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Wednesday, February 08, 2012

Indiana Officials Issue Safety Violations For State Fair Stage Collapse

The Indiana Department of Labor today cited concert and event production company Mid-America Sound Corp. of Greenfield, IN with three safety violations in the collapse of an outdoor stage at the Indiana State Fair last August just prior to a show by the country band Sugarland. Seven people died and 58 were injured after the stage collapsed when a gust of wind toppled equipment that hung over the stage.

CBS News also reports that Mid-America Sound Corp. has been issued a $63,000 fine, being the company that provided the stage rigging and chose the workers to erect it.

“The evidence demonstrated that the Mid-America Sound Corporation was aware of the appropriate requirements and demonstrated a plain indifference to complying with those requirements,” Commissioner Lori Torres stated in the report.

The department also issued a small fine against the Indiana State Fair Commission for “failing to conduct proper safety evaluations of its concert venues,” and the International Alliance of Theatrical Stage Employees (IATSE) Local 30 also came under fire, accused of five workplace violations.

Sugarland was not penalized, with the agency noting that the band didn’t employ the workers and wasn’t responsible for building the stage.

One stagehand, Nathan Byrd, was among those killed in the collapse. At least nine other union members were injured.

Further investigations are ongoing. The State Fair Commission has also hired Thornton Tomasetti, an engineering firm based in New York City, to investigate the rigging collapse, while Indiana Governor Mitch Daniels has hired Witt Associates, a public safety and crisis management consulting firm, to conduct a “comprehensive, independent analysis.”

The CBS News report is here.

One of many videos of the stage collapse is here.

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Virtual Sound Checks Without A High-End Digital Console

Here are a few ways to get it done

 
Here are some thoughts on doing virtual sound check if you don’t have a DiGiCo or Avid digital console at your disposal.

Disclaimer: This is not going to be exhaustive. There are hundreds of hardware/software combinations that will get you the same result. These are some ideas only.

Also, it should be noted that “cheap” is a relative term. All of these solutions are going to cost money, real money.

However, if you church is serious about raising the level of audio technician performance, it’s money well spent. On we go…

First, let’s define “virtual sound check.” It is simply the ability to record the band with each channel on it’s own track and then being able to play that recording back, in place through the same channels on your console.

To illustrate with a very primitive example, let’s say your “band” is a worship leader with an acoustic guitar. To facilitate virtual sound check, you would need a way to record the vocals and guitar on separate tracks, and you want those sources to come off the board before any EQ or dynamics.

Typically, you’re using direct outputs or the insert outputs. When you get ready to practice, you do a little patching (in software or hardware) and play back that recording through the same channels you use if the worship leader and his guitar were live in the room.

One thing should be immediately apparent here; the bigger your band (and the more sources you have), the more elaborate the system you’re going to need for virtual sound check. If you are running 30-40 inputs every weekend, this post is really not for you as that system is not going to be cheap.

Rather, I’m focusing on those who run fewer than 24 channels per weekend (a number that is not arbitrary, as you’ll see in a minute) and using an analog board. Here are a few ways to get it done.

Audio Interface(s)
The simplest way of doing this job is with a USB or more likely a FireWire interface such as the M-Audio ProFire 2626, a Focusrite Saffire Pro 40 or similar interface with 8 analog inputs and 8 analog outputs.

The first thing you’ll notice when shopping for an interface is that manufacturers get very creative in the way they count I/O. For example, the ProFire 2626 is listed as having 26 inputs and 26 outputs, which it does. But only 8 of them are analog.

And if you’re using an analog console, that’s all you care about. If you have a digital console with ADAT I/O, you gain you an additional set of 8 useable channels.

Now, the catch here is that there aren’t any interfaces with more than 8 channels of analog I/O (at least I can’t find any). So that means if you’re running 12 channels of audio, 4 get left behind. Unless you get creative. You might ask why you can’t just connect two 8-channel interfaces to your computer and send those inputs to your recording software.

The issue is that most DAW software won’t support multiple I/O devices simultaneously. If your DAW of choice doesn’t support multiple I/O devices, there is a workaround, at least on the Mac.

In Audio/MIDI settings, you can create what’s called an Aggregate Device, which allows you to create a virtual device that is made up of two or more actual devices. You then chose the Aggregate Device as your I/O source in your DAW, and all the inputs and outputs on all devices that make up the Aggregate Device are available to the DAW.

So an example system might be made up of two Focusrite Saffire Pro 40 interfaces combined into an aggregate device and recorded using Reaper on a Mac Mini. That would give you 16 channels of recording and playback for around $1500, give or take. That seems pretty reasonable; at least until you consider the next option.

Hard Disk-Based Recorders
There exist on the market a couple of hard drive-based recorders, most notably the Alesis HD24. This little 3-rack-space wonder is capable of recording or playing back 24 tracks of 48 hHz, 24-bit audio.

The HD24 has 24 channels of analog I/O (plus 24 channels of ADAT I/O) and costs about $1600. Really, this is the way to go. It requires no computer, is simple to set up and operate and is rock-solid reliable. Add 24 channels of TRS patch cables and you’re done.

