Thursday, November 15, 2012

James River Assembly Moves Up To DiGiCo SD5/SD10 Consoles

New consoles help accommodate growing congregation on two campuses

James River Assembly (JRA) is among the first venues in the United States to take delivery and install a new DiGiCo SD5 consoles.

Taking advantage of a limited-time “Trade Up” offer from U.S. distributor Group One Limited–and working through Special Event Services (SES)–the 12,000-member house of worship in Springfield, MO, traded up their outdated large-format digital console with credit toward the purchase of two SD5s, plus an SD10 to accommodate its growing congregation.

The consoles were also a perfect fit for their staff, offering stellar flexibility, expandable I/Os for future growth, and a streamlined audio footprint for their South Campus as well as their West Campus, which opens in December.

JRA audio director Stephen Maddox selected the SD5 for the main campus FOH after seeing its debut at the 2012 InfoComm show, adding an SD10 for monitors. An additional SD5 was purchased for FOH at the West Campus with technical engineer Brian Roggow overseeing the installation there.

JRA pulled the old consoles—which freed up loads of space—after a Sunday morning service and had the DiGiCo units up and operational for rehearsals the following Tuesday.

“With the audio system revamp, we waited because we knew it would be a larger undertaking,” explains Maddox, “and one that would entail putting in an entirely new speaker system but also modifying the architecture of the room, and updating the microphones, soundboard, etc. We moved forward with the vision of our lead pastor John Lindell in mind. It was a team effort from the whole staff here. When the previous system was designed, the church was doing a classic contemporary musical style, which included a choir and orchestra.

“Since that time the church has transitioned musically to a rock/contemporary style of worship, but the old sound system wasn’t able to keep up with the output,” he continues. “The speakers were the main culprit and got the conversation started for the revamp. Once we decided on the PA—an Outline GTO with Powersoft K8 amplifiers and XTA DSP—we moved on to looking for consoles.”

Having consistency with the consoles at both the South Campus and the West Campus—running at 96 kHz—helped JRA achieve a streamlined sonic clarity as well. “It’s not like each venue is different,” Maddox says. “It’s the James River sound. Our tagline is, ‘one church, two locations.’ We want the experience at either campus to be the same. The PM1D was a nice board at the time we bought it but it lacked the warmth and depth that the DiGiCo has. When we first turned it on, I could hear the difference. It’s a very clear console, very warm, and the clarity alone is a nice feature in itself.”

The console’s flexibility was another bonus for both engineers. “I think the DiGiCos are very easy to understand and get around on,” Roggow offers. “You can see a lot of information at a quick glance. I like the fact that they can easily be a multi-user, multitasking console. I like that much better than a single-channel control approach, too.”

“With a lot of the other manufacturers’ consoles, you’re limited by its design in how you have to use it,” Maddox adds. “Since I started programming on the DiGiCo, it’s been very much of a, ‘this is what we’re giving you, how do you want to use it?’ experience. That was extremely beneficial to me because no longer was it how do I have to do it but rather how do I want to do it. Part of that intuitiveness is being able to lay out the console like you want to, with either a musical or logical layout, for a specific part in our service; it’s very straightforward and easy to program. Some of the specific features I like are the floating inputs, allowing you to put anything where you want it. The virtual sound check is a huge performance boost allowing you to mix without people on stage.

“Also, the Set Spill option is great. Even though I have each piece of the band individually laid out in layers and banks how I want, it’s very nice to be able to spill them out and it allows me to have the whole band show up on both banks with quick access to them. The macros are a huge plus, too, giving you the flexibility to program them for any scenario you can imagine because you can virtually program anything on the desk to a Macro.”

The Waves SoundGrid Bundle also adds to the overall palette. “Having them as an option is great,” explains Maddox, “but everything onboard is pretty phenomenal and the console itself is good enough on its own—from the DigiTubes, which give you the nice dynamic, tube-quality EQ, to the onboard dynamic EQ, to the multiband compressor or built-in De-esser they all sound nice. I would be fine if I didn’t have the Waves plug-ins, but they give you more to paint with and they’re a nice option to have. The PuigChild, modeled after the Fairchild, is a favorite, or the CLA compressors. The C6 is a great multiband compressor and it’s very easy to use. The Renaissance De-Esser is good, as is the SSL Compression or the SSL Channel Strip with its compressor and EQ options.”

Along with their new in-ear monitor systems, comprised of a wired Shure PSM 600 for the six-to-seven-piece band, wireless Shure PSM 900 for the three to four main leaders and soloists, and Sennheiser EW500 G3s, (each with a stereo mix), for the five to six background singers, the SD10 at monitor world has been another seamless experience for JRA.

“I don’t think I’ve ever had such an easy experience on a monitor console,” offers Maddox. “It’s all right there and a very nice, simple process. Our in-ears are so much clearer and sound so nice. It’s been a seamless upgrade and a great experience for everyone.”

The console’s main monitor engineer, Tucker Fredock, has found a lot to love onboard the console. “The layer feature makes it simpler to move around the board,” he says “Also, since we have so many different musicians that play each week, I can save their mixes as mix presets. It’s great to have a touchscreen on the SD10. The DiGiCo Gain Tracking is very accurate since in our setup FOH has gain control and it seamlessly follows when FOH makes adjustments. I love the multiband compressors and use them on the output mixes.

“Also, because we’re running the console in 96 kHz, it brings out more of the clarity in each instrument. The drums sound fuller and the voices stick out more in the mixes, kind of like a 3D image. Overall, I think the SD10 is a better product than what we were using before.”

