Friday, October 12, 2012
New Fillmore Silver Spring Live Performance Venue Gears Up With JBL VerTec Line Arrays
System also includes Crown Audio amplifiers and dual Soundcraft mixing consoles
The newest Fillmore venue in Silver Spring, MD is outfitted with a sound reinforcement system headed by JBL Professional VerTec line arrays and driven by Crown Audio amplifiers, with mixing handled via two Soundcraft consoles.
The 28,000-square-foot, 2,000-capacity Fillmore Silver Spring was built as a collaborative effort between the State of Maryland, Montgomery County and Live Nation.
The state and the county put up the money and Live Nation leases the facility from both on a long-term basis. Sound Image of Escondido, CA was contracted for the audio system design.
The Fillmore’s main loudspeaker system includes two columns of eight JBL VerTec VT4888DPDA powered midsize line array elements flown left and right of the stage, complemented by eight ASB7128 subwoofers across the bottom front of the stage, four VRX915M stage monitors and 10 SRX712M monitors.
The delay system consists of pairs of JBL AC28/26 loudspeakers for upper balcony fill, VIP areas at the sides of the stage, under balcony fill and for the rear bar are. Amplification is supplied by 14 Crown XTi 6000 amplifiers and Harman HiQnet System Architect software is used for system monitoring JBL DrivePack enclosures and Crown power amplifiers.
Completing the system, a pair of Soundcraft Vi6 96-input digital live sound consoles handle front of house and monitor mixing duties.
“First and foremost, the new PA system needed to sound amazing,” says Dan Schartoff, VP of production for Live Nation. “We needed a system that would be adaptable for all types of events and live performances by the likes of Deadmau5, Guns N’ Roses, Mary J. Blige, Kid Rock, Trey Anastasio, Childish Gambino…the list goes on and on.”
Schartoff noted the many criteria that must be met to achieve that amazing sound. “We needed clarity from the audio system,” he adds. “The system needed to pack a punch and have the ability to be a blank canvas for visiting engineers while also being rider-friendly.”
The two Soundcraft Vi6 consoles at the Fillmore Silver Spring also provide tremendous benefits, Schartoff notes. “We have Soundcraft consoles in many of our venues and they give the engineers a chance to store their show settings on a drive. They can make changes offline or have them ready to go when they arrive at the venue. They sound great and enable users to eliminate most of their outboard gear.”
Fillmore Silver Spring
Thursday, October 11, 2012
David Labuskes Named Next InfoComm International Executive Director/CEO
Succeeding executive director/CEO Randal A. Lemke, Ph.D. who is retiring at the end of 2012
David Labuskes, CTS, RCDD, will become the next executive director/CEO of InfoComm International, effective January 1, 2013, succeeding executive director/CEO Randal A. Lemke, Ph.D. who is retiring at the end of 2012. The announcement was made by the board of directors of InfoComm International
For more than 13 years, Labuskes has served as vice president of RTKL, now a division of ARCADIS, a leading architectural and engineering firm.
He is the founder of the company’s Technology Design Practice, overseeing the delivery of audiovisual, voice, data, wireless, environmental media, electronic security and acoustics services. Responsible for the operational, financial and marketing of technical services globally, Labuskes has led projects in the corporate, government, commercial and healthcare spaces.
Prior to joining RTKL, Labuskes served as president and CEO of Premier Technology services, a software and systems design consulting firm.
Previously he was executive vice president of Accelerated Payment Systems, an electronic payments processing firm. He holds a B.A. in International Politics and Business from Penn State University, and an MBA from Loyola University of Maryland.
“David Labuskes is a leading expert on the intersection of technology and the built environment. He is the executive who can help chart the course for InfoComm and our industry when it comes to the future of networked AV, smart building technology and more,” states Greg Jeffreys, president of InfoComm International. “His commitment to creating excellent environments will help the industry continue the quest for quality experiences for the industry’s end-customers.”
Labuskes has been a long-time volunteer with InfoComm, BICSI, National Systems Contractor Association (NSCA) and TIA, offering guidance in industry training, best practices and credentialing.
“As an integrator, I think David Labuskes is the ideal choice to share his unique insights on the future of AV in the built environment, and advise the integration community on the IT skills that our businesses will need to acquire, says Jim Ford, PE, chairman of the InfoComm Leadership Development Committee. “I believe he will be able to use his experience as an industry instructor to share practical advice that will benefit InfoComm’s heritage members – AV dealers and integrators.”
The executive search committee remained committed to finding a candidate with practical business experience. “As the president of a leading company in the live events space, it was very important to me that the next executive director and CEO of Infocomm International have a solid corporate background,” said Johanne Belanger, InfoComm secretary-treasurer. “InfoComm is a $40 million association, and keeping the organization in solid financial condition is of paramount importance if the group is to continue serving the industry with training, certification, networking opportunities and more.
“I believe David Labuskes will be an excellent steward of the association’s funds. As an MBA and a former CEO, he has the experience needed to keep InfoComm on solid financial ground.”
Labuskes will begin preliminary work at InfoComm on November 26, transitioning into the executive director/CEO position on January 1, 2013. “I’m looking forward to becoming the leader of InfoComm International, and getting to know more about the concerns and aspirations of all segments of this great industry,” he says. “Whether you are an integrator, live events professional, manufacturer, programmer, technology manager or design consultant, I am eager to work alongside you, with InfoComm’s professional staff, to make this great association even better.”
Studer 5 Console Does Double Duty For First Baptist Church At The Mall In Florida
Growth of church prompted need for a more capable live sound console
The First Baptist Church at the Mall in Lakeland, FL has a rich history of serving its parishioners since it was founded as the First Baptist Church of Lakeland in 1885.
