight shift of level from the mains into the front fills can make an acoustic instrument sound more believable by balancing sources of energy in the room making it feel like there is more coming from the stage.

The biggest problem with front fills (mostly with standing audiences) is the sometimes enormous change that occurs in their relationship with the stage when people are standing in front of them. The change can be minimized by placement, but I hover over the front fill EQ at the top of every show to compensate for the change that I know the audience will generate.

Level Considerations
I’ve put a lot of thought in to what many call “saturating the room.” I’m referring to that point of loudness, particularly in more reverberant rooms, where reflections seem to equal or even exceed what is coming from the loudspeakers – and everything turns to mush. This phenomenon is caused not by the acoustic qualities of the room, but by the fact that we hear longer “tails” on louder sounds.

How loud you can mix is directly related to the density and quality of the reflections in the room. Most rock clubs have dealt with this by deadening the room and optimizing high frequencies in the system to restore clarity and detail, but in a concert hall it can happen at a very low level. The only way to avoid it is by turning everything down, and focusing on tone instead of loudness.

It’s my observation that most production sound systems, particularly in terms of testing and setup, are optimized for a much louder level than the room can reasonably stand with acoustic instruments. For instance, a frequency sweep in an empty room at 100 dB will invariably be compensating for any destructive interference that is entirely likely to be going on, as well as acoustic anomalies caused either by loudspeaker placement or architecture.

If system diagnostics are run at a level that exceeds what the room can handle, it can make musical balances more difficult to achieve. It’s far easier to exceed this threshold in an empty room, but I’ve found that it’s raised by as much as 15 dB or more (depending on ambience) when the audience is in. An empty room can also suffer extreme loudspeaker placement problems like excessive comb filtering or bad flutter echoes that can completely disappear when the audience is in place.

There are now many concert halls that have seating designed to minimize the difference between empty and full rooms, but that’s really all they do. The effect of an audience on amplified sound is something I never tire of experiencing. It’s subtle in some rooms and very pronounced in others, but always there, and always makes it better.

What I observe to be happening is a change in the way sound energy is moving in the room, particularly in terms of how the loudspeakers are interacting with the stage. This is partly about the front fill issue, but also about how the house sound is reflected off or absorbed by the audience, the ambient temperature and humidity, and the expectant energy they bring with them. This all has a profound effect on how much sound level the room will support and will always offer more than an empty room sound check.

Eliminating Variables
So several hundred sound systems later, after balancing and tweaking and fixing and moving, some things are very clear to me in the amplification of acoustic instruments. The physical arrangement and setup of a sound system – and it’s relationship to the stage – is the most important factor in getting a musical balance.

Further, the choice of optimizing for what is coming out of the loudspeaker instead of how the loudspeaker is presenting itself to the room can make this job much tougher. Making an entirely isolated sound source astonishingly loud on a loudspeaker is easy, but tuning multiple feedback loops to achieve a musical balance is remarkably difficult.

In eliminating as many variables as possible on what I was plugging in to a sound system, it became easier to get this balance and to isolate problems. Instead of building an input structure to accommodate the loudspeakers, I’m manipulating the sound system to balance in a musical way with my input structure.

One aspect that clearly stands out at every show and on every system: any change in any loudspeaker system that’s in conflict with what the musicians need to hear to play properly is simply wrong. The technical side of amplifying sound in a room should always act in support of what is going on musically and not just in regard to playback fidelity. Just like an acoustic instrument is entirely reactive to the space it is in, so should be the sound system making it louder.

Dave Sinko is a Nashville-based audio engineer in studio recording, mixing and mastering, live touring and production sound mixing. Current projects include Punch Brothers, Goat Rodeo Sessions,  and T-Bone Burnett.