Group One Limited

Posted by Keith Clark on 11/15 at 06:33 AM
AVLive SoundChurch SoundNewsProductionAudioConsolesInstallationMixerMonitoringSound ReinforcementPermalink

Harman Professional Deploys Regional Approach To U.S. Sales And Support

Emphasizes local engagement, deeper support and advanced training

Harman Professional has deployed a Regional Sales Office (RSO) sales management infrastructure in the Unites States that emphasizes local engagement, deeper support and advanced training. 

The new organization is led by Mark Posgay, senior director, US sales, Harman Professional, who pointed to the market-wide shift towards a systems methodology as a driver for this new tactile approach.

“As the professional audio and systems integration markets have migrated from components, to systems, to application-engineered systems, Harman Professional provides unique value — not simply because we offer best-in-class performance at the component level for each device in the signal chain but because HiQnet makes highly sophisticated systems more uniform, intuitive and easier to configure and control,” Posgay says.

“This seismic shift to deeper integration and software control requires a strong commitment to training and support so that our customers and channel partners can extract maximum value from our offerings,” he continues. “With the addition of new staff Harman Professional is meeting this commitment head-on and, we hope, empowering our customers and partners to capitalize on a compelling technology and market opportunity.

Posgay announced the appointment of seven-year Harman veteran Jim Ure to the position of business development manager, installed sound, Eastern region. Rob Lewis and Tom Der — who previously led North America sales for Studer and Souncraft respectively — also join the new team.

The new RSO organization sees Michael Schoen appointed to serve as senior manager, national accounts, Anton Pukschansky appointed territory sales manager, (Sound Marketing West) and Bill Raimondi named senior manager, U.S. distribution & strategic accounts. Leading the new organization’s training initiative is Chris Vice, another experienced Harman Professional veteran.

“This new organization enables Harman Professional to think globally and act locally,” notes Scott Robbins, Harman Professional executive vice president of worldwide sales. “We have all of the R&D resources, manufacturing capabilities and efficiencies of a global organization and now we have put the right people on the ground to listen to the needs of customers and partners in local and regional markets. I look forward to working with Mark and his team to ensure that Harman Professional provides customers with strong technical and economic value that, in turn, support their business ambitions!”

Harman Professional

Posted by Keith Clark on 11/15 at 05:17 AM
AVLive SoundRecordingChurch SoundNewsAVBusinessManufacturerPermalink

Wednesday, November 14, 2012

Change Agent: AVB And Its Potential Impact On Sound Design

The era of fractured, proprietary networking formats will soon be a thing of the past

Live sound system designers are faced with many choices, and designing a system with digital audio networking often makes those decisions even more complex.

The open AVB standards created by the IEEE have the potential to simplify these choices by removing the transport mechanism from the equation when it comes to choosing equipment.

The AVnu Alliance is taking things one step further by providing a certification program and logo for devices that support the AVB standards. Let’s take a look at how these will change distribution for live sound, not just in the design phase, but in installation and troubleshooting, as well.

The first major change is the addition of a clear indication that an audio device correctly supports the AVB standards, [in the form of] the AVnu “N” logo. This logo shows not only that the manufacturer has implemented the AVB standards in that device, along with some additional AVnu-specific requirements to ensure interoperability with other AVnu-certified devices, but also that the device has been tested and certified at a neutral third-party test facility.

This represents a significant development in an industry that currently has very few standards certification programs, outside of electrical and safety standards required by federal law. While cinema sound has THX certification for both products and venues, and many live sound products already implement a number of relevant standards (including those created by standards organizations, as well as less formal “industry standards”), the AVnu Alliance is the first to create a formal certification program.

So, if we’ve gone this long without it, why do we need certification now? As anyone who’s worked in product design knows, simplicity on the surface often comes at the expense of complexity and advanced automation under the hood. And as networks and computerized systems become more complex, it becomes more likely that a stated requirement could be interpreted differently, or implemented in two different products in such a way that makes them mutually incompatible.

In order for professionals and consumers alike to benefit instantly from successful AVB networks, it’s vital that each product is compatible with other manufacturers’ products on the market, along with future products that haven’t even been invented yet. Certification provides an extra layer of verification to keep everyone on the same page.

Expanding Scope
But what exactly does “AVnu certified AVB” mean?

For starters, there’s the entire suite of AVB standards: protected bandwidth for time-sensitive streams, deterministic low latency, network clocking, and a defined protocol for discovery and control of devices.

These standards also define the packet format for a stream, the bandwidth optimization rules for network switches, and the information a device can report about itself so that the user can choose which connections to make, and quickly re-route individual channels on the fly without affecting the rest of the system.

And it’s not just live sound products that will benefit from certification. Since these fully-ratified standards are now part of Ethernet, they have also been adopted by other, much larger industries, including manufacturers of consumer entertainment and automotive equipment.

Participation from these other industries means that there’s an even wider range of products that need to interact. Is it likely that you will ever need to mix a rock concert in your car? Probably not, but maybe you have an iPod that you would like to listen to in your car, and then digitally connect it to your console for sound check.

But even with all these components clearly defined, it’s still possible for devices that meet the standards to not actually be able to share audio.

That’s where the additional interoperability requirements from the AVnu Alliance come in: they define a common stream and channel format that audio devices must support in order for a product to be certified. This ensures that regardless of the make of your AVnu-certified devices, as long as one can send audio and the other can receive it, you will be able to get audio flowing.

This is just one example of a number of AVnu-specific requirements that will ensure every user’s success when using AVnu certified AVB products.

In the past, system designers were often forced into certain product choices based on the transport protocol they had chosen. With all the manufacturers currently involved in the AVnu Alliance, and more joining all the time, a designer who chooses an AVB network isn’t locked into specific products or licensed IP from a single manufacturer.

This is especially relevant for large-scale systems like theme parks, where one area or attraction may be added or completely remodeled, with the rest of the park remaining unchanged.