Other options include the Tascam X-48, which is a full-blown 24 channel workstation (and almost $5,000) and the excellent, but somewhat pricey JoeCo BlackBox, which will set you back almost $3,000 by the time you add a drive.

Caveats
There are a few caveats with any of these solutions. First, if your board has direct outputs, it’s a fairly simple matter to patch those direct outs to the inputs of whatever recording solution you use.

Getting back in, however, will require some re-patching. You’ll want to pull your mic inputs, and patch the outputs from the recorder or interface(s) into the Line Inputs on your console.

If you don’t have direct outs, you’ll need to use the inserts. One cool thing about the JoeCo BlackBox is that the inputs are normaled back out to the outputs during every operation except playback.

That means that for recording (or just sitting there), the insert signal is returned and you can continue to use the board normally. When you hit “Play,” it opens the normal and sends the recorded signal back to the return on the board. From a user interface standpoint, that’s really nice. However, it will cost you twice what an HD24 costs…

When using the inserts, you will likely need to push the cables into the console until the first click. An insert jack is a TRS (tip, ring, sleeve) connector, so it has 3 contact points. Most consoles use the ring as the send, so if you push a TS cable in to the first click, you get the equivalent of a direct out (albeit an unbalanced one). Pushing it in all the way will interrupt the signal, so you’ll only do that on playback.

Using inserts is going to mean a fair amount of patching and some experimenting, so don’t decide to try this out at 8:50 on Sunday morning.

Once you get the system up and running like you want, start recording your services in all their multi-track glory. Then during the week, you can practice and experiment just like the band is there, only they aren’t.

Keep in mind, you won’t have any acoustic energy coming from the stage, so things like drums and vocals will be a little different. But this is still a great tool for training and experimenting with various processor settings.

Like I said, this isn’t exhaustive; I only intended to give a few examples. Hopefully though, it will get you thinking about how you can implement a virtual sound check system in your church.

 

Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.

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Tannoy Unveils VLS Series Passive Column Array Loudspeakers

Tannoy has introduced the VLS Series passive column array loudspeakers offering a balance of performance and cost, when active beam-steering may neither be required nor affordable.

The VLS Series is the first Tannoy product to incorporate FAST (Focused Asymmetrical Shaping Technology), which delivers unique acoustic performance benefits. Central to this is its asymmetrical vertical dispersion, gently shaping the acoustic coverage towards the lower quadrant of the vertical axis. By the nature of a typical application, an “ideal” column loudspeaker should be biased in the vertical plane, toward the audience and away from reflective surfaces above (like ceilings) which are detrimental to intelligibility.

FAST also facilitates quicker, easier installation with less need for tilting or specific concern for optimal mounting height. Mounting is handled via supplied wall brackets.

Tannoy has packaged this performance in a slender and narrow profile, aesthetically refined, powder-coated aluminum chassis with curvilinear aluminum grille. Each model is available in either black or white as standard, with custom RAL finishes available at additional cost and lead-time.

Three models are available – VLS 7 (7 × 3.5-inch LF) designed for speech-only applications, VLS 15 (7 × 3.5-inch LF with 8 × 1-inch HF) and VLS 30 (14 × 3.5-inch LF and 16 × 1-inch HF), both of which are designed for more demanding full-range applications as well as speech.

All are IP64 rated for dust and water ingress and are salt spray and UV resistant as well as subject to rigorous high/low operational temperature and humidity testing.

Specification is aided by the addition of an exclusive Tannoy edition of EASE Focus v2.0 software, allowing systems to be designed with predictable results, along with the ability to specify VLS Series in conjunction with Tannoy’s existing column loudspeakers – including I Series and QFlex.

Tannoy

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Hosa Technology Mogan Elite Omni Earset Microphone Now Shipping

Hosa Technology has announced that the new Mogan Elite omni earset microphone, the newest addition to the Mogan brand of subminiature microphones, is now shipping.

The Mogan Elite earset mic is outfitted with a moisture-resistant, 2.5 mm omni-directional capsule with -45 dB nominal sensitivity that is designed to be positioned farther from the user’s mouth.

Delivering full-frequency audio performance (20 Hz – 20 kHz) and high gain before feedback, this microphone provides a natural, resonant sound quality.

The new mic also offers an innovative earpiece designed to be worn comfortably for extended periods. With a fleshy ear cushion concealing its fully adjustable, sprung-steel (stainless) mechanism, the mic feels natural when worn over one’s left or right ear.

An interchangeable cable system allows connection of the mic to most popular wireless transmitters, including models from Shure, AKG, Sennheiser, and Audio Technica. Each mic ships with a detachable, Kevlar-reinforced cable with a hardwired connector.

The new Mogan Elite Omni Earset Microphone is available in either beige or black to blend with a variety of skin tones. Additionally, each unit includes a foam windscreen and a single mic clip. The entire package ships in an impact-resistant, compression-molded neoprene zippered case.

Jonathan Pusey, Hosa Technology director of sales and marketing, states, “The Mogan Elite earset microphone delivers impressive performance, enabling this microphone to be right at home in a number of high-end applications, including broadcast and theater, in which audio performance is critical. In addition to world-class audio quality, the Elite earset mic is very comfortable to wear and may be worn without distraction for hours on end.