Posted by Keith Clark on 09/27 at 02:41 PM
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Oslo Concert Hall Upgrades To Meyer Sound MICA Line Arrays

A quarter century after installing eight UPA-1 and two USW-1 conventionally powered loudspeakers, the venue again turned to Meyer

The Oslo Concert Hall (Norwegian: Oslo Konserthus) in Norway has relied on its Meyer Sound equipment since 1986. A quarter century after installing eight UPA-1 and two USW-1 conventionally powered loudspeakers, the venue again turned to Meyer when it came time for an upgrade in the form of MICA line array loudspeakers supplied and installed by Oslo-based AVAB CAC.

The 1,404-capacity Oslo Concert Hall regularly features a diverse schedule of classical, blues, jazz, and rock performances. The hall’s legacy Meyer Sound equipment has proven its worth, and venue management was ready for a system with more power and flexibility.

“I think it’s unique that our UPA-1 system was so old yet sounded so good,” says Jan Olsen Skare, long-time production manager of Oslo Concert Hall. “Now with the MICA, we have a system that really delivers great reinforcement for many different types of music. In terms of coverage, every seat in the house is a great seat.”

Oslo Concert Hall added 18 MICA loudspeakers, four 600-HP subwoofers, and a Galileo® loudspeaker management system with a Galileo 616 processor for system management and processing. The new components are used alongside the venue’s existing Meyer Sound inventory, including four each UPA-1P, UPM-1P, and CQ-1 loudspeakers.

“MICA delivers the high output and smooth, extended high-frequency response that Oslo Konserthus requires,” says Asle Nilsen, head of sound at AVAB CAC.

Since the audio upgrade, the concert hall has hosted many jazz and classical performances, as well as American singer/songwriters Emmylou Harris & Rodney Crowell, Melody Gardot, Nanci Griffith, and Norwegian singer Wenche Myhre.

Meyer Sound

Posted by Keith Clark on 09/27 at 12:29 PM
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Wednesday, September 25, 2013

Tweaking DSP for Stage Monitors: Tips & Tricks To Maximize Potential

Deploying some DSP horsepower and modern test equipment to minimize problems

In the days when digital signal processing (DSP) first stalked the arena, it was the guys at front of house that had all the fun. They would ignore their comm light for long periods of time while staring intently at asymmetrical crossovers on their laptop displays.

But now that DSP is ubquitous, the guys at the other end of the snake are beginning to experience the joys of audio in the digital domain. 

The real beauty of DSP in monitorland is in optimizing wedges and side fills so that little or no equalization is needed to minimize feed back.

You might have a rack full of graphics, but the show will start out with all of them set flat. These units will only be used for on-the-fly feedback reduction or to ”personalize” a mix.

And with DSP, a preset can be designed for each model in your inventory, particularly useful if you have multiple wedge types. So let’s get on to the tweaking part.

Very few monitor cabinets are inherently “flat”. There are compromises made due to factors like box size versus low frequency driver selection, box size versus horn selection, low frequency driver dispersion versus. crossover frequency, etc.

Your favorite microphones also have frequency anomalies, especially in off axis response. Combine the two and things can get quite complicated.

Many of these problems cannot be completely cured, but with some DSP horsepower and modern test equipment, they can be minimized.

The first step is to achieve flat on-axis response from your monitors. I would highly recommend that if you don’t already own Gold-Line TEF, Rational Acoustics Smaart, or at least a real-time analyzer (RTA) then beg, borrow or steal some audio analysis equipment for this portion of the procedure.

Measured with TEF, a high-frequency device before equalization. (click to enlarge)

At an AES convention a few years ago, an equipment manufacturer that I have a relationship with set up a loudspeaker, an EQ and a pink noise source in their booth. Passersby were invited to try to equalize the pink noise by ear to achieve a flat response from the loudspeaker.

One person was spot on, but most of the results were pretty scary, even though the participants were all audio professionals. We need accuracy of +/-1 or 2 dB to give us a truly flat baseline to work from.