Using an open standard allows for much more long-term flexibility when it comes to swapping out individual components, whether it’s a single processor or an entire subsystem, and still be able to interface with the existing systems, without being limited to a single vendor.

This grants a huge amount of freedom to system designers to choose equipment based on features instead of transport mechanism, and allows owners to spend money on system performance instead of licensing.

Another benefit of specifying an open standard for audio transport is that it helps establish better (grounds of communication) between the audio department and IT.

Vikram Kirby, the director of technical design at Thinkwell, a design firm that specializes in large-scale attractions, notes, “When we talked to IT about what we wanted the system to do, we were able to point them to specific IEEE standards, which was much more palatable to them than trying to describe the traffic patterns of a proprietary format.”

This ability to speak the same language established a common ground between many different departments, and ultimately led to a network design that included a complete AVB backbone.

Unifying the network in this way represents a significant reduction in infrastructure costs, since it eliminates the need to have multiple, separate networks for different kinds of traffic, which also leads to a reduction of installation and maintenance costs.

Because the AVB protocols are able to safely partition legacy traffic from audio content, the IT department can have visibility into the behavior of all network traffic, which also reduces some of the burden on audio professionals to perform their own network troubleshooting.

Stepping Forward
Using AVB streams provides optimizations in other areas as well. Because the switches are intimately involved in the audio distribution, they are able to intelligently optimize the bandwidth required by streams that are received by multiple other devices.

For instance, if a single-channel, pre-recorded announcement is sent from a playback device to a single loudspeaker, the bandwidth on the network link between the playback device and the network switch is the same as if it were distributed to an entire building’s set of loudspeakers.

Another priority for many system designers is clocking and consistent time-alignment. An AVB network includes a deterministic, visible network clock to guarantee fixed low-latency across all connected devices.

Audio networks no longer have to be designed to account for the worst-case latency, as was common practice when using legacy Ethernet. Designers can now rely on latency as low as 125 microseconds per network hop, which for small networks moves network latency almost below the threshold of consideration.

All of these changes represent a significant and inevitable step forward in networking technology for live sound, and with the certification program for network switches starting soon, and endpoint certification starting not long after, the era of fractured, proprietary networking formats will soon be a thing of the past.

Ellen Juhlin is digital products analyst for Meyer Sound.


Posted by Keith Clark on 11/14 at 06:52 PM
AVProductionFeatureBlogStudy HallProductionAudioAVEthernetInterconnectNetworkingSignalPermalink

Bose RoomMatch Facilitates Versatile System At New Iowa High School Gymnasium

New system provides sound reinforcement of sporting events as well as concerts

Electronic Sound Company (ESC), a Des Moines, IA-based AV systems integrator, recently outfitted the newly constructed gym at Webster City High School (Webster, IA) with a new sound reinforcement system cable of supporting concert performances as well as live sporting events.

The new system is headed by RoomMatch RM12060 array modules from Bose Professional Systems Division.

Specifically, ESC deployed eight RoomMatch RM12060 array modules driven by three Bose PowerMatch PM8500 configurable professional power amplifiers, along with two RoomMatch RMS215 subwoofer modules and a Bose ControlSpace ESP-00 processor.

The system, along with acoustical treatment design, also by ESC, allowed the high school to move an upcoming fundraiser concert by Iowa-born operatic basso-baritone vocalist Simon Estes from the school’s auditorium, which seats only 650, to the new gymnasium, which can seat up to 2,500.

“RoomMatch is the best speaker that I have ever used for control over the sound,” states Al Osborn, CEO of ESC. “It is a highly controllable speaker. In this situation, we needed to keep the sound away from very specific parts of the room, such as the lower parts of the walls, because as a gym we couldn’t put acoustical treatment materials on the first 16 feet of the wall height.

“So we had to have a sound system that was precisely controllable and highly directional. That system includes Bose RoomMatch array module loudspeakers.”

The ControlSpace ESP engineered sound processor provides control and audio processing in a single, expandable unit and offers eight expansion card slots accommodating a range of input/output options, including control via RS-232 and serial over IP, analog audio, AES-3, Dolby Digital, DTS and PCM digital stream formats, as well as CobraNet network audio.

The ControlSpace ESP-00 offers an asymmetrical I/O matrix capability where up to 64 digital channels or 32 analog channels can be routed and processed, a function that Osborn used to allow the sound system to operate in specific zones as needed. This further helps control the sound energy in the room, supporting the intelligibility of the sound, by minimizing reverberation.

And finally, the two RoomMatch RMS215 subwoofer modules that Osborn incorporated into the system design play a very special role at Webster City High – the school’s sports mascot, a lynx, needed its own sound effect for games played in the gym. Osborn found a recording of what he called “an appropriately angry lynx” that provided the full-throated roar the school was looking for.

“The two RoomMatch RMS215 subwoofers are just what they needed to get that extra level of excitement across,” he says.

Bose Professional Systems Division

Posted by Keith Clark on 11/14 at 04:26 PM
AVLive SoundChurch SoundNewsAmplifierAVInstallationLine ArrayLoudspeakerProcessorSound ReinforcementPermalink

Eureka Volunteer Fire And Ambulance Company Relies On Ashly Processing And Amplification

Stage Masters, a local AV integration firm, called on Ashly Audio processing, amplification, and user control to deliver a new system that is perfectly tuned, reliable, and easy to use.

The Eureka Volunteer Fire and Ambulance Company, which serves the greater Stewartstown, Pennsylvania area, boasts a fine social hall that it rents for meetings, wedding receptions, and other events.

At over 4,000 square-feet, the social hall can comfortably accommodate up to three hundred people but until recently its sound reinforcement system delivered poor coverage with marginal fidelity.

The company hired Stage Masters, a local AV integration firm, to renovate the room’s A/V system. Stage Masters called on Ashly Audio processing, amplification, and user control to deliver a new system that is perfectly tuned, reliable, and easy to use.