“The earpiece is highly adjustable and the boom mechanism facilitates precise positioning of the microphone’s capsule. I am quite confident the Elite earset will be right at home in a number of demanding audio environments.”

MSRP for the Mogan Elite omni earset mic is $400.

Hosa Technology

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Enter The PSW Sweepstakes To Win An Audio-Technica Microphone Or Headphones

Enter to win an Audio-Technica microphone or headphones in the first PSW Sweepstakes of 2012.

ProSoundWeb is giving away three Audio-Technica 50th Anniversary Limited Edition products each month in January, February and March.

Specifically, for each drawing, we’re giving away:

1st prize - AT4050/LE Multi-Pattern Condenser Microphone
—Special 50th anniversary edition in silver-colored metallic finish with etched-on serial number and carefully crafted wooden carrying case
—Transparent uppers/mids balanced by rich low-end qualities combine with advanced acoustic engineering for extensive performance capabilities and highest quality
—Dual-diaphragm capsule design maintains precise polar pattern definition across the full frequency range of the microphone
—The 2-micron-thick, vapor-deposited gold diaphragms undergo a five-step aging process so that the optimum characteristics achieved remain constant over years of use
—Three switchable polar patterns: omni, cardioid, figure-of-eight
—Transformerless circuitry virtually eliminates low-frequency distortion and provides superior correlation of high-speed transients
—State-of-the-art surface-mount electronics ensure compliance with A-T’s stringent consistency and reliability standards
—Switchable 80 Hz hi-pass filter and 10 dB pad
—Custom shock mount provides superior isolation
—Valued at $995.

2nd prize - ATM25/LE Hypercardioid Dynamic Instrument Microphone
—Exclusive 50th anniversary edition in silver-colored metallic finish with serial number etched on the surface
—Ideal for kick drum, toms, and other highly dynamic instruments
—Handles very high SPL at close range
—Big, warm low-frequency response with excellent presence
—Multi-level grille and rugged construction
—Offers very full sound on close-up vocals and dialogue
—Corrosion-resistant contacts from gold-plated XLRM-type connector
—Rugged, all-metal design and construction for years of trouble-free use
—Valued at $489

3rd prize - ATH-M50s/LE Professional Studio Monitor Headphones
—Special 50th anniversary edition in silver-colored metallic finish
—Exceptional audio quality for professional monitoring and mixing
—Collapsible design ideal for easy portability and convenient storage
—Proprietary 45 mm large-aperture drivers with neodymium magnet systems
—Closed-back cushioned earcup design creates an outstanding seal for maximum isolation
—Adjustable padded headband for comfort during long mixing/recording sessions
—Single-sided straight cable terminates to gold-plated mini-plug with screw-on 1/4-inch adapter
—Valued at $209

Go here to enter the latest PSW Sweepstakes. Note that entrants are asked to register to receive the ProSoundWeb Daily e-newsletter.

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Tuesday, February 07, 2012

Revolabs Enhances HD Control Panel For Entire HD Line Of Wireless Microphone Systems

Revolabs has announced that the company’s Windows-based HD Control Panel software has been enhanced to support the entire HD line of wireless microphone systems, bringing the monitoring and configuration tools found on the Executive HD to the HD Single/Dual Channel and the HD Venue systems.

In addition, based upon customer feedback, Revolabs has created several new features for the HD line, including a DIP switch display, mute groups for Executive HD systems, and an expanded control system API.

With the HD Control Panel, users can monitor and control networked HD wireless microphone systems from a single PC software program with an intuitive graphical user interface.

The HD Control Panel allows users to control the mute status and gain of each microphone, and to lock out presenters from using the mute button.

The software also provides the ability to monitor each microphone closely for its real-time status, such as battery level.

The monitor tab of the HD Control Panel has been enhanced to provide the DIP switch status for each system, eliminating the need to look on the back of the system to see which switches are active.

Revolabs has also added several commands to the HD systems’ API, allowing A/V control systems to send global commands, turn off microphones, and even initiate pairing, all from the convenience of a room’s touch panel.

Finally, Revolabs has bolstered the Executive HD with the ability to assign systems to mute groups. This allows all systems in a building to be bussed together without muting each other, unless they are assigned to the same group.

“We are pleased to bring the capabilities of the HD Control Panel to users of our HD Single/Dual Channel and HD Venue systems, in addition to offering powerful new features across our entire HD line,” says JP Carney, CEO of Revolabs. “We take pride in listening to our customers as we continually strive to meet their evolving needs. New features, such as those released today, are a direct result of customer feedback.”

The enhanced HD Control Panel and new features are available through a firmware update (version 2.6.1) to both the base station and microphones. The update is available now at www.revolabs.com/downloads.

New feature enhancements require a Gold unlock code provided as part of a Revolabs service plan. Any system that has previously been unlocked will automatically receive the new features upon completion of the firmware upgrade.. 

Revolabs

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Church Sound: How To Transition From Analog To Digital Mixing

A digital mixer is a whole new way of doing the same old things

 
I’m in the process of helping one of my churches transition from an analog mixer to a digital mixer.

They were in need of more channels than their Allen & Heath 16-channel MixWiz with some outboard gear (front of house EQ, couple of compressors, effects unit) could provide.