Some may have the ears to accomplish this, but most of us don’t. If you’re using an RTA, be sure to get one that displays in increments as small as 1 dB.

Again with TEF, the data of a high-frequency device after equalization. (click to enlarge)

Measuring the low frequency response of a loudspeaker minus the room’s acoustic contribution is a somewhat tricky proposition. A 40 Hz wavelength is approximately 28 feet long.

In order to measure those frequencies properly with a time windowed measurement system the window has to be long enough to contain at least one full wavelength.

Unfortunately that means that it is long enough to contain room reflections that contaminate the measurement. At higher frequencies this is not a problem because the wavelengths become short enough for the windowing to provide anechoic measurements.

Because real-time analyzers are time blind they include room reflections at all frequencies.

Testing Methods
There are a couple of ways to deal with this. One is to do your testing outdoors far enough from any buildings to minimize reflected energy.

My favorite approach comes from Don Keele, one of the really smart guys in our industry. He places the measurement microphone about one inch from the center dome of the woofer, takes a measurement, then places the mic one inch from the port, takes a measurement, and then sums the two responses.

The signal-to-noise ratio of the measurement is improved greatly because of inverse square law gains that result from being so close to the source.

A word of caution here: if the device under test has a maximum output of 120 dB at one meter, it’s output at one inch will approach 136 dB. This may be enough to do bad things to your expensive measurement microphone.

So start out at a fairly low volume level and work your way up. This test will give you a good idea of what is going on from 200 Hz on down.

The first parameter I set is a high-pass filter to prevent signals that the box is not capable of reproducing from wasting power and potentially damaging components.

Then correct any large EQ anomalies with parametric filters. The microphone can be moved to normal listening distance for this and all subsequent tests.

Make sure that the distance is at least three times the longest dimension of the box under test. This puts us in the far field. I do this test with no crossover engaged for the low-frequency device.

If you test from 200 Hz to, say, 5 kHz, you get a good idea of the total low frequency response curve of the box.

When you’re done, a configured system should look something like this on the DSP software. (click to enlarge)

Next, look at the upper range of the device’s response curve. There will be an obvious point where the amplitude drops off or the speaker gets into breakup modes represented on the test display by narrow notches and/or peaks.

Use parametric filters to flatten the response as much as possible within the useable range of the device. Choose an upper crossover point for the woofer that filters out the nasty modes and only utilizes the relatively flat part of the speaker response.

The high-frequency test is next. I prefer doing this test with no crossover engaged, however, the sweep frequency must be started at a high enough frequency to avoid damaging the driver.

Check the manufacturer’s recommendation for the lowest suggested crossover point and start your sweep there. If full range pink noise is being used as the test signal, start with the crossover engaged to protect the driver.

Using the parametric filters in your DSP of choice, correct for frequency response anomalies.

If the box is using a constant directivity horn you may need to use a shelving filter to increase the high frequency output above 2 to 3 kHz. Get the response as flat as you can across the full frequency spectrum.

Remember, if you leave a 3 dB peak in the response and it happens to coincide with a 3 dB peak in the vocal mic response, it will cost you 6 dB of headroom.

You should discover an overlapping frequency range where the low- and high-frequency devices are behaving in a fairly linear fashion. The crossover can be set anywhere within that region. As a general rule, if the horn is small, set the crossover towards the high end of the overlap zone.

Larger horn mouths provide pattern control down to lower frequencies, so if the horn is larger, you can set the crossover point lower while maintaining good directivity from the device.

Next, the levels and time alignment between the low- and high -requency sections should be set. With the mic on-axis and centered between the horn and woofer, do a full range sweep. Set the crossover outputs so the average volume level is the same across the entire frequency spectrum.

Then look at the frequency and phase response at the crossover frequency. Pretty ugly, eh?

Using an impulse response or ETC measurement, look at the arrival times for the two devices. Set the alignment delay on the DSP to eliminate the time arrival offset.