The new system is configured to flexibly meet the needs of all potential users and renters.

Shure wireless microphones provide users with a ready and convenient means of amplify voice and are joined by a stereo iPod input, a handful of mic- and line-level input jacks, and the outputs from the video playback system.

Shawn Fife, owner of Stage Masters, selected the Ashly ne24.24M processor based on its flexibility. For example, its I/O count is modular, and he initially outfitted the social hall’s ne24.24M with twelve inputs and four outputs.

“With the Ashly ne24.24M, I can scale the processing to the needs of a particular job,” said Fife. “Moreover, the unit is network ready, right out of the box, which makes it easy to help clients remotely.”

The Ashly ne24.24M’s modularity was especially useful in this instance, because the officials requested additional output channels after the system was already installed.

“I gave them Shure headset microphones for use on bingo nights, among other things,” said Fife. “After they used the system a few times, they changed their minds and wanted to give the caller a monitor channel to listen for bingo calls.

“Since we were already using all four of the installed output channels, they would have been out of luck were it not for the ne24.24M’s modularity. We were able to meet their request at little expense simply by adding another output card together with a small powered monitor at the caller’s position.”

The original four outputs from the Ashly ne24.24M feed four zones: the entrance hall, the kitchen, main hall left, and main hall right. Because the room is always oriented the same way for video presentations, it was possible to enhance the experience with stereo.

A new BenQ SH910 projector gives the room’s users crisp video presentations. Powered by an Ashly ne4250.70 amplifier, Pure Resonance SD4 Lay-In 2x2 drop ceiling speakers give the system impressive punch and clarity with no hot spots.

“For the power that they deliver, the Ashly amplifiers are extremely cost-effective,” said Fife. Although he hasn’t yet hooked into them as he has the ne24.24M, the Ashly ne4250.70 is also network-ready right out of the box.

User control of the system is simple and intuitive. An Ashly WR-5 push-button wall remote placed at the front of the room provides volume control of an adjacent microphone jack and computer input, as well as volume control of the system’s four wireless microphones.

A second Ashly WR-5 located in the kitchen provides volume control of two adjacent microphone jacks, as well as an adjacent iPod jack.

Ashly Audio

Posted by Keith Clark on 11/14 at 11:52 AM
AVLive SoundNewsAmplifierInstallationProcessorSound ReinforcementPermalink

PreSonus UC 1.7 Expands VSL’s Smaart Analysis Capabilities

Easily view the frequency response of a venue, quickly calculate and set delay-system timing, and verify output connectivity

On November 19 (this coming Monday), PreSonus will release version 1.7 of its Universal Control, a free update to the company’s control-panel software that offers a significant expansion of the Rational Acoustics Smaart measurement technology that is integrated into the Virtual StudioLive section.

The update will be available for download at www.presonus.com/support/downloads.

In Universal Control 1.6, PreSonus added Smaart Measurement Technology’s Spectra module to its Virtual StudioLive control/editor/librarian software for all StudioLive mixers, which is part of Universal Control. This version also gave StudioLive owners access to an RTA and Spectrograph.

With Universal Control 1.7, StudioLive 24.4.2 and 16.4.2 users gain the abilities to easily view the frequency response of a venue, quickly calculate and set delay-system timing, and verify output connectivity. (Note that these additional capabilities are not available for the StudioLive 16.0.2 due to its different architecture.)

To accomplish this, PreSonus has added three Smaart system-check wizards to VSL. To use these tools, the user connects a measurement microphone to the StudioLive mixer’s Talkback input. A high-end mic is not required; most measurement mics can do the job.

Smaart Room Analysis Wizard
The Smaart Room Analysis (SRA) Wizard is an automated process that guides you through the steps of acquiring a frequency-response trace and then overlays the resulting trace on the VSL display for a StudioLive 24.4.2 or StudioLive 16.4.2 Fat Channel parametric EQ. (A frequency-response trace is the plotted result-frequency and amplitude-of the system measurement.) The user can then adjust the parametric EQ to get rid of unwanted anomalies in the room.

The SRA Wizard can do a basic analysis, which requires a single-point measurement, or an advanced analysis employing three separate mic position measurements and averaging them together.

Smaart System Delay Wizard
The Smaart System Delay (SSD) Wizard calculates and sets the correct amount of delay time between two full-range speaker systems, using the StudioLive’s subgroup-output delays. This helps synchronize the outputs of secondary (generally, side and rear) loudspeakers with the output of the main front loudspeakers in a front-of-house P.A. system. The user can synchronize multiple secondary systems using this tool and the StudioLive 24.4.2/16.4.2 mixer’s four subgroup outputs.

Smaart System Output Check Wizard
The Smaart Output Check (SOC) Wizard verifies that your system outputs are routed correctly and are passing signal. By momentarily taking over the routing and volume control of an output and patching pink noise to it, this tool lets you quickly discover which speaker is connected where and helps you quickly get to the root of a routing problem.


Posted by Keith Clark on 11/14 at 11:36 AM
AVLive SoundChurch SoundNewsProductProductionConsolesMeasurementMixerSignalSoftwareSound ReinforcementPermalink

Sennheiser Offering Rebate Program For Wireless And Wired Microphones

Applies to evolution Wireless G3 and XS Wireless, and select wired microphones

Sennheiser is offering end-user rebates on evolution Wireless G3, XS Wireless and select wired microphones, valid on all systems purchased from an authorized U.S. Sennheiser dealer between November 1, 2012 and December 31, 2012.