From its beginnings with 12 charter members worshipping in a 32-foot by 50-foot wooden frame structure, the church has been steadily expanding to where it now stands on the site of a former mall (hence its name) that can accommodate almost 2,500 congregants.
With growth came the need for a more capable live sound console: a Studer Vista 5 M2, which the church uses for both front-of-house and monitor mixing.
“We simply outgrew our old console,” says Daniel Livingston, program technology supervisor at the Church at the Mall. “We now have three services on Sundays alone, which along with our pastor can include a full choir, a praise band, recorded music and more.
“We reached a point where we needed to upgrade our audio facilities, and we wanted to make sure we acquired a console that was not only up to the task but could handle our future needs.
“Input count was a major factor in choosing the Studer Vista 5 M2,” Livingston continues. “We need to accommodate more than 100 FOH and monitor inputs for a Sunday service—not only the pastor, choir and band but also pre-recorded music sources and effects sends and returns.
“The Vista 5 M2 was one of the few digital consoles we looked at that we could configure with the required number of inputs, and we went with 104 channels. Another important consideration was that we can upgrade the console in the future with even more inputs, DSP and other functionality if we need to—having a future-proof console is a big deal for a church like ours.”
Reliability and redundancy are also paramount, Livingston adds. “The console simply has to work. In addition to our own busy schedule of services and events, we regularly bring in national Christian artists like Christy Nockels, Tenth Avenue North and Avalon, so we need the console to be fully functional, 100 percent of the time,” he said. “The Vista 5 M2 has backup power supplies and redundancy and if there’s ever an issue, the console will identify the problem and you can correct it immediately. However, we haven’t had any problems. The Vista 5 M2 has been solid as a rock.”
In addition, the Vista 5 M2 is easy to use. “The Vistonics interface and the console’s meter bridge provide lots of information,” Livingston says. “Everything just seems to be right there when you need it, or else you can get to it quickly. Every knob has a label and with the console’s color coding there’s no guessing about what anything does.”
“The Vista 5 M2 sounds absolutely amazing,” Livingston said. “The preamps are incredible. The Vista 5 M2 sounds better than any analog or digital desk I’ve ever mixed on.”
First Baptist Church at the Mall
NTP Technology Introducing Penta 721 IP-Compatible Audio Interface To U.S. Studio, Broadcast Markets
Eight AES/EBU input/output channels are provided, plus up to three MADI input/outputs, two IP Audio Ethernet inputs/outputs, and an interface allowing control from Pro Tools
NTP Technology is introducing the new Penta 721 IP-compatible audio router and distribution interface to U.S. professional audio and broadcast markets at the upcoming 133rd Audio Engineering Society Convention in San Francisco, booth 1237.
“The Penta 721 is a versatile and flexible core for distribution of digital audio via AES/EBU and MADI as well as routing via Dante IP Gigabit Ethernet and optical fibre networks,” states Mikael Vest, sales director at NTP. “The interface versatility and compactness of the Penta 721 make it the audio equivalent of a Swiss Army knife, as will be appreciated by sound engineers needing to integrate audio equipment across various locations at a production venue.
“A full-bandwidth uncompressed network can be established quickly and easily, connecting an entire remote site or studio premises over a single Cat 5 Ethernet cable.”
Eight AES/EBU input/output channels are provided, plus up to three MADI input/outputs, two IP Audio Ethernet inputs/outputs, and an interface allowing control from Avid Pro Tools. A mini-module slot accommodates ST optical MADI connection or dual SFP MADI optical in/out.
The IP Audio protocol is based on the Audinate Dante digital audio network technology and will interoperate with Dante-compliant third-party products.
IP Audio routing allows low-latency tightly-synchronized transport of uncompressed signals over Gigabit IP Ethernet Layer 3 networks using-off-the-shelf switches and routers.
A total of 512 channels can be routed on a 1 Gigabit network; more if the network capacity is higher.
The Penta 721 is TCP/IP controlled via one or two Ethernet ports. IP Audio routing of large systems is manipulated via NTP Technology’s RCCore router control system which allows easy setting of connections on Gigabit IP Ethernet.
Dedicated control software compatible with Microsoft Windows can be used for controlling and setting up the Penta 721 on a unit-to-unit basis when less advanced operation is required. The Penta 721 can also be controlled via Audinate Dante IP Audio routing controller software.
Available now at $3,490 (U.S.), the Penta 721 occupies a one rack unit chassis and can be fitted with a single or main-and-redundant dual power supply.
Atlona Appoints Mark Vecchiarelli As New Director Of Worldwide Sales
Will lead company's sales activities across its retail, commercial, and professional A/V product lines
Atlona has appointed Mark Vecchiarelli to the position of director of worldwide sales, where he will lead the company’s sales activities across its retail, commercial, and professional A/V product lines through its network of distributors, dealers, manufacturers’ sales representatives, and support channels worldwide.
“Mark’s vast range of technology sales and business development experience make him an excellent fit to lead our team into the next phase of growth,” says Ilya Khayn, president and CEO, Atlona. “He has a proven track record of building strong sales organizations for both new and established technology companies, and this is just what we are looking for to take us to the next step in expanding Atlona’s global presence.”
Vecchiarelli has almost 30 years of experience in sales, marketing, and operations with sub-system, software, and semiconductor companies includes experience in building and managing global sales, support, and distribution in more than 25 countries.
Previously, he held the position of vice president of sales and technical support for RedMere and Analogix, where he accelerated design wins, percentage growth in value-per-design, and revenue growth.
Vecchiarelli has also held senior sales positions in leading semiconductor firms such as ZettaCom, AMCC, and TranSwitch. Vecchiarelli holds an MBA in strategy and marketing from Pepperdine University, as well as a BSEET in engineering from DeVry Institute of Technology.