Now look at the frequency and phase again. Better? You will need to do a little fine tuning to get the flattest possible phase line.

If you’re using an RTA, this part is harder. Try inverting the polarity of the high output on the crossover. You should see a notch at the crossover frequency. Adjust the delay until the notch is at its deepest. Reset the high polarity back to normal.

If the time offset is correct the notch should disappear. If your RTA has a 1/12th octave mode, it will be easier to see.

Some real time analyzers have a loudspeaker timing analysis feature as well. Using asymmetrical crossover slopes can produce better (or worse) off-axis response.

Experiment with this if you have time, but this magazine isn’t long enough to cover all the possible permutations. Program in some brick wall limiters just before the little red lights on the amps start to dance.

Less From Two Than One?
Save these settings as a preset in your DSP of choice and repeat the process for each type of monitor wedge in your inventory as well as side fills and drum monitor rigs. You can also use these settings as a basis for multiple wedge setups.

But remember that when you use multiple cabinets of any sort, comb filtering will occur because of the time arrival differences. These peaks and notches are non-minimum phase. That means that they are not “EQ-able”. (Is that a word? It is now!)

Because of this, sometimes it’s possible to get less output with two wedges than with one.

But riders being what they are, go ahead and do a preset for dual wedge setups. The crossover and time alignment settings will remain the same, but you may get some summing in the low frequencies.

Use the RTA to check the frequency response because a TEF sweep will be too frightening to look at. If you need a preset for absolute maximum output with a particular vocal mic, try putting it on a stand exactly as if you were setting up for a show. Plug the mic into the test microphone input on the test rig.

The response will be a combination of the speaker under test and the off axis response of the microphone. EQ the response to be as flat as possible. It may not sound pretty, but it will get loud. (At least until the singer cups the microphone, sealing off the back of the cartridge, turning it into an omni).

Voila! A look at the final measurement result, courtesy of TEF. (click to enlarge)

You may also want to do presets for full range response or one with a higher frequency on the low-cut filter for vocal only. Sometimes it is beneficial to attenuate the lows for an acoustic set to avoid exciting acoustic guitars or pianos. Save a preset and switch back and forth at the appropriate times.

With the current crop of DSP devices, it’s not uncommon to find configurations like four input, eight output that work perfectly for either four two-way mixes or two three-way side fills and a cue wedge. Look for routing flexibility and the ability to store lots of presets.

But most of all, listen to the units. Audio quality varies as much with digital equipment as with analog.

It’s too easy to buy this type of device based on a laundry list of features and functions when the most important thing is great sound.

Bruce Main has been a systems engineer and FOH mixer on and off for more than 30 years. He has also built, owned and operated recording studios and designed and installed sound systems.


Posted by Keith Clark on 09/25 at 05:07 PM
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Rusty Waite To Lead EAW Worldwide Sales

EAW President Jeff Rocha has announced the appointment of Rusty Waite as VP Sales.

EAW has announced the appointment of Rusty Waite to the position of VP Sales.

Waite, a 25-year audio industry veteran, will lead the worldwide EAW sales network from his new home base in EAW’s design center in Whitinsville, MA USA.

The announcement was made by Jeff Rocha, EAW President, on the heels of the breakthrough Anya adaptive touring sound system rollout, and recent marquis EAW installations at Barclay’s Center (Brooklyn, NY USA) and the soon-to-be-commissioned Singapore Sports Hub.

“EAW is poised to leapfrog others in our space with the Anya system, and we need a powerhouse leader of our sales network to plan for, manage and compound this growth,” explains Rocha. “Rusty has just the right mix of hands-on product and industry knowledge, audio engineering and production chops, marketing savvy and business development experience to take EAW to the next level and beyond.”

“Throughout my 25 years in the industry, I’ve looked up to the EAW brand for its phenomenal sound and reliability,” commented Waite. “But as I got closer, I was flat-out blown away by the newer lines such as Anya and the QX series.