Based on the maximum number of channels that they anticipated needing over the next five years, I recommended the PreSonus StudioLive 24.4, one of the least expensive 24-channel digital mixers on the market.

The church has two audio volunteers that are pretty much average in their knowledge of sound and sound systems so this would be a typical transition for a lot of churches in the 100-400 person attendance range. Volunteers selected more for their willingness to serve than their knowledge of audio. I know that nothing has been touched with the front of house EQ, compressors and FX since I helped them set it up about a year ago.

Some things that you need to consider in this transition is how uncomfortable the volunteers are going to be until they make the paradigm switch from the analog WYSIWYG (what you see is what you get) to the digital layers.

Depending on the digital board, layers control everything from different grouping of faders (1-8, 9-16, etc) to control over the aux sends, FX, etc. Outboard gear usually goes away and everything is now handled with the digital mixer. It’s a big transition and you shouldn’t minimize it, but treat it with care and planning and the transition will go smoothly.

Getting Started
What I recommend is that the digital mixer not be put into service immediately but be brought into a two-to-four-week training duty cycle. It requires some mics and cables as well as a couple of speakers for monitors and front of house stand-ins. If you have instruments that you can plug in that helps as well. Keep the existing analog system going as the production system until everyone has been trained and is comfortable with the digital board.

Before you start with the digital mixer, make sure everyone has reviewed the user manual. A digital board is a computer with knobs and faders and is significantly more complex than an analog mixer. While they are pretty robust, you can still mess them up and repairs can be costly.

An Investment of Protection
One thing to invest in if you haven’t is a top-line power conditioner like those from Furman. I also recommend a computer UPS (battery backup) from a company like APC or Tripp Lite. Get a decent capacity one. The reason is that because a digital mixer is a computer, when power is interrupted you can’t just switch it back on like an analog mixer. You need to boot it up and, depending on the mixer, that could take anywhere from a minute to several minutes.

Having a UPS unit, the mixer will stay powered on, so even if the rest of the system is knocked offline by the power interruption, when the power comes back on, the mixer will still be up.

Unboxing The Mixer
Once you get the mixer unboxed, check for any damage. If everything looks good bring all faders down to minimum and turn on the mixer. I like to let the mixer “burn in” for about four hours with nothing going on or plugged in just to let all the electronics warm up to full operating temperature. This will check to ensure that nothing is shorting out. Be aware of any burning electrical smell or smoke. If you detect either one shut the mixer down immediately and unplug it. Contact the vendor.

Preparing For Training
The StudioLive is close to an analog board in that all the channel faders are on one surface as opposed to layers. This makes the transition somewhat easier. All effects, aux send levels are controlled through the center “Fat Channel.” That will be where most of the confusion is going to come in so be prepared to spend a lot of time going through this area.

The StudioLive is set up pretty easy so I was able to figure 85% of the board out without looking at the manual. There are also a ton of video tutorials on the PreSonus site and YouTube that can help with anything to do with the board. But for volunteer sound techs it will be a bit of a challenge.

Building A Mini-System
Hook up a mic to channel 1 on the mixer and hook up a speaker to aux send 1 and to front of house. This will be the basic training setup.

Once you get it hooked up, bring up the gain to an appropriate level. A digital board is less forgiving about exceeding the 0 level than an analog board before going into clipping so run the level less than needed for training until you get comfortable with the way the board handles signals.

Don’t worry about EQ settings or FX yet. All you want to do is to learn the signal flow from the channel to the aux send and FOH.

Once you’ve figured out how to adjust the aux send levels for the channel and you can adjust FOH level you’ve gotten over the initial hump.

Using EQ
The next thing you’ll want to learn is how to adjust EQ’s for each channel. Depending on the digital mixer you’ll either have a screen that will have a parametric equalizer, or in the case of the StudioLive, you’ll have the knob adjustments for high, high mid, low mid and low bands. As with all digital mixers you are able to set the frequency points for all these bands as well as the Q, which is the width of the frequency adjustment. This is a lot more adjustability than what an analog mixer has and is worth spending some time practicing.

After the channel EQs get figured out you’ll want to adjust the front of house EQ. On the StudioLive it’s set the same way that the individual channel EQs are set. One nice advantage about digital mixers is that most of them have a library of preset EQs that you can start with. The StudioLive has built in a nice set of professional quality EQ presets that are good enough to leave alone and assign to each channel.

The other nice feature of digital boards is the ability to save all your settings to a scene. So you are able to set up multiple scenes for different worship teams or different instruments and recall them just by dialing up the scene and pressing the load button. So no more needing to reserve channels based on who’s playing that day.

Enter Effects
The power of digital mixers means that you can assign FX to each and every channel, both to auxes and to front of house, so you’ve got a lot of flexibility. Just remember that just because you can doesn’t mean you should. Less is more, at least in the beginning. Some boards give you more FX capabilities than others. The StudioLive offers two channels of FX, others more.

Multi-track Recording
Another advantage that digital mixers have is that they usually provide some form of multi-track recording capability. In the case of the StudioLive, it’s provided by a FireWire port into the provided Studio One software. This means you can record each channel separately into your computer, as long as it has a Firewire port.