“Atlona’s award-winning professional and commercial A/V technology is redefining the traditional economics of some of today’s most important distributed A/V connectivity trends,” says Vecchiarelli. “This makes it an exciting time for the company and I look forward to working with the team and executing on the opportunities ahead of us.”
Wednesday, October 10, 2012
Smooth Performer: Sound Reinforcement For The new eTown Hall
Alcons Audio pro-ribbon based compact line arrays deliver even coverage throughout the space
Recognized nationally as a premier radio music program and broadcast on over 300 stations in North America, eTown has recently completed the renovation of its own 17,000-square-foot specialty venue called eTown Hall, located in Boulder, CO.
The weekly radio broadcast heard from coast to coast on NPR, public and commercial stations is taped in front of a live audience and features performances and interviews with top musical artists.
The format has attracted former presidents and icons including Jane Goodall, Jimmy Carter, Sarah McLachlan, Bob Weir, Lyle Lovett, JJ Cale, Mavis Staples, Willie Nelson and James Taylor.
Founder and host of eTown Nick Forster has been working to create eTown Hall for many years.
“We’ve been dreaming of having our own space, one that could serve as a multi-purpose, media-making performance hall and community center – all powered by the sun – and now we have it. For a non-profit organization, the process of designing and building the space took some time, and we had to assemble an amazing team to get it right,” Forster explains.
The result is a space that meets the needs of eTown’s performance and broadcast criteria while paying heed to eco-friendly practices.
A recent performance at new eTown Hall.
The structure that houses eTown Hall was formerly a Church of the Nazarene built in 1922. Transforming the 90-year old building into a state-of-the-art music hall presented multiple challenges.
To meet the ambitious design requirements, which include not only the offices of eTown but also a live recording facility, multiple edit suites, a music cafe and the main performance hall, Forster chose Sam Berkow and SIA Acoustics, based in New York City, as acoustical engineers and consultants.
“Nick had a very specific vision for this new facility. The scope of that vision included multiple spaces that required isolation in a number of areas.” Berkow explains. “We started with the structure – down to the new steel beams and all new walls and floors – and then we added treatment.
Sam Berkow of SIA doing the nitty-gritty work onsite, ironing acoustical fabric. (click to enlarge)
“Then, Nick and I listened to a few different speaker boxes. Not only did the live performance room require special acoustical treatment, but we had to choose a sound system that worked well with the size of the space and the more ‘rootsy’ type of music performed.
“There were also a number of other spaces to consider. In particular, isolating the recording spaces from the live venue proved somewhat challenging, but these are the types of projects where SIA excels. At the end of the day, every detail and design consideration had to mesh with all of the practical challenges presented by eTown as an organization.”
A look upward at one of the venue’s five-element Alcons LR14 arrays.
To provide even coverage throughout the space, Berkow and Forster chose Alcons Audio pro-ribbon based compact line arrays, with five full-range LR14/90 cabinets flown per side. The LR14 incorporates two 6.5-inch woofers joined by the Alcons RBN401 4-inch pro-ribbon drivers.
A stage monitor package and small café system also utilize Alcons loudspeakers, including six VR8 and two VR12 enclosures. Both are 2-way designs that are also outfitted with pro-ribbon HF devices.
Left to right, key contributors to the project, including Sam Berkow, David Rah, Marc Nutter and Preston Smits, also of Sound Sense.
The compact nature of the monitor systems worked well in the design due to sightline considerations, as well as the desire to provide the same level of fidelity for the artists as well as the patrons.
All loudspeakers in the project are driven by Alcons ALC2 class G amplifiers with integral DDP digital drive processing modules handle amplification. These amplifiers are designed by the company for exceptional low-noise and high-fidelity to ensure that the ribbon drivers are able to reproduce the original source faithfully.
Alcons ALC2 class G amplifiers driving the loudspeakers on the project.
Sonic Sense of Denver performed the system installation, with owner Marc Nutter working closely with Berkow to measure and tune the system. On-site factory assistance was provided by David Rahn, North American sales manager for Alcons, to help achieve the final outcome.
“We spent a lot of time listening to a number of high-end systems but ultimately chose Alcons Audio for the smooth performance in the top end, which we all agreed did the best job of presenting the type of acoustic oriented performances that eTown is known for,” Berkow concludes.
Atlas Sound Now Shipping 3-Way Dual 12-Inch AH Series Stadium Horns
Provide high-output, full-range paging and musical reproduction at sports complexes, campuses, and other venues
Atlas Sound has announced that three new 3-way stadium horn models in the AH Series are now shipping.
The AH Series is designed to provide high-output, full-range paging and musical reproduction at sports complexes, campuses, and other venues.
All three new models, as with the rest of the AH Series, offer weather resistant construction, including UV-resistant molded fiberglass enclosure and a three-stage, corrosion resistant steel mesh filter system that prevents weather and unwanted pests from damaging internal parts.
The 3-way design of the new models all include dual 12-inch woofers, a 1.5-inch-exit compression midrange driver and a 1-inch-exit high frequency driver. A specially designed crossover provides frequency division and extensive driver protection for each band pass.
All models offer 8-ohm impedance and can be easily bi-amped (LF/MF+HF) by removing an exterior jumper for use with high-powered systems where greater system control and protection is required.
More information on the new models:
40- x 20-degree dispersion
100 Hz—17 kHz frequency response
750 watts RMS power
65- x 65-degree dispersion
100 Hz—17 kHz frequency response
750 watts RMS power
90- x 40-degree dispersion
100 Hz—17 kHz
750 watts RMS power
Anatomy Of A System Measurement Rig: Probes, Preamps & Processors
A look at the basic dual-channel analysis setup
Feed The Brain. The primary job of a measurement rig is to acquire electrical and acoustical signals and feed them to the processor so that it can analyze, compare, slice, dice, fold, spindle and mutilate those signals and produce multi-colored charts, graphs and the all-important squiggly lines.