During the promotion, customers can take advantage of the following rebates on Sennheiser wireless systems:

—XS and G3 LE: $25 rebate
—ew 100 (excluding EW 100 ENG): $50 rebate
—ew 300 and EW 100 ENG: $75 rebate
—ew 500: $100 rebate
Sennheiser is also offering rebates on the following wired microphones:

—e609, e614 and e835: $10 rebate
—e906, e914, e935 and MD 421: $20 rebate
—e965, Neumann KMS 104, Neumann KMS 105: $30 rebate
To receive a rebate on eligible products, customers will need to mail the following to Sennheiser following their purchase:

—A completed rebate form, which can be downloaded from the Sennheiser website at http://www.sennheiserusa.com/micrebate
—The original UPC code (no photocopies will be accepted)
—A copy of the sales receipt, dated between November 1, 2012 and December 31, 2012, from an authorized Sennheiser dealer for a product listed above.

All rebates must be postmarked no later than January 31, 2013.


Posted by Keith Clark on 11/14 at 10:59 AM
AVLive SoundChurch SoundNewsProductAVBusinessManufacturerMicrophoneWirelessPermalink

Shure Wireless Systems Called Upon For “Country Music’s Biggest Night”

Artists and engineers rely on UHF-R and PSM1000 systems for 46th CMA Awards

Shure wireless and personal monitor systems were utilized for a range of live performances by country music’s top artists at the recent 46th annual Country Music Association (CMA) Awards show held at the Bridgestone Arena in Nashville and airing live on ABC-TV.

This year, monitor engineers Jason Spence and Tom Pesa relied on Shure PSM 1000 personal stereo monitor systems for most of the live performances.

“The PSM 1000 system has become one of my cornerstones for providing a solid mix to the performers on this show,” says Spence. “Delivering clean, dynamic audio with rock solid RF has never been better.”

Specifically, Spence and Pesa used 16 channels of PSM 1000 with more than 50 receivers.

In addition, Shure endorsers Little Big Town (Jimi Westbrook, Kimberly Schlapman, Karen Fairchild, and Phillip Sweet), who won Vocal Group of the Year (their first CMA Award win), performed “Pontoon” using Shure UR2/SM58.

Several other Shure endorsers also performed using UHF-R wireless systems with SM58 transmitters, including Dierks Bentley, Luke Bryan, Brad Paisley, and Vince Gill. The Band Perry also performed, although they opted for UR2/KSM9.

Most of the other performances also used Shure UHF-R wireless and PSM 1000 systems, including those by Hunter Hayes (UR2/SM58), Faith Hill (UR2/SM58), Taylor Swift (UR2/Beta 58), Eli Young Band (UR2/SM58), Brantley Gilbert (UR2/SM58), Keith Urban (UR2/SM58), Kenny Chesney (UR2/KSM9HS), Willie Nelson (UR2/SM58), and Kelly Clarkson (UR2/SM58).

“It’s no surprise to me that a majority of the artists on this show are relying on Shure UHF-R wireless systems with the SM58 capsule for their performances,” states audio producer Paul Sandweiss before the broadcast. “The RF stability and sound quality continue to be the benchmark.”

In all, the 16 channels of UHF-R and all of the PSM 1000 systems were used during the evening’s performances, provided by ATK Audiotek.


Posted by Keith Clark on 11/14 at 09:54 AM
AVLive SoundNewsProductionAudioAVConcertMicrophoneSound ReinforcementWirelessPermalink

Grund Audio Introduces GT ‘01’ Series Loudspeakers

Feature horns that are strictly wave devices, with no throat-loading that can induce distortion

Grund Audio Design has announced the availability of the GT ‘01’ Series loudspeakers designed for fixed installations as well as for corporate presentations, meetings and other special events.

Members of the product line—the GT-601, GT-801, GT-1601, GT-1001, GT-1201, GT-1202, and GT-5301—all utilize 2-way designs with a horn (with the exception of the GT-5301, a which is 3-way).

Flat frequency response, low distortion, and wide dynamic range are attributes central to the new GT ‘01’ Series. Enclosures are manufactured from 13-ply birch, available in black or white finishes as well as natural wood that can be stained to preference.

All models utilize a horn that is also made from 13-ply birch and carved out of the baffle board. All utilize a 90-degree circular horn pattern.

Of particular note, the horn is strictly a wave device. There is no throat loading, which can induce distortion.

Another distinguishing characteristic of the new GT ‘01’ Series is their ability to deliver a consistent sound, regardless of the enclosure size. For example, a contractor may elect to deploy GT-601 enclosures (with 6-inch LF transducer and 1-inch HF compression driver) in one area of a theater while placing GT-1201 enclosures (with12-inch LF transducer and 1-inch HF compression driver) elsewhere in order to achieve greater SPL capability, the overall sonic quality of the sound remains consistent.

All models incorporate powder-coated, perforated steel grilles to protect the transducers and, for connectors, provide either NL4 Speakon or 5-way binders.

The various loudspeakers are outfitted as follows:

• GT-601: 6-inch LF transducer / 1-inch HF compression driver
• GT-801: 8-inch LF transducer / 1-inch HF compression driver
• GT-1601: Dual 8-inch LF transducers / 1-inch HF compression driver
• GT-1001: 10-inch LF transducer / 1-inch HF compression driver
• GT-1201: 12-inch LF transducer / 1-inch HF compression driver
• GT-1202: 12-inch LF transducer / 2-inch HF compression driver
• GT-5301: 15-inch LF transducer / 6-inch midrange cone / 1-inch HF compression       driver

Mounting and installing these loudspeakers is accomplished via Grund Audio hardware. All models can be outfitted with U-brackets, a swivel wall mount, or 3/8”-16 threaded mounting points.

Frank Grund, president of Grund Audio Design, states, “The GT ‘01’ Series are a terrific choice for fixed installations and are equally well-suited to portable sound system applications such as corporate presentations and meetings. These systems deliver broad dispersion and, since all models share a common horn design, they are a great choice for music reproduction, as the horn doesn’t induce any coloration typical of horn-loaded enclosures. The ability to mix various size enclosures without negatively impacting the consistency of the sound is also a significant benefit.”