One very cool reason for doing this for the worship team is the ability to do what’s called a Virtual Sound Check. What that means is that you don’t need the worship team there to set up the board. You can play back the individual tracks back into their respective board channels and use those tracks as the sound check.

Then, once the band gets in, sound check is very minimal. It’s also a great way for the sound team to train on the board and allows them to massage settings without needing the musicians.

Saving Scenes
Once you get everything set the way you want it remember to save your settings to a scene. I usually recommend naming the scene with the church name and 1. That way you can always recover your baseline settings.

Sound techs should create their own “sandbox” scene, which allows them to manipulate settings and save it to their own scene without affecting the master scene. Make sure that no one other than the lead sound tech saves to the master scene.

Once you’ve got the master scene saved it won’t matter what changes people make to the board during the week. Bringing back the master scene will only require a quick push of a button, and in the case of the StudioLive, resetting the gain and adjusting the faders. In other digital boards, gain settings and fader positions are saved within the scene.

Making The Switch
Once the sound techs are comfortable with the digital board then it’s time to switch out the old analog board with the new digital one. Check all your settings. Be sure any settings you change are saved to the master scene once you’re happy with how everything sounds.

Finally, when you shut things down, do NOT shut things down by just turning off the power conditioner. This WILL damage the digital mixer. Follow the shutdown procedure in the manual. It can be anything from just powering off the mixer with the mixer’s power switch to a shut-down procedure on the screen.

Summary
A digital mixer is a whole new way of doing the same old things. It’s exciting as well as terrifying for volunteers, so go slow. Take it one step at a time and ensure they are comfortable with the new system before putting it into production. You’ll achieve a seamless transition and have fun doing it!

Brian Gowing has helped over 30 churches meet their technology requirements. Brian works towards shepherding the church, analyzing their technical requirements, sourcing the equipment, installing the equipment and training the volunteer personnel.  As he likes to say, “equipping the saints with technology to help spread the Good News.” Contact Brian here.

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Joe Peavey And Steve Spittle Join QSC Audio

QSC Audio Products has announced the addition of two new members to its professional team, with the appointments of Joe Peavey to the position of product manager, software and Steve Spittle to the position of business development manager.

Peavey will be working with the Q-Sys team to identify and define improvements and additions to Q-Sys software functionality as well as providing high-level technical support. He has a lengthy background in the installed sound market by his work with the family business, Peavey Electronics, specifically working in manufacturing, tech support and finally product manager of the MediaMatrix line of DSP products.

Since leaving Peavey Electronics in 2006, Peavey has focused on creating hardware and software solutions for various audio manufacturers and consulting services for integrators in the U.S. and Canada.

“In the many months since my first interactions with the company, QSC continually amazes me with their attention to the market, their workforce and quality,” says Peavey. “I am proud to join forces with an organization of their caliber and reputation on a product at the top of its game.”

Spittle, in his new role at business development manager, will focus on expanding opportunities for growth in the company’s integrated systems business. He was previously western U.S. sales manager at Avid, and a vice president/owner at Millar Electronics, a manufacturers’ rep firm located in the southeastern U.S.

“QSC makes great products and cares about its customers,” he says. “I’m looking forward to working with this dynamic team to continue to build on this foundation for growth.”

Spittle is located in QSC’s Costa Mesa headquarters, while Peavey is located in the company’s satellite offices in Boulder, CO.

QSC Audio

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Mojave Audio Debuts MA-101SP Matched Pair Cardioid Condenser Microphones

Mojave Audio has introduced the new MA-101SP, a matched pair of MA-101fet cardioid condenser microphones for use in a range of stereo recording and live sound reinforcement tasks, with instruments such as drums and guitar amplifiers, as well as capturing room ambience and general stereo recording.

Each MA-101fet in the matched pair provides warm, full-bodied reproductions of instruments without the shrillness and high frequency artifacts so often encountered with modern condenser microphones.

The microphone’s warm FET circuitry and externally polarized capacitor mic element combine to deliver low noise and high quality performance.

The MA-101fet features both omni and cardioid polar patterns by way of interchangeable capsules and is outfitted with a 3-micron thick, .8-inch diameter gold sputtered diaphragm.

As one would expect from a David Royer designed microphone, each MA-101fet in the MA-101SP matched pair offer performance specifications that are impressive. Frequency response is 20 Hz - 20 kHz (+/- 3 dB), sensitivity rating is -40 dB (1 volt per pascal), and the distortion rating is less than 1 percent @ 120 dB SPL (-15 db pad off) and less than 1 percent @ 135 dB SPL (-15 dB pad on). The microphones operate on standard 48-volt Phantom power.

Mojave Audio president Dusty Wakeman states, “The new MA-101SP matched pair of microphones is the result of countless requests from the audio community. Drawing upon the strengths of the MA-101fet, these mics are a terrific choice for a wide range of stereo recording tasks where imaging is critical. 

“Engaging the 15 dB pad allows one to take advantage of the fast transient response on instruments such as snares, toms and loud guitar amps. The MA-101SP is a remarkably versatile general purpose recording and sound reinforcement tool that, I’m confident, will find a home in a wide variety of environments.”

The new Mojave Audio MA-101SP ships in a single carrying case that includes a stereo bar. MSRP is $1,195, and availability is Q1, 2012.