“But my software can produce squiggly lines all by itself without all those bothersome wires, preamps and microphones. Isn’t that enough?”
It depends on whether you are getting paid to pose or produce results.
We shall assume that you fall into the latter category, and therefore, the reason you have employed an analyzer is to measure your system and learn something about the signals passing through it, and in turn, what your system is doing to those signals as they pass through.
Your job is to decide what you want to measure, and from that, determine what measurement signals you need.
The point here is, the effectiveness of an analyzer is tied directly to its ability to acquire the measurement signals you need — and of course, those signals must be of a usable quality* (see note) and format.
With this basic functionality in mind, and for the purposes of this discussion, we shall divide our measurement rigs into three basic parts: probes (signal acquisition), preamps (signal transmission) and processors (signal analysis).
Probes (Signal Acquisition)
Put simply, our probes (sounds so scientific) are where we grab our measurement signals. We can split this group into two types: electrical and acoustical.
Once we have determined what electrical signals we want to grab — the points in the system signal flow we want to use as measurement points – accessing those electrical signals is basically a wiring exercise, generally accomplished via patching into device outputs or by splitting the signal path.
(click to enlarge)
This is why the measurement rigs for engineers who work on many, varied systems normally include a wiring kit with a healthy selection of adapters, y-cables, impedance matching connectors and other wiring knick-knacks/doohickies (pardon the technical jargon).
When grabbing electrical signals, it is important to note that, while standard practices of splitting the signal path and routing it into your preamp/audio I/O normally does not produce noise issues (worse case: noise introduced into the signal path), it is a good idea to always be aware of system grounding and is often a good idea to carry some isolation transformers in your bag o’ tricks just in case.
OK, microphones. There, we’ve said it.
Microphones are a critical part of our measurement rig. They are our analyzer’s window onto our acoustical environment and the signals that are arriving at our audience, artists’ and our own ears.
As tiny transducers, they are also the most variant component in our measurement rigs; from mic to mic, and also over time.
In a perfect world, our microphones would act as completely neutral acoustical probes — perfectly omni-directional with razor-flat frequency response from DC to light and 200-plus dB of dynamic range.
In the world in which we actually live and work, this is sadly not the case. It is only the ideal to which our mics aspire. So let’s get real about our measurement microphones.
The short take on the measurement mics we use for our rigs is that we need need to be honest about how close to our “ideal” mic we actually need.
It is relatively simple (and inexpensive) proposition in this day and age to produce a microphone that has a good free-field, omni-directional pattern with a respectably flat frequency response between 50 Hz and 5 kHz (and reasonably flat from 20 Hz to18 kHz), and with a dynamic range that is generally usable for measurements between 30 dB and 130 dB (SPL).
For a large number of our real-world applications, that may be all you require for your rig (and you can save money to spend on other cool gear.)
The microphone costs start increasing when you:
* expand the flat FR (particularly in extending and flattening the VHF response)
* extend the dynamic range, either raising the max SPL or dropping the self noise
* require tighter overall sensitivity ranges (mic to mic)
* require exactly matched responses
* require individual measurement plots for every mic
* increase the ruggedness and environmental capabilities
All of this is to say, you always can spend huge money on a measurement microphone if you so desire, but you may not need to for every single application.
Preamps (Signal Transmission)
This section should really be called, “Preamps, Cables and Audio I/O” - but that would defeat our cute alliterative naming scheme.
Also, while “signal transmission” includes all the connecting cables in your measurement rig, we will, for the purposes of this discussion, assume they are of professional quality and functioning properly (but don’t just go making that assumption in practice — check ‘em), and focus on the preamps and computer audio I/O (interface.)
Often these two functions are combined in one device, but not in all cases. Here, we shall address the two functions separately. (Also, please read this signal path quality note.)
Measurement rigs require preamps to perform four critical tasks:
1. Allow adjustment of incoming measurement signals to appropriate levels for our computer audio interface. In determining choices of preamps, we must consider what type/level of signals we will be accessing (mic, instrument, commercial line level, pro line-level), and what type of connectors will be needed (XLR, 1/4-inch, RCA, BNC).
2. Allow adjustment of measurement signals for appropriate levels for our measurement purposes. Throughout the course of standard measurement processes, it is often desirable to be able to finely adjust the levels of multiple measurement signals relative to each another.
3. Allow measurement signal selection and routing. In many cases, you may be using multiple mic and line signals which you need to select from over the course of your measurement sessions. While one can employ the old stone-knives-and-bearskins approach of just re-patching cables on the fly, multiple, routable preamps (mixers, switchers) make the job easier, cleaner, and less error prone.
4. Provide phantom power for measurement microphones.
There are many ways that these preamp requirements can be met. In touring and permanently installed systems, it may be beneficial to build the measurement preamp requirements into the system’s existing signal preamp and routing scheme (i.e., feeds directly from the mix console or system DSP units).
It’s important however to remember requirements 1 and 2 above, and make sure that the “built-in” measurement signal feeds have their own, separately adjustable levels apart from the main system drives — we can’t very well go asking the mixer to turn up or down during a performance just to make our measurement signals happy.
Computer Audio I/O
Once we have our measurement signals, the final step along the signal transmission path is the analog to digital conversion (A/D) and the journey into the computer processor (sorta sounds like an Orlando theme park ride).
The big question: “How do we get there from here?” The most convenient path is to use the converters built into the computer, their stereo line-level inputs, however, over the past 10 years, most PC laptops have dropped that input from their built-in components in favor of a simple mono-mic input (Mac laptops still have them standard.)
If one is available to you, it is certainly a viable option as those inputs usually meet/exceed our humble requirements (again, see the note on measurement system signal path quality.)
In the all-to-frequent case that your laptop does not have a stereo line-level input, or where your measurement rig requires more than two input channels, the standard solution is an external audio I/O unit.