All Grund Audio Design GT ‘01’ Series loudspeakers are currently available.

MSRP pricing:

GT-601:  $179
GT-801:  $349
GT-1601:  $619
GT-1001:  $469
GT-1201:  $699
GT-1202:  $1,049
GT-5301:  $889

Grund Audio Design

Posted by Keith Clark on 11/14 at 09:14 AM
AVLive SoundChurch SoundNewsProductAVLoudspeakerSound ReinforcementPermalink

Tuesday, November 13, 2012

QSC K Series For Surround Sound At Award-Winning Game Developer Demo Room

7.2 System At Treyarch, creator of Activision’s Call of Duty: Black Ops II

CCS Presentation Systems recently installed QSC Audio K Series loudspeakers with KW Series subwoofers in a 7.2 surround sound configuration for the award-winning video game developer Treyarch, creators of Activision’s Call of Duty: Black Ops II.

The new loudspeakers are part of an upgrade to the A/V system in a presentation theater at Treyarch’s studios in Santa Monica, CA.

The system comprises three K10 two-way, 10-inch active loudspeakers for the left, center and right channels, plus four K8 two-way, 8-inch active speakers for the left,  right, and rear surround channels, together with a pair of KW181 18-inch subwoofers positioned at the front of the room.

Sources are controlled by an Onkyo PR-SC5508 AV preamplifier-processor, which is integrated with a Crestron DigitalMedia matrix switcher.

“The right and left speakers are on stands, the center channel is hung from the ceiling above the screen, and the surrounds are hung from the ceiling on the sides and in the rear,” says Eric Diemert, account manager at CCS Presentation Systems’ office in Hawthorne, CA.

“I really like the QSC speakers,” he continues. “They were really simple to install and sound great. And Criss Nieman at Audio Geer, QSC’s manufacturer’s rep, was amazing. He hauled a K Series speaker system over there and set them up for a demo that blew me away.”

According to Diemert, the equipment upgrade coincided with a new theme of the décor at Treyarch’s presentation room, which measures approximately 60 ft. x 25 ft., incorporates rows of home theater-style seats on risers, and is used for video game introductions and demonstrations.

Diemert reports that Treyarch’s in-house audio engineers tuned the new system. The setup, which replaces a home theater system that was less than adequate for the dimensions of the room, is now capable of delivering the realistic sounds of warfare in the developer’s games at a very high volume, he adds: “They’ve got about 9,000 watts in there/”

CSS Presentation Systems
QSC Audio

Posted by Keith Clark on 11/13 at 05:42 PM
AVLive SoundNewsAVInstallationLoudspeakerSound ReinforcementSubwooferPermalink

NSCA Opens Registration For 15th Annual Business & Leadership Conference

Grow your business and learn new strategies to help you adapt to the fast-paced changes bombarding your business, customers and the economy now and into the future.

Grow your business and learn new strategies to help you adapt to the fast-paced changes bombarding your business, customers and the economy now and into the future. Join NSCA and industry peers at the 15th annual Business & Leadership Conference (BLC), February 21-23, 2013, at the Arizona Grand Resort & Spa in Phoenix, Arizona.

Visit the NSCA website to get all of the event details. Plus, register for the conference by January 4, 2013, and receive a $100 discount on the admission price.

Over the last 15 years, there have been many industry transitions ranging from the convergence of AV and IT, an evolving distribution model, buyouts, a more competitive bid market, policies and regulations to hiring strategic positions, engaging in profitable partnerships and much, much more. NSCA’s BLC provides business owners and managers key information on industry trends and issues, marketing tips and business tactics specific to the systems integration industry.

“Each year we bring more staff to this event because everyone from the owner to the inventory manager takes away so much knowledge,” said Danielle Hagen, General Manager of Communication Specialists Inc. “The market, the paradigms – they are all rapidly changing. We learn more about the business and take those business strategies back to the office which helps us to be more competitive and efficient in our processes.”

Sessions at this year’s event will provide attendees tactics to:
• gain a competitive advantage with strategies to accept and implement change
• influence buying decisions
• dissect contracts and their issues
• successfully utilize new technologies and create service models
• transition into diverse business models

This year the BLC returns to popular Phoenix, AZ, at the Arizona Grand Resort & Spa – the area’s only AAA Four-Diamond hotel is located near the South Mountain Preserve, the largest municipal park in the world and the most visited hiking destination in the state.

With a state of the art spa, 7 acre Oasis Water Park, stunning 18-hole golf course, and all suite rooms, the Arizona Grand is a perfect place for attendees to bring their spouses or families for a getaway.

Sponsors of the 2013 event include: Atlas Sound/IED (host sponsor); Systems Contractor News (media sponsor); Almo Professional A/V; AMX; Behringer; BIAMP Systems; Bosch Security Systems, Inc; Chief Manufacturing; Cisco; Cooper Notification; FSR, Inc; Herman Pro AV; Kramer Electronics; Listen Technologies Corporation; Magenta Research; Rauland-Borg Corporation; Shure Incorporated; Solutions360; Stealth Acoustics; SurgeX; Synnex Professional AV; Tannoy; and West Penn Wire. The event is endorsed by USAV Group.

Register today for this industry only event now through January 4, 2013, at the discounted price of $999. For more information on speakers, events, and sessions visit the website or call 800.446.6722.


Posted by Keith Clark on 11/13 at 02:14 PM
AVChurch SoundNewsProductionAudioBusinessEducationInstallationPermalink

David Cooper Joins L-Acoustics As Regional Sales Manager For Asia

Responsible for building relationships as developing sales targets in the region

L-Acoustics has announced the appointment of David Cooper as regional sales manager for the territories of China, Japan, Korea and Southeast Asia.