Mojave Audio

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Monday, February 06, 2012

Meyer Sound Promotes Miguel Lourtie To European Technical Services Manager

Meyer Sound has announced Miguel Lourtie as its new European technical services manager, where he will supervise the company’s technical support team in Europe and assume primary responsibility for sales support and design services in the region.

“Customer support is paramount at Meyer Sound,” says John Monitto, Meyer Sound’s director of technical support worldwide. “Our customers expect an extremely high level of technical expertise and customer service. With his outstanding technical skills, customer rapport, experience in the field, and fluency in several languages, Miguel is a great fit to lead our technical group in Europe.”

Lourtie joined Meyer Sound European technical services in 2007, and has played a vital role in supporting a number of major Meyer Sound projects across the continent, including the Mantziusgården Culture Center, Montreux Jazz Festival, and the Grimaldi Forum. He also serves as a seminar instructor as part of Meyer Sound’s extensive education program.

Prior to joining Meyer Sound, Lourtie founded Lourisom, an audio consulting and distribution business in Portugal and previously a Meyer Sound distributor.

“To ensure a seamless show, high-quality audio tools and the person driving the system are equally crucial,” says Lourtie. “The Meyer Sound tech support network has some of the best sound engineers in the industry, and I look forward to working even more closely with them to help our customers get the best out of their Meyer Sound equipment.”

Lourtie will continue to be based in Lisbon, Portugal.

Meyer Sound

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Full Compass Systems Appoints Jim Ripp As Assistance Sales Manager

Full Compass Systems has named Jim Ripp as its new assistant sales manager, bringing a wide range of music industry related sales and management experience to the role.

Ripp studied at the University of Wisconsin - School of Music with a dual degree in Piano Performance and K-12 Music Education. While there, he began working at Forbes-Meagher Music Company as a sales/general manager, and also served as director of education. 

In addition to handling sales and accounting functions at Forbes, Ripp managed a team of 18 and developed music training programs for youth and seniors. 

In 1993, Ripp began working concurrently for Falcetti Music Co. as a store manager, sales representative and teacher, which had him managing a team of employees and teachers while gaining experience in sales, customer service and technical support.

Roxanne Wenzel, vice president of sales and marketing for Full Compass states, “Jim is a great fit for our organization. His skills and experience will greatly complement the sales management we already have in place and help us continue our double-digit growth.”

Full Compass Systems

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Church Sound: How To EQ Speech For Maximum Intelligibility

The problem is unclear words are a distraction from the message

 
Haven’t we all had stories of misheard words? It could have been a song lyric or you misheard your spouse?  Maybe they mumbled a word or it just wasn’t clear what was said. This has been the cause for a few hilarious moments at our dinner table. 

The problem is unclear words are a distraction from the message. 

In the church environment, the pastor’s words must be clear. We can ensure this maximum intelligibility through proper speech EQ.

There are four topics to consider when it comes to the EQ’ing needs for the spoken word. 

1. Microphone location. We are fortunate in that most pastors now use wireless microphones. This means that the distance between the mic and their mouth is pretty consistent.  In the case of the headpiece, this is especially true.

In the case of the lapel mic, remember they should drop their chin to their chest and put the mic directly below that point. Long ago, I was taught “a fist away from the chin.” The point here is that we want the best sound isolation we can possibly get while having a good gain structure in place. 

Remember, the closer to the source, the more the proximity effect comes into the equation and you’ll need to EQ out some of that added bassiness.

2. The speaker’s natural voice. Just as every guitar has a unique sound, so does every person. You want to bring out the best qualities of their voice. You don’t want them to sound like a different person. Their vocal characteristics are also “what you have to work with.” 

This means you’ll need to know how to deal with quiet speakers, bassy talkers, and nasally preachers, just to list a few.  Not everyone has a great radio voice.

3. Presence of background music. Depending on your church, your pastor might talk with a running soundtrack. There is definitely an art to being able to play the right music for this. 

However, any type of music bed means you now have to make a space for the voice amidst the instrumentals. Instrumentals can easily blur the spoken word so you’ll have to plan on tweaking the EQ for the musicians as well. 

4. The environment. Just because a vocal boost at 400 Hz sounds good in one room doesn’t mean it will sound good in another room. One of myreaders runs audio outside…in Egypt. Any EQ work must take the environment into account. The settings for a “quiet room” won’t be the same for an echo-y room or an outdoor venue.

Now that we’ve got those out of the way, let’s turn to…

The Frequency Make-Up Of Speech

Our speaking voice has three frequency ranges that need to be understood:
1. Fundamentals. The fundamental frequencies of speech occur roughly between 85 Hz and 250 Hz.
2. Vowels. Vowels sounds contain the maximum energy and power of the voice, occurring between 350 Hz and 2 kHz.
3. Consonants. Consonants occur between 1.5 kHz and 4 kHz. They contain little energy but are essential to intelligibility.

In short, this means that the “power” of the voice does not equate to the intelligibility of the voice. Think of it like this…just because a person has a booming voice doesn’t mean they are easy to understand. 