Over the same past 10 years (not coincidentally), there have been a number of computer audio interfaces that have come on the market that satisfy our requirements — most of which including our required preamps.
When considering an audio I/O unit for a measurement rig, the primary concerns (apart from preamp requirements) in general are:
* Physical Connection Format - USB, USB 2, FireWire (IEEE1394) 400, FireWire 800, PCMCIA card, dixie cups on strings? The question is which is easiest, any will it carry the number of signals you need. USB (1 and 2) are the most commonly available connections built into laptops and are generally the preferred connection type for simple two channel (stereo) input. USB 2 and FireWire connections are required for multi-channel input (3-plus channels).
* Audio Driver Format - Just because the signals get into your computer doesn’t mean your measurement software can use them. It’s very important to determine what driver formats your program can access (i.e. wav/wmd/mme, ASIO, coreaudio). This issue is further compounded by OS version issues and is the source of severe headaches for users and developers alike.
* Powering Mode - bus-powered or externally powered. Simple stereo USB units often utilize the buss power available via the USB connection (500 mW max). This is extremely convenient as it adds portability (no need to plug in to AC) and ease of set-up to your rig. It’s also a great feature when traveling between countries that use different standard AC voltages because the bus-powered unit gets its power from the computer, which normally utilize auto-ranging power supplies. Once you are into multi-channel I/Os, it’s pretty much guaranteed that bus power will not suffice and it will need to be plugged into local AC for power.
* Form Factor - simply put, what type of audio connectors does it have and how big is it. For those of you who need an extremely portable measurement rig, rack-mount gear is most probably too big for your requirements. A corollary to this issue then is ruggedness/roadability — sure it’s portable in size, but is it really built to withstand the transportation demands/conditions placed on it?
The proper choice of audio I/O and preamps is truly defined by the intended use for the measurement rig — what systems are going to be measured, under what condition and whether or not (and how) the rig is going to be transported. No one solution works for every user and use case.
Often, it’s preferable to field a basic set of stereo preamps and I/O, and then supplement that with additional preamps and signal routers (mixers, switchers) when the complexity of the rig and system requires.
Jamie Anderson is a founding member of Rational Acoustics, which provides training courses, hardware products/packages, and professional consulting for sound system measurement, analysis, and alignment. He has been teaching and working in the field of sound system engineering, measurement and alignment for almost 20 years. During his career, Jamie has worked as a technical support manager and SIM instructor for Meyer Sound Laboratories, as a system engineer on tour for A-1 Audio (kd Lang) and UltraSound (Dave Matthews Band), and most recently, as a product manager and instructor for SIA and EAW. Also check out the Rational Acoustics Store for a selection of many of the components discussed in this article.
Shure Announces Fall Rebates For SM Microphones And Wireless Systems
SM57 and SM58 Included in Shure rebate promotion for SM microphones and wireless systems.
Shure Incorporated has announced it is offering fall rebates of up to $40 back on select wireless microphone systems and microphones.
Customers who purchase a SM57, SM58, DMK 57-52, PGX digital wireless system, PGX wireless system, or a Performance Gear wireless system between October 1, 2012, and December 31, 2012, are eligible to receive the rebate. Also included in the promotion is Shure’s X2U XLR-to-USB signal adapter bundles, available with the SM57 or SM58.
Providing outstanding performance, reliability, and application diversity, the legendary SM57 and SM58 microphones offer clean sound and extreme versatility. The SM57 is most often relied on for musical instrument pickup and vocals, as it delivers a bright, clean sound and contoured frequency response.
Designed for professional vocal use in live performance, sound reinforcement, and studio recording, the SM58 is a global standard for performing consistently, outdoors or indoors. When combining the microphones with Shure’s X2U adapter, users have plug-and-play USB connectivity for convenient digital recordings at home and on the go.
A universal drum microphone kit, the DMK 57-52 comes with three SM57s and one Beta 52A, an ideal supercardioid microphone for the kick drum.
The PGX wireless and PGX digital wireless systems deliver tailored wireless solutions for vocalists, guitarists, and presenters, combining the trusted legacy of Shure’s microphones with state-of-the-art wireless technology. A comprehensive solution for audio engineers at any skill level, Performance Gear wireless is engineered with superior sound quality and ruggedness, while providing a hassle-free setup.
“Our customers have trusted Shure for more than 80 years, relying on our products to be consistent workhorses wherever audio performance is a top priority—on the road, in the studio, and on the stage,” said Terri Hartman, director of marketing communications, Shure Americas. “We acknowledge all of our fans—from beginning artists to top-selling performers—and extend our gratitude for their ongoing commitment to produce great sound.”
Rebates include $10 back for a SM57 or SM58 purchase; $15 back for a SM57 or SM58 X2U XLR-to-USB signal adapter bundle; and $30 back for the DMK 57-52 drum kit. Wireless system rebates include $40 back for the PGX digital wireless system; $30 back for the PGX wireless system; and $20 back for the Performance Gear wireless system.
For more information on Shure’s fall promotion and how to take advantage of the current rebates, please visit www.shure.com/rebates.
SynAudCon Releases “Sound Reinforcement For Designers” Web-Based Training Course
SynAudCon is pleased to announce the release of their 5th web-based audio training course titled "Sound Reinforcement For Designers". This web-based course covers the fundamentals of loudspeaker selection and placement to achieve acceptable speech intelligibility in reverberant spaces. It is for anyone who selects and places loudspeakers in rooms.
SynAudCon is pleased to announce the release of their 5th web-based audio training course titled “Sound Reinforcement For Designers”.
This web-based course covers the fundamentals of loudspeaker selection and placement to achieve acceptable speech intelligibility in reverberant spaces. It is for anyone who selects and places loudspeakers in rooms.