“That coupled with the fact that the majority of EAW speakers are once again manufactured in the US, our momentum is catching on both domestically and internationally, and I cannot wait to see what we as a team do next.”


Posted by Julie Clark on 09/25 at 10:22 AM
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D.A.S. Audio Manufacturers 10,000 Aero 12A System

D.A.S. Audio has announced that the company manufactured the 10,000th unit of its successful Aero 12A powered line array systems in July.

D.A.S. Audio has announced that the company manufactured the 10,000th unit of its successful Aero 12A powered line array systems in July.

The powerful D.A.S. Aero Series 2 range of line array systems is the result of the company’s continuous effort and work developing successive generations of Aero Series line array systems over the past 10 years.

The versatility of the D.A.S. Aero 12A systems has led to their use in a range of concerts staged by major stars, including world-renowned artists George Benson, Justin Bieber, Alan Parsons, Guns N’ Roses and Simple Minds.

The systems have also become regulars at festivals around the world: Ultra Music Festival in the U.S., FIB and Arenal Sound in Spain, Eristoff in India or Gods of Metal in Italy, to name a few.

The Aero 12A´s have also been on hand to ensure superb sound at a variety of non-musical events on an international scale, like the F1 European GP and Barack Obama’s campaign in the United States.

In addition to live sound applications, the Aero 12As are equally ideal for permanent installs. The Aero 12A´s have been installed in theaters, concert halls, clubs and arenas worldwide, including the Paradise Theater and Stage 48 Club in the U.S., the Broendby Hallen Arena in Denmark and the Buesa Arena in Spain.

D.A.S. Audio

Posted by Julie Clark on 09/25 at 09:34 AM
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Masque Sound Reinforces The Glass Menagerie On Broadway

Masque Sound continues its Broadway run by providing a custom audio equipment package for the groundbreaking new production of Tennessee Williams’ masterpiece, The Glass Menagerie.

Masque Sound  continues its Broadway run by providing a custom audio equipment package for the groundbreaking new production of Tennessee Williams’ masterpiece, The Glass Menagerie.

For sound designer Clive Goodwin, there were two main goals: To make the play sound as natural as possible and to ensure that the whole audience, even members seated in the very back of the mezzanine, could clearly hear Nico Muhly’s beautiful composition.

“Nico’s music sets the theme for the show,” says Goodwin. “The Glass Menagerie is a memory play about the character Tom’s recollections, and Nico’s music beautifully creates that atmosphere.

“It makes it slightly surreal, so as the sound designer, I wanted to do justice to Nico’s composition and try to make it sound even more exquisite than it already is, and Masque Sound helped make that possible.”

Goodwin’s biggest challenge was dispersing the sound on stage to the back of the theater without it sounding mic’d.

In order to do this, Masque Sound provided him with an array of Funktion-One speakers, an unorthodox speaker for the Broadway stage.

“The Funktion-One speakers that Masque Sound provided really allow me to control the dispersion, especially upstairs,” adds Goodwin. “By using the Funktion-Ones, we have the most control possible and are able to minimize room reflections that would otherwise muddy the sound.”

Masque Sound’s custom equipment package also included a Midas Pro1 compact digital console, Rode RF-Bias Shotgun mics, Sennheiser wireless microphones, XTA DP428/448 audio management system, D&B E3 speakers, Meyer UMS-1P subwoofers and a Clear-Com Intercom system.

“Masque Sound has always been great to work with,” adds Goodwin. “They work very hard and go the extra mile to make me, as a sound designer, happy.”

The Glass Menagerie is currently playing on Broadway at the Booth Theatre. The production officially opens on Thursday, September 26, 2013, and its 17-week run is slated to conclude on January 5, 2014.

Masque Sound

Posted by Julie Clark on 09/25 at 09:18 AM
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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.”
This article is provided by ChurchTechArts.


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.”

Mission accomplished.

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.