Now that you understand the audio dynamics (fundamentals, etc) in a voice and the environmental concerns (background music), let’s turn to…

What You Can Do To Provide The Maximum Speech Intelligibility For Your Pastor

There are three things you can do for tackling the EQ’ing process for the spoken word:

1. Make room for the voice. As I mentioned above, the environment makes a difference in how you EQ the spoken word. We can only control what is coming into the mixing board, so wind and rain aside, let’s talk about music. 

Mixing a large band means making space in the sonic spectrum where each instrument/vocal can sit and sound unique; and of course then blending these sounds together into a tight mix.

The spoken word needs the same treatment when music is played underneath it. This can happen in two ways—

—A. Adjust volume. This can be done using compression or simple volume adjustments. The general rule-of-thumb is the music is there to support the spoken word – to sit underneath it.  Therefore, look to cut volume levels of instruments before you boost the volume of the speaker. You can also use compression to bring volume levels up and down as you wish.
—B. Adjust the mix. Cut the frequencies of the instruments where they are the same as that of the speaker. Boost the spoken word EQ in those areas a little if needed to present the music and the voice as two distinct sounds.

2. Know sibilance and how to avoid it. Sssssssibilance in vocals is when the sound of the letter “S” sounds more like a hissing snake. You can accentuate vowel sounds/add presence by increasing the EQ in the 4.5 kHz to 6 kHz. 

However, the “S” sound lives between 5 kHz and 7 kHz. Therefore, be careful when adding presence because you can easily go from a great sound to a hissy sound.

3. Focus on vocal quality. There is no simple 1-2-3 process to EQ’ing the spoken word. Therefore, take these points into consideration:

—Roll off the low frequencies if the proximity effect is causing unusual bassiness.
—Don’t roll off so much low end as the voice loses some of its umph. Yes, I’m using “umph” as a technical word.
—Boost in the 1 kHz to 5 kHz range for improving intelligibility and clarity.
—Boost in the 3 kHz to 6 kHz range to add brightness. This can help with speakers with poor intonation.
—Boost in the 4.5 kHz to 6 kHz range to add presence. Note that too much boosting in this area can produce a thin lifeless sound.
—Boost in the 100 Hz to 250 Hz for a boomy effect.

In Case Your Head Is About To Explode From Information Overload, Remember:

—The above points can contradict each other. There is no hard and fast rule. Mixing is as much an art as a science. Trust your ears over everything else.
—It’s possible that once you EQ the vocal channel that it’s a little lacking in the low end. Boost it a bit give it that full sound. Again, trust your ears. Close your eyes and ask yourself if it a) sounds natural and b) sounds clear.

Finally

EQ’ing the spoken word is about improving the quality of the sound so it sounds clear, is easy to understand, and sounds natural.

So much of our mix time goes towards the band. Make sure you spend those few crucial minutes working on the pastor’s vocal as well. 

Church was about the sermon long before music, skits, and cool videos rolled onto the scene.

Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.

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PreSonus Adds New Control Options To StudioLive Mixers

PreSonus has announced new updates to its StudioLive Series digital mixers, including a number of features not found on any other digital mixer from any manufacturer. 

New features include:

QMix. Up to 10 musicians can simultaneously control their PreSonus StudioLive monitor (aux) mixes using an iPhone or iPod touch and PreSonus’ QMix app, a free download from the Apple App Store. QMix/VSL is the only solution that allows multiple users to each control their own aux from separate iPhones.

Smaart Engine Technology. PreSonus has begun incorporating Rational Acoustics Smaart Measurement Technology for sound-system analysis and optimization directly into PreSonus Virtual StudioLive remote-control/editor/librarian software.

With Smaart technology and VSL, you’ll be able to precisely identify nasty feedback frequencies and get your loudspeakers to play nicer with the room-all without having a degree in acoustical engineering.

The first version of VSL to incorporate Smaart technology will be part of PreSonus Universal Control 1.6, which is expected to be available later this spring.

Universal Control 1.5.3 and StudioLive Remote 1.2. Universal Control 1.5.3 features an improved version of Virtual StudioLive that supports the new QMix iPhone app, including QMix permissions (so that each user controls only one specified aux mix) and the ability to name aux buses.

Universal Control 1.5.3 also adds VSL features that work with PreSonus StudioLive Remote 1.2 for iPad to enable SL Remote permissions so that iPad users can only control front-of-house mixer features or a specified aux. Tap tempo has been added to both VSL and StudioLive Remote.

VSL adds the ability to copy and load channels, copy main mix to aux mix (and aux to aux), link channel faders so that they can move together, and make your StudioLive mixer default to Fader Locate Mode once a fader has been adjusted in VSL or in StudioLive Remote for iPad.

PreSonus

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Sunday, February 05, 2012

RFvenue Releases New Long-Range UHF Antenna For Wireless Mic & IEM Systems

RFvenue has introduced the new high-gain, foldable, circularly polarized CP Beam antenna for use with wireless microphone and in-ear monitoring systems used in applications such as concert touring, location sound, audio/visual, and broadcast.

The CP Beam antenna is optimized for long-distance applications in the 470-698 MHz UHF range.

“The patent-pending CP Beam is a convenient, easy-to-use, full-size beam antenna for long-range applications that can be very quickly utilized when needed,” states RFvenue CEO Chris Regan. “Customers wanted a compact and durable high-gain antenna without the size, weight, and additional cost of hard plastic or metal designs.