The content paves a logical path from the site survey through the design process, describing and demonstrating what must be considered each step of the way.
“We have computer room-modeling programs to crunch the numbers,” explains SynAudCon President Pat Brown. “This course teaches how to think about the room and the complex interaction between it and the loudspeaker.”
As with all SynAudCon web-based training, this course consists of 10 online training lessons that includes quizzes to ensure concepts are learned throughout the course. Each lesson also comes with a .pdf of a workbook to use to follow along with the course. And, because it is web-based, materials can be reviewed as many times as needed during the 6 week training period. The course is approved for 20 Renewal Units.
“In the practical world, you have to get in and out quickly, and collect the information you need with minimal setup time and equipment,” Brown explains. “I have spent many years involved in the development of efficient, accurate techniques for making room acoustics measurements – and I share these techniques in this course. They will change the way that you perform room/sound system evaluations and system performance verification.”
With a goal to shorten the participants’ learning curve, the course includes animations, graphics and practical examples of the material. Based on SynAudCon’s popular in-person training of the same name, the web-based version provides the flexibility to take and complete the course when convenient for the participant.
Visit SynAudCon to watch the first 5 minutes of the first lesson for a first-hand experience of the course materials. It will become apparent that this course is vital for anyone who selects and places loudspeakers in rooms.
Sound Image Doubles Adamson Systems E15 Line Source Loudspeaker Inventory
One of first American companies to invest in Project Energia has doubled its commitment
Leading touring and installation company Sound Image of Escondido, CA-based Sound Image recently doubled its inventory of Adamson Systems Project Energia E15 line source loudspeakers.
Sound Image was one of the first American companies to invest in Project Energia, and less than a year, the company has doubled its commitment. The Southern California company also carries Adamson Y18, Y10 loudspeakers and T21 subwoofers in its rental inventory.
“Sound Image is a strong beta partner and a leader with Adamson in the USA,” states Jesse Adamson, director of marketing and sales. “We are extremely happy with their contribution to Project Energia. Their positive feedback and consistent push with our product is invaluable.”
Dave Shadoan, president of Sound Image, adds, “The E15 is gaining popularity quickly among touring engineers. This comes as no surprise, as the product is well thought out and is extremely high performance.
“We’re looking forward to a series of new releases in Project Energia over the next six months. The system will be even more powerful as it expands.”
Sound Image is currently providing E15s for the Rob Zombie and Marilyn Manson North American tour.
Capital Sound Chooses Martin Audio MLA For Shakedown Festival At Stanmer Park
Capital Sound general manager Paul Timmins suggested to the company’s brand new purchase, Martin Audio MLA, for the event
Capital Sound of London supplied Martin Audio MLA loudspeakers for the large audience in attendance at the recent Shakedown Festival at Stanmer Park in Brighton, England.
Having been approached once again by Ronin Events to supply audio for the Shakedown Festival at Stanmer Park, Brighton, Capital Sound’s Charles Ellery swung into action to provide a Martin Audio system for the event.
“Initially, I was looking at using a mid size Martin Audio LC system,” says Ellery. “But I had seen the site plans and I realized we were looking at a different part of Stanmer Park for the main stage arena, and an increase from an 8,000 to potentially a 15,000 audience capacity in this arena, so I changed the spec to a large format cabinet with the Martin W8-L.”
However, Capital Sound general manager Paul Timmins suggested to the company’s brand new purchase, Martin Audio MLA, for the event.
“A few calls to Martin Audio confirmed we would have the system at the warehouse in time, so I adjusted the spec, selecting 12 MLA per side for the hangs with nine Martin Audio WS218X subs per side in a cardioid array,” explains Ellery. “Ian Colville, our technical manager and someone closely associated with our move to learning about and eventually buying the system, started working with the client to make sure we had the ideal situation for flying the PA on site.
“This involved an increase in the height of the PA towers from nine to 12 metres in height to accommodate the 12 per side hangs of MLA and get the best result for the crowd.”
Crew on the job was also changed to make sure one of Capital’s freelance system techs, Marty Beath, a recent graduate of the MLA course that took place at the company’s base in London, was on board to manage the system on site.
Beath performed system tech duties, with Joseph Pearce utilizing both Soundcraft Vi6 and Yamaha PM-5D for house consoles. Islyana van Oostende worked monitors at the controls of another PM-5D, sending mixes to d&b audiotechnik M2 wedges, Meyer Sound 600HP and UPQ side fills, plus a pair of Martin Audio 218 subs and three LMs stacked per side for the DJ monitors for Dizzee Rascal, who headlined.
Stage crew were Craig Bruce, working the patch and RF world, assisted by Nathan Hayward, Marina Martinez and Harry McCann.
“Fantastic control was achieved with the MLA system performing exactly ‘as it says on the tin’,” says Ellery. “Noise levels were set at 98dB Leq A over 15 minutes at FOH in the noise management plan by Brighton & Hove City Council due to the multiple sensitive sites to the sides, rear and in the direct firing line of the PA.
“Following discussions with the Environmental Health officer from the Council on the day, this was lifted to 101 dB, and then again later on after 7 pm to 103 dB, as it became clear the MLA was performing exactly as it should, with maximum energy where we wanted it in the crowd and no superfluous energy being directed to anywhere else on the site, or to any of the sensitive residential sites all around us,” he adds.
Peaks of 106 dB were measured at front of house (54 meters from the stage) during Katy B’s set. The noise0sensitive offsite area of Stanmer village, 700 metres away and in the direct firing line of the PA, only received a high average measurement of 61 dB (Leq A) over 15 minutes.
“This was a superb result from the new Martin Audio MLA technology,” Ellery concludes. “It proved itself on a very difficult site, which required a system that delivers power where you need it and none where you don’t.”