Posted by Keith Clark on 09/25 at 08:55 AM
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Tuesday, September 24, 2013

Amadeus Releases ML 28 Subwoofer

Uniquely designed, powerful new subwoofer with dual 18-inch speakers now available in European and Asian markets

France-based Amadeus has announced that their highly-anticipated ML 28 subwoofer is now shipping and available through dealers in Europe and Asia.

The ML 28 subwoofer was premiered at the 2013 Musikmesse Prolight + Sound Frankfurt Expo. The new subwoofer, fitted with dual 18-inch speakers, combines a set of unique acoustical properties with high-timbral precision and extraordinary power handling capability.

The ML 28 system is designed for use with conventional loudspeakers as well as line array systems manufactured by Amadeus, including the company’s PMX, UDX and DIVA Series, and includes standard presets that are compatible with all well-known digital crossovers.

Pricing on the new ML 28 subwoofer for Europe is € 3999.00 (VAT excluded).

One of the innovations for the new Amadeus ML 28 subwoofer is the proprietary internal reinforcement structure designed to neutralize any standing waves and to suppress energy loss caused by vibrations.

This unique construction technique creates an unmatched tonal accuracy for low frequencies, even at high sound pressure levels.  Engineers at Amadeus worked to compute these detailed and important internal physical characteristics, then built prototypes and tested them, then refined the structure until their sonic expectations were met.

Initially designed for the Chinese market, Gaetan BYK, Marketing Manager at Amadeus, describes the genesis of the ML 28.

“The Amadeus brand philosophy is built on the long-standing, close and productive relationships we maintain with our customers, both in France and around the world for over 35 years,” he explains. “The custom-made products manufacturing, following acoustical or technical issues as well as market demands, is a glorious tradition at Amadeus, which serves this philosophy.

“Originally designed for the Chinese market, the new ML 28 is the concrete expression of this ethos.”

Wymen WONG, CEO of Sign King Limited, which was appointed exclusive distributor of Amadeus brand in Hong Kong, Macau and Mainland China, adds: “The rapidly growing professional A/V equipment market aimed at rental companies and entertainment infrastructures in China, including sporting and leisure sites, live music clubs, performing arts centers and various nightlife venues, required us to propose more and more sophisticated, consistent and efficient technical solutions.

“We needed to provide our customers with a dual 18-inch speaker subwoofer, combining unmatched sonic properties, cutting-edge technology with hand-craftsmanship to make it unique. Charmed by the Amadeus history, values and savoir-faire and maintaining very close relationships with their teams, it seemed natural to ask them to develop this new subwoofer project for our market, which is going to be successful.”

As a dual speaker enclosure, the ML 28 is equipped with two 18-inch (46 cm) high-power transducers with ventilated voice coils, including high-density neodymium magnets and Double Silicon Spider (DSS) to improve excursion control and linearity.

The two drivers are positioned in direct-radiating mode. The ML 28 is also equipped with low-velocity laminar ports using progressive termination, which optimize the air streams to limit the effects of port compression and extremity diffraction. The ML 28 offers an extraordinary sound pressure level of up to 141 dB with a power handling capacity of 5.600 W at nominal 4-ohm impedance.

Revealing a proprietary manufacturing technique, applied to ML 28, Bernard BYK, co-founder of Amadeus and CEO of Atelier 33, the parent company of the Amadeus brand, explains: “This unique construction technique, resulting from a unique marriage of traditional craftsmanship and advanced technology, creates, among other things, a dramatic cut in the level of cabinet coloration of the sound using a longitudinal and transverse reinforcement crossed structure.

“It is partly inspired by the internal technicality of our custom studio monitoring systems, which result from processes borrowed from the aircraft industry and in particular the construction of the airplane wings.”