“It’s great that it folds up for storage, yet is quickly deployable. There’s no need for a dedicated flight case to store the CP Beam – it fits in a two RU rack drawer when folded.”

Robert J Crowley, inventor of Crowley and Tripp microphone technology and chief of Soundwave Research, which operates RFvenue, adds, “The new RFvenue antennas all incorporate ergonomic and human factors that have been ignored in the past. RFvenue’s products make wireless systems easier to use, more dependable, and eliminate guesswork in an increasingly complex RF spectrum.

“The CP Beam is a high-gain, broad bandwidth, directional antenna that is excellent for long-range wireless mics or IEMs, as well as point-to-point RF links.”

The new CP Beam has a $499 list price. The company’s products are shipping worldwide through distributors, dealers, and a nationwide manufacturer’s rep force in the US.

RFvenue

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Friday, February 03, 2012

Audio Video Electronics Implements Tannoy QFlex At St. Frances Basilica

Home to one of the oldest Catholic congregations in the United States, the Cathedral Basilica of St. Francis of Assisi is sometimes referred to as the heart of Santa Fe, and for good reason. Although the Cathedral Basilica was dedicated in 1887, the site has been a focal point of worship for the community since 1610.

The first church built where the current structure now stands was destroyed during the Pueblo Indian Revolt of 1680, but another was built to replace it in 1714; a portion of which still stands within the existing building – a small adobe chapel dedicated to Our Lady La Conquistadora housing the oldest representation of the Virgin Mary in the nation.

In addition to serving the spiritual needs of it’s own congregation, the Cathedral Basilica’s rich history attracts approximately 100,000 visitors annually. As beautiful as the Cathedral is, however, for some years it has had a problem, says Wanda Vint, Director, Development and Donor Relations at the Cathedral Basilica. Put bluntly: “You couldn’t clearly hear the word of God.”

With the August 2011 installation of a state of the art sound system that depends heavily on a pair of Tannoy QFlex digitally steerable column arrays that’s all changed.

The major thrust behind the project was a 2009 visit made by the Cathedral’s Rev. Monsignor Jerome Martinez y Alire to the Basilica of Saint Mary in Minneapolis. Surprised that such an old and similarly acoustically reverberant space could sound so good, he asked who had designed and installed the system and was referred to Minnesota-based, Audio Video Electronics (AVE).

The project that initially caught the Monsignor’s ear was undertaken before QFlex was available, explains Kevin Crow, AVE’s VP of sales and marketing, but both spaces had similar issues: “In the Cathedral Basilica of St. Francis of Assisi, the RT was 5 to 6 seconds in the mid frequency band.”

In order to meet the Cathedral Basilica’s needs, Stefan Svard, AVE President and system designer, specified a pair of Tannoy QFlex 40s. Placed on a pair of columns roughly 6 feet above the floor just in front of the altar, the QFlex provide coverage to approximately 75 percent of the 1200-capacity, 90 - 65 foot space. They also provide low frequency support throughout the nave, the south transept, and the Our Lady La Conquistadora chapel.

Additionally, smaller Community Entasys arrays were installed as rear fills for the nave, and to provide reinforcement for the chapel and other ancillary spaces.

Basic EQ, tuning and system commissioning was done via Tannoy’s proprietary VNET software, with the processing handled by the onboard DSP within each QFlex, Svard says, but the system also incorporates a Lectrosonics Aspen DSP to handle mic mixing for the Earthwerks FM series podium condenser microphones specified by AVE, and matrixing for both the QFlex 40 and the additional fill speakers. The Lectrosonics Aspen is controlled by an iPad, which allows users to adjust volume levels easily depending on the type of service in progress, how much of the space is in use and the number of congregants present at any given time.

Naturally, the Cathedral Basilica’s atmosphere had an impact on the choice of QFlex, as did the ability to diagnose any issues the church might have using the QFlex array’s remote monitoring capabilities. But the main reason for choosing QFlex, Svard says, was experience. When he first heard QFlex he was cautious in his assessment. After a shootout with a competitor’s product in a St. Cloud, Minnesota house of worship, however, his opinion changed.

“We’ve done a number of Catholic churches, going back 6 or 7 years, using various steerable array products. Every product has strong points and weak points, but in that case, Tannoy’s QFlex was the clear winner.”

“If I’m in the front, middle or back of a room, the EQ that I need to correct is the same,” he continues. “Other products I’ve used shift in character. QFlex is the only product of its kind that retains its frequency response – its spectral consistency – across its coverage pattern.”

The result is a dramatic improvement in speech intelligibility and the sound quality of both background music and live performances by the Cathedral Basilica’s choir. Still, Vint was concerned some parishioners would not welcome the technology, particularly those who were uncertain they needed a new system, or that it might detract from the church’s historic atmosphere. “But the sound is so clear, we haven’t had any complaints at all.”

Monsignor Jerome Martinez y Alire is equally satisfied: “The sound quality is incredible, as is the appearance of the loudspeakers themselves. We were concerned about how modern speakers would look in such an old, historic church – with custom paint finish to match our walls they all but disappear. The clear, audible sound is a gift to our parishioners and visitors alike.”

Tannoy

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