Tuesday, October 09, 2012
Florida Church Upgrades Sanctuary Sound System With Allen & Heath iLive
Atlantic Pro Audio worked closely with church personnel in developing a tailored solution
Grace Church in Longwood, FL recently updated the sound system in its sanctuary with a new Allen & Heath iLive T-112 digital mixing system installed at the house mix position, outfitted with an Audinate Dante card, MixPad app and iDR-48 stage box. (Allen & Heath is distributed in the U.S. by American Music & Sound.)
The new system was implemented by Atlantic Pro Audio (APA) of Altamonte Springs, FL, which worked closely with church personnel in developing a tailored solution.
“We did everything step-by-step, starting with the acoustic panels,” explains Abigail (Abby) Dolbear, worship pastor at Grace Church. “We have our service and then we have the youth service, also in here [in the sanctuary]. The youth like to have a different sound and, with the new board, we have the ability to save scenes. It’s really nice to have that flexibility.”
The Dante card has also proven popular by facilitating the ability to do multi-track recording of services, rather than just stereo.
In addition, the MixPad app has been a very useful addition to the package. “Not only do we now have the freedom to move around the room with just our iPads, but it’s now possible for our sound engineers to adjust the monitor mix from the performer’s perspective on stage,” says Dolbear.
Grace Church head pastor Clark Witten is also pleased with the results of the new system project. “We really felt like APA was working in our best interest,” he states. “We are extremely happy with the Allen & Heath iLive system, the process and the results.”
“It was everything that we wanted in the end; it was perfect,” Dolbear concludes.
Allen & Heath
American Music & Sound
Atlantic Pro Audio (APA)
Portable 2-Way Loudspeakers - What Can’t They Do?
They're great tools, but what can’t they do?
Portable loudspeakers are amazing in their versatility, able to serve as mains, fills, delays, stage monitors and much more, providing solutions for hundreds of applications in live sound reinforcement.
These 2-way miracle workers usually include an 8-inch, 12-inch or 15-inch ported woofer and a compression driver on a horn or waveguide, with dispersion (6 dB-down points) commonly at 40 degrees (v) by 90 degrees (h) or 40 degrees (v) by 120 degrees (h).
The primary purpose of this dispersion is to focus sound on the audience while not sending it to reflective ceilings.
Coaxial designs, with the compression driver/horn suspended above the cone woofer, are also a popular option.
Fans of coaxial loudspeakers like that high- and low-frequency sounds originate from the same point in space to provide more of a “point source” coherence. Coaxial models also typically take up less real estate.
Not all portable loudspeakers are created equal.
Frequency response should be able to reproduce the sound source accurately, so for speech-only applications, a response of 100 Hz to 12 kHz is usually sufficient, while an acoustic guitar-singer will be better served by a response of 80 Hz to 15 kHz and a rock band should have a response of 40 Hz - 15 kHz (or higher).
These frequency limits are typically 10 dB down or less from the level at 1 kHz. Of course, the flatter the response over the passband, the more accurate is the reproduction.
Another choice is passive or active. Passive loudspeakers need to be driven by separate power amplification and might also require additional outboard processing, while active designs have the amplifier and processing built into the box.
Many active systems are also bi-amplified, with individual amplifiers for the woofer and compression driver.
Bi-amplification can offer several advantages. Distortion frequencies caused by clipping the woofer’s amplifier will not reach the tweeter, so there is less likelihood of tweeter burn-out. In addition, clipping distortion in the woofer amplifier is made less audible.
Further, intermodulation distortion is reduced, peak power output is greater than that of a single amplifier of equivalent power, direct coupling of amplifiers to transducers improves transient response (especially at low frequencies), and the inductive and capacitive loading of the power amplifier is reduced.
Finally, the full power of the tweeter amp is available regardless of the power required by the woofer amp.
On The Stick
When portable loudspeakers serve as mains, typically they’re deployed to either side of the stage on stands.
This provides good overall coverage and also positions the loudspeakers toward the “dead” rear of the cardioid pattern of microphones, reducing the potential for feedback.
Also, they should be high enough to clear the crowd; otherwise, people in the back will hear muffled sound because the crowd attenuates the high frequencies.
Raising the loudspeakers also prevents sound from blasting the front rows of the audience.
Articulation is best if the direct-sound level is high relative to the reflected-sound level.
This is achieved by locating the loudspeakers close to the audience and aiming them to direct their sound on the audience, not on reflecting surfaces.
While placement is often done on an “eyeball” basis, sometimes a bit more preparation can make a big difference.
Using graph paper or a computer, make a scale drawing of the venue. Inside this venue, draw the loudspeakers and their sound radiation angle, both in a top view and a side view.
Experiment with angling and placement so that the maximum amount sound is directed to the sound absorbing audience.
At left, positioning loudspeakers in a central location and splaying them can help prevent phase interference and comb filtering. At right, a look at the A/B Technique.
In some scenarios, when loudspeakers are placed on either side of the stage, audience members seated near the front row can be located too far from those loudspeakers, as well as too far off-axis.
This problem is usually overcome by adding a fill loudspeaker (or a few) near the front edge of the stage in the center.
An alternative approach is to position two loudspeakers in a central location, then stack and splay them.
This helps prevent the phase interference and comb filtering that can occur with spaced loudspeakers.
Hanging & Mounting
In a theater used for drama or musicals, there might be the need to place the loudspeakers closer to the audience than a typical over-the-stage location.
Options include hanging the loudspeakers over the audience (high enough to not block the view of the stage); mounting the loudspeakers on portable stands; and mounting the loudspeakers on the side walls (if enough coverage can be delivered to the center of the audience).
Keep in mind that the flying of loudspeakers must be done by a licensed professional rigger who follows safety standards.
Delay To Extend
For larger coverage areas, there might be the need to place additional loudspeakers 50 feet or more from the stage.