Cooper has developed a strong career in sales and marketing for a number of professional audio manufacturers, notably in the Asia Pacific region.

In his new role as regional sales manager, he will be responsible for building solid and supportive relationships with Asian counterparts to aid in distribution as well as developing sales targets in Asia.

An important part of this activity is maintaining and fostering business activities with L-Acoustics representatives and supporting their projects through technical and commercial advice and support.

Director of business development Jochen Frohn states, “Here at L-Acoustics, we take our appointments very seriously, and for this position we were looking for someone with management expertise who would also be an asset in the Asian region. With David’s track record of performance and technical expertise, we are confident that we will enjoy collaborating with him and are very pleased to have him on board.”

“L-Acoustics has become a leader in its field and I’ve long admired the way this status has been achieved,” comments Cooper. “My perception is that the brand has innovation and quality at its core, and this was evident during my visits to the HQ at Marcoussis. I was also struck by the very friendly, team-based, vibrant atmosphere that runs through the company, and this made a very big decision actually quite easy for me.

“I am now really looking forward to reinforcing the L-Acoustics brand values and developing strong relationships with our distribution partners in the Asian region as well as adding value to the team as a whole.”


Posted by Keith Clark on 11/13 at 09:58 AM
AVLive SoundNewsBusinessLoudspeakerManufacturerPermalink

Symetrix Extends Factory Warranty To Three Years

Symetrix announces that all products purchased on or after October 1, 2012 are covered by a three-year limited factory warranty.

Symetrix announces that all products purchased on or after October 1, 2012 are covered by a three-year limited factory warranty.

The new warranty period covers all Symetrix Edge, Radius, Solus, Jupiter, Integrator Series, AirTools, analog hardware, and user interfaces.

“We at Symetrix are proud of our reputation for building dependable equipment,” said Brooke Macomber, director of marketing. “The newly extended factory warranty period reflects the confidence we have in our product quality. Our equipment stands up to the stresses and strains of the working world with tremendous resilience.

“Consultants and contractors will find our warranty is clear-cut and devoid of loopholes. Even more importantly, Symetrix products are reliable, and are designed and manufactured to outperform industry product life expectations.”


Posted by Keith Clark on 11/13 at 08:33 AM
AVLive SoundChurch SoundNewsProductionAudioDigitalInstallationManufacturerSound ReinforcementPermalink

Nuances Of Networked Audio Transport

The pros and cons of transforming your analog signals into a digital stream

In the world of networked audio transport, there are two major categories in which a system may fall: a fully standards-based network, or a proprietary network that may or may not use standards-based transport.

Both have their advantages and disadvantages and, of course, are subject in varying degrees to the problems associated with transforming an analog signal into a digital stream and then back again.

Let’s first explore the biggest question that a system designer should ask when someone shows them a new piece of digital gear: “What is the latency as it relates to audio being transported in large networks?”

Latency, or what some manufacturers call “propagation delay,” is the amount of time that an audio signal is delayed due to digital processes including analog to digital conversions (A/D), digital to analog conversions (D/A), and digital signal processing (DSP).

For live sound applications, excessive amounts of latency can wreak havoc on the audience as well as the performers by creating listener fatigue and poorly reconstructed audio.

Generally speaking, the extent to which latency will cause a problem is a function of the ratio between the direct sound and the sound that is delayed.

In a large-scale sound system, where there is little to no direct sound, the delay does not become a problem until there is a noticeable delay between sight and sound.

Pick your audio network transport highway: good ol’ CAT5 and up-and-coming fiber optic.

However, when the direct sound is within 8 dB or so of the delayed signal, 20 milliseconds (ms) to 30 ms difference will be audible. As for performers on stage, the acceptable time is generally shorter, especially when it comes to their monitors.

Echo from a system - whether it is acoustic or electronic - contributes to performer and audience fatigue.

A performer using an in-ear personal monitoring system will be conscious of any latency in the system to as little as 5 ms to 10 ms.

Small latencies on the order of 100 microseconds (µs) can cause phasing problems that can result in high frequency roll-off when they are added back into the mix with an un-delayed source.

A Common Clock
It should be noted that all digital network components should be synched to a common clock.

The best way to achieve this is with the use of a separate word clock.

This will allow all devices in the chain to be locked to each other and thus eliminate any phase shifts that may occur between devices from un-synched sampling rates.

With advances in digital technology, latencies for audio gear have dropped dramatically, to the point that some digital mixers are below 3 ms for analog in to analog out with no added time for DSP.

The problem lies in cascading multiple digital devices together if not digitally linked. When having to make A/D and D/A conversions for every piece of digital gear in a signal chain, it’s easy to see how the latencies from conversion alone can add up to an unacceptable total.

Two key factors in comparing digital audio networks are the sampling rate and the bit resolution that the network supports.

The sampling rate is the ”rate” at which a digital device ”samples” the composite analog sound waveform over time.

The sampling rate is important for the way digital audio can describe the frequencies in a sound.

Bit depth, or resolution, describes the potential accuracy of a particular piece of hardware or software that processes audio data. In general, the more bits that are available, the more accurate the resulting output from the data being processed.

For example, audio recorded with a 48 kHz sampling rate and 24 bits of resolution will have 48,000 measurements of which there are 16.7 million different values that each measurement can be per second.

Sampling rates and bit resolutions are important to digital audio networks because they need to remain consistent to keep the latency to a minimum.

A sampling rate conversion typically takes the same amount of time as an A/D conversion.

The major limitation on the number of channels a network can support is bandwidth, which is the amount of information that can be sent down the chain at one time.