Michel DELUC, lead designer at Amadeus, adds more detail about this construction method, “It has been designed to neutralize the standing waves affecting both the sonic clarity and definition of the lowest frequencies. We achieve this through an extremely complex internal reinforcement structure, using a combination of interlocking panels arranged in two perpendicular planes, each hosting several tuned notch resonators.”

Designed to be used on the road for live events fitted with accessories to make its handling easier, the ML 28 subwoofer can also be used within fixed or long-lasting installations for amazing low-end audio reinforcement.

The subwoofer is available in a highly wear resistant black (water-soluble) paint finish, but is also available in several standard colors. Or the ML 28 can be ordered in a ‘made to measure’ finish, based on registered or non-registered colors and/or materials. To offer more possibilities, the cabinet and the acoustical fabric covering the front grill can each have a different color.


Posted by Julie Clark on 09/24 at 01:37 PM
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Monday, September 23, 2013

“David Bowie is” Art Exhibit On Tour With Sennheiser

Sennheiser help create immersive sound experience through gallery display.

After an extremely successful run at the renowned Victoria and Albert Museum in London (V&A), the “David Bowie is” exhibition is now touring other leading venues of art and design around the globe.

The next stop is the Art Gallery of Ontario (AGO), Canada, one of the most important art museums in North America.

As at the V&A, which curated this unique exhibition, audio specialist Sennheiser is working hand-in-hand with the AGO to ensure an immersive sound experience for visitors.

Sennheiser’s guidePORT audio guide system and an elaborate 3D sound installation will again be an integral part of the exhibition.

Integrating more than 300 objects from Bowie’s personal archive, including original stage costumes, handwritten lyrics, album covers and photos, the exhibition celebrates Bowie’s incredible 50-year career.

A highlight of the visitor experience is the two 3D surround simulations that evoke the sound and feel of a live concert.

Bowie’s producer Tony Visconti commented on the unique audio experience, “I swear I don’t know how they do it. This is magic to me. Things were coming from over my head, down by my feet, over my right shoulder, over my left shoulder, in front of me, to the sides of me.”

Gregor Zielinsky, International Recording Applications Manager at Sennheiser, made this exceptional sound experience possible by upmixing old mono and stereo material using a special algorithm.

Currently onsite at AGO to fine-tune the experience for visitors to the venue, Zielinsky explained the setup: “There are two multi-channel music experiences at the exhibition. No. 1 is a huge screen installation with live concert material by Bowie, and we also have a fantastic collage of Bowie songs arranged by Tony Visconti.

“The audio is played through hidden Neumann and K+H speakers.”

In addition to the 3D installations, Sennheiser also equipped the AGO with an audio guide system.

A guidePORT system will provide visitors with all soundtracks, music and video sound throughout the tour, taking them on an unforgettable journey through Bowie’s music, art, and fashion.

The system automatically plays the audio when the visitor approaches an exhibit, thus enabling the guests to enjoy an individual tour and to explore the exhibition in whatever order and at any pace.

Sennheiser’s guidePORT expert Robert Généreux is on site to install and configure the system at the AGO.

“This is a once-in-a-lifetime opportunity to celebrate a living artist whose radical artfulness of identity has had an enormous influence on art, design and contem-porary culture as we know it,” said Matthew Teitelbaum, director and CEO of the AGO. “We are thrilled to have the assistance of Sennheiser in creating a truly immersive experience, allowing our visitors to delve into the provocative genius and vision of David Bowie.”

Daniel Sennheiser, CEO of the Sennheiser Group, commented: “Through the splendid work of the V&A, “David Bowie is” has recreated the ‘universe’ of music, fashion and art that this extraordinary performer embodies.

“The exhibition has rightly been an unparalleled success in London, and I am happy that the Art Gallery of Ontario will now make this record-breaking exhibition accessible in North America.

“With its innovative architecture, the AGO is a perfect venue for ‘David Bowie is’, providing an ideal setting for this thrilling multi-media retrospective and event.”


Posted by Julie Clark on 09/23 at 12:59 PM
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