This is not quite as simple as we sometimes make it out to be. It’s important to delay the signal to these loudspeakers so that sound localizes on stage rather than at the nearest loudspeaker.
This technique utilizes the Haas (or precedence) effect, which states that humans localize sound to the earliest sound-wave arrival.
For our discussion here, I’m assuming that the system has a digital processor for setting/optimizing delay settings.
Delay time - T - is critical. On the processor, set T slightly greater than D/C, where T = delay time in seconds, D = distance (in feet) from the performers on stage to the audience near the loudspeakers, and C = speed of sound (1,130 feet per second).
For example, suppose the audience members who are near the loudspeakers are 30 feet from the actors.
Delay the signal to the loudspeakers slightly more than D/C, or 30/1,130, which is .026 second (26 milliseconds).
The Rayburn loudspeaker mounting method.
This will insure the audience hears the sound as if it originated on stage instead of the nearest loudspeaker, because the first arrival of sound at their ears will actually indeed be from the stage.
The same concept applies when utilizing compact loudspeakers in performance venues to cover listening areas under balconies where the output of the main loudspeakers is shadowed.
Now let’s investigate a few unusual methods that can help in troublesome situations.
A/B Technique: Suppose you’re using a few floor microphones for sound reinforcement of actors and want the amplified volume to be as high as possible without feedback occurring.
One way to do it is to reduce the number of open microphones. Every time the number of open or “on” microphones is cut in half, gain-before-feedback (GBF) increases 3 dB.
So if you start with four mics and turn up only two, GBF goes up 3 dB. If you turn up only one mic, GBF goes up 6 dB - a big improvement.
Using fewer mics also increases clarity by reducing the pickup of ambient sound reflections.
When two mics pick up the same actor, you hear the actor’s voice doubled or smeared in time, which causes comb filtering or a hollow, muddy sound.
Ideally, one mic’s fader is turned up at a time, following the action on stage. Of course, that can be inconvenient, as the sound mixer will be occupied for the entire show. An approach called the A/B Technique makes riding the faders unnecessary.
By the way, I also call this the “one speaker per mic” method, and other names might be “one PA system per mic” or “one amp channel per mic.”
Basically, if you have three floor mics, the signal of each goes to a separate amplifier channel and loudspeaker.
The three loudspeakers are mounted close together. The number of open microphones per PA “system” is one, providing maximum GBF and clarity.
And what’s great is that all three mics can be turned up the entire time, which is much easier than trying to follow the action and riding mixer faders to minimize the NOM.
The three loudspeakers are placed close together so the audience does not hear the sonic image shift as an actor moves across the stage.
One might argue in favor of separating the loudspeakers, and having the stereo image follow the actor. However, that effect is usually valid only for people sitting near the center of the audience.
Stacking Loudspeakers: Rather than deploying loudspeakers to each side of the stage, instead mount a single loudspeaker at one corner of the audience, shooting across to the opposite corner.
Or, stack two loudspeakers (horn to horn), which narrows their vertical dispersion. (And be sure to clamp them together solidly so they don’t “vibrate” apart.)
This approach, suggested by consultant Ray Rayburn (which I also pointed out in the June 2010 issue of LSI) eliminates the comb filtering that occurs with two loudspeakers at different distances, and also produces a clearer sound with less reverb because of reduced ceiling reflections.
Hiding Loudspeakers: In some venues, there’s a desire for the loudspeakers to be hidden from view. visible.
Common tricks to conceal them, without blocking their sound, include painting loudspeakers to match surrounding walls/scenery; concealing them behind plants; covering them with thin fabric of the same color as walls/scenery; and covering them with silk or nylon so they look like artwork.
Bipole Assembly: Mount two loudspeakers back-to-back and wired with opposite polarity.
This creates a quasi-bidirectional or figure-8 dispersion pattern, dramatically reducing sound radiation at the sides. This allows placement of loudspeakers to the side of a person speaking by reducing the chances of feedback.
Unlimited Opportunities: There are also wireless options, body-worn options, loudspeakers in enclosures that look like rocks, and so on.
Listing all of the alternatives and applications of the humble yet mighty 2-way loudspeaker could fill the rest of this issue. There’s virtually something for everything, with possibilities limited only by the imagination of the user.
AES and SynAudCon member Bruce Bartlett is a microphone engineer, sound system designer, recording engineer, and audio journalist.
Unit Audio Announces The “New Unit” 16 x 2 Analog Summing Mixer
16 balanced line inputs on two D-sub connectors plus choice of 1/4-inch TRS or XLR mic level output connectors
Unit Audio has introduced the “New Unit,” a 16 x 2 analog summing mixer
It offers 16 balanced line inputs on two D-sub connectors, plus a choice of either 1/4-inch TRS or XLR mic level output connectors.
Easy setup makes this a great companion to digital audio workstation, with a small footprint that takes up very little space.
Unit Audio analog summing mixers are hand-wired, point-to-point in Nashville, TN, using quality Xicon metal film resistors and Neutrik connectors
Price of the New Unit is $299 (U.S.), plus shipping.
Unit Audio design engineer Terry Auger says, “Loosely quoting Shakespeare one might say ‘To analog sum or not to analog sum?’ This has been a point of controversy with digital recording for quite some time.
“With modern DAW software, mixing within the computer has resulted in some great sounding recordings, but I’ve long been intrigued by the concept of analog summing. I was not prepared to pay $800 or more to test that theory, so I engineered and built my own.
“Then to test the theory, I set out to see if there was any difference in the mixed sound. Much to my amazement and pleasure, I did notice a subtle but very pleasing difference in the stereo separation and placement of the instruments compared to my ‘in the box’ mixes.”
Unit Audio offers “Sound Samples” here to evaluate an in the box mix and a mix run through a Unit Audio analog summing mixer.