The bandwidth required to pass one channel of audio varies with the sampling rate and the bit depth selected. As both increase, so too does the bandwidth demand reducing the maximum number of channels which can be transported over a particular network’s architecture.

Well, what if you could have just one A/D conversion at the start of the signal chain and one D/A at the end with all the other gear still in the chain?

Now we’re talking about the digital audio networks. Again, there are two groups of these digital audio networks, so let’s take a closer look at both.

Published Rules
A fully standards-based network abides by a published set of rules, which specifies a recommended interface for serial digital transmissions. This includes the description of data format for transport and the method of transport.

The best part of a standards-based audio network is that any manufacturer can implement the format into their products without having to pay the often-expensive licensing fees of a proprietary solution.

This allows the use of “manufacturer A’s” reverb unit with “manufacturer B’s” digital console without the conversion to analog, resulting in a total system latency that is much lower than if a D/A and A/D conversion was necessary.

The most common standard is AES3, more often called the AES/EBU (Audio Engineering Society/European Broadcasting Union) standard. AES3 uses a 110-ohm shielded twisted pair cable to send two channels up to 300 feet. The standard allows up to a 24-bit resolution with no maximum sample rate.

Another standard developed by the AES is AES10, more commonly called MADI (Multichannel Audio Digital Interface).

MADI offers 64 channels at a 48 kHz sample rate and 32 channels at a 96 kHz sample rate with a resolution of up to 24 bits per channel. Transmission over a single 75-ohm coaxial cable has a limitation of 150 feet, but the use of fiber-optic cables can extend the length to two miles.

Other multichannel standards include ADAT Optical (ADI), Sony/Phillips Digital Interface (S/PDIF), and Tascam Digital Interface (TDIF). See Table 1.

The only latency added into standards-based digital audio distribution is the act of making the A/D and D/A conversions and any subsequent DSP that occurs. This is unlike some proprietary solutions that also require additional time to transcode and transport the data.

Viable and Affordable
There are a multitude of proprietary systems in the ever-evolving marketplace which use the standard IEEE 802.3 Ethernet protocols.

Ethernet is used because it’s relatively inexpensive, reliable, and the technology has been, and will continue to be, upgraded by the computer industry.

There was a time when 10baseT was the best thing going. Now, gigabit (100baseT) Ethernet has become viable and affordable.

Table 1: A point of format comparison.

The problem with Ethernet is that it is a non-deterministic system, meaning that the data will arrive when it feels like it.

However, several manufacturers have developed systems that make the arrival times very predictable and allow for the network to be synchronized with only a small margin of error.

For instance, one standard’s master unit regularly broadcasts beat packets onto the network either from its internal clock or an external master clock.

Other devices on the network which utilize said standard lock onto the arrival time of this packet and regenerate the clock locally. The error in clock delivery is ±1/4 sample period; this translates to about .005 ms at a 48 kHz sampling rate.

While digital seems to be the future of audio networks, it still requires attention to detail and proper setup.

Monitoring overall system latency and keeping a consistent sample rate and bit resolution are both new requirements for the digital age.

The answer to the question of what system is best for your situation is still the one that best meets your needs, which might be good old-fashioned analog.

Digital is just another hammer in the sound designer’s tool belt which can either drive the nail home if used correctly or smash a thumb if its requirements are ignored.

David A. McNell, CTS,  is an AV engineer working in the Special Technologies Group of Newcomb & Boyd, Atlanta. He is a Certified Technology Specialist (CTS) and an EIT.

Posted by Keith Clark on 11/13 at 04:23 AM
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Monday, November 12, 2012

Hosa Technology Debuts USB-200FB Series High-Speed USB Cables With Pivoting Connector

Design facilitates space-saving right angle connections

Hosa Technology has announced the introduction of the USB-200FB Series high speed USB cables with pivoting A connector designed to conserve valuable space in tight surroundings.

Available in 3-, 6-, and 10-foot lengths (USB-203FB, USB-206FB, and USB-210FB), the new USB-200FB USB cables feature a pivoting Type A connector—the end that typically connects to a computer—that can be set to either straight or right angle positions.

This enables one to use the cable in its straight orientation when space permits and in a right angle position in cramped quarters.

In its right angle position, the computer would typically be placed on an elevated laptop stand (so the cable can hang off the side) or at the edge of the work surface.

For DJ production rigs commonly consists of a laptop computer, mixer, turntables, and a digital controller, the new USB-200FB USB cables can be a valuable means of making connections in an environment where space is frequently limited.

To ensure interoperability with a wide range of USB peripherals, the new Hosa USB-200FB USB cables are fully compliant with the USB 2.0 serial bus interface standard and are backward compatible with the USB 1.1 standard. They also support burst data transfer rates up to 480 Mbps.

As a result, the new cables are a good option for connecting an audio interface, USB microphone or instrument, or most computer peripherals to a PC.

“The new Hosa USB-200FB USB cables provide DJ’s, musicians, and others with a valuable means of making equipment connections in the cramped spaces they frequently find themselves working in,” says Jose Gonzalez, Hosa Technology product manager. “At Hosa, we’ve been providing the cables musicians and audio pros require for over two decades, and these new USB products address a common challenge we’ve all encountered at one time or another.

“All three cable lengths feature our unique connector pivot design, exhibit superior workmanship throughout, and carry pricing that is comparable to most common USB cables lacking these features. I’m certain these new USB cables will be well received by our industry.”

The new Hosa Technology USB-200FB USB Series cables are expected to become available in January 2013.

MSRP pricing:

• USB-203FB: $8.95
• USB-206FB: $10.95
• USB-210FB: $11.95

Hosa Technology

Posted by Keith Clark on 11/12 at 03:36 PM
AVLive SoundRecordingChurch SoundNewsProductProductionAudioAVInterconnectSound ReinforcementStudioPermalink
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