Mixer

Tuesday, June 28, 2011

Peavey Pro Audio Amps The King Of Wake Professional Wakeboarding Tour

King of Wake isn’t just demanding on the athletes—it presents a demanding sound reinforcement environment for audio engineers, as well

With its dramatic aerial acrobatics, wave-hopping stunts and adrenalized production, professional wakeboarding is a summertime extreme-sports tradition.

Since 2005, the King of Wake championship tournament has tied the sport’s most prestigious contests together into one ultimate tour to see who rules the wake.

But King of Wake isn’t just demanding on the athletes—it presents a demanding sound reinforcement environment for audio engineers, as well. With single events drawing up to 10,000 spectators, there is a high-power need over a large audience area, all under extreme heat and humid conditions.

The King of Wake tour has proven the perfect match for Peavey Electronics’ vast line of professional audio loudspeakers, processors and microphones, as well as control systems from Peavey Architectural Acoustics and power amplifiers and mixers from Crest Audio.

“King of Wake is a non-stop, high-energy professional wakeboarding competition tour,” said Chris Bischoff, King of Wake Competition Director. “We demand the highest performance in all areas of production to provide our fans with an unforgettable experience.”

“The addition of a professional sound system from Peavey Electronics keeps our athletes and fans amped with total audio clarity, full-range sound reproduction, and more power than we’ll ever need.”

The centerpiece of the King of Wake system is a stage where a DJ-and-drummer combo performs for the crowds, powered by seven Crest Audio Pro 7200 amplifiers and backed by four Peavey SP 4 loudspeakers and two Peavey SP 218 subwoofers through a Crest Audio X 20R rack-mount mixer.

A pair of Peavey Impulse 12D powered loudspeakers—featuring Peavey’s revolutionary IPR amplifier technology and true ribbon drivers—serve as monitors for the performers.

Announcements, event emcee and program music are handled through the stage P.A. and two flight lines consisting of six Peavey PR 15D powered loudspeakers each. A Peavey Architectural Acoustics Digitool processor provides loudspeaker management and control for the entire system, flanked by a Peavey CEL-2a compressor/limiter/expander, Peavey Dual DeltaFex multi-processor and Peavey QF215 and QF131 graphic equalizers.

In addition, the entire event operates on Peavey professional wired and wireless microphone systems, including the PVM 480 condenser and PVM 45ir dynamic mics, PVM DMS-5 drum mic system and Pro Comm U-1002 wireless systems.

The NBC Sports Group is televising the 2011 King of Wake series this summer on the VERSUS network to nearly 80 million homes. Remaining dates on the King of Wake tour include July 9 in Monroe, Wa.; July 27-31 in Minneapolis, Minn.; Aug. 18-21 in Indianapolis, Ind.; and Aug. 27-28 in Knoxville, Tenn.

Peavey Electronics’

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Posted by Pro Sound Web on 06/28 at 03:33 PM
Live SoundNewsPollAmplifierConcertLoudspeakerMicrophoneMixerSound Reinforcement • (0) CommentsPermalink

Church Sound Files: Back To The Basics

The only thing that will be consistent about our mixes is that they will be hit or miss on a consistent basis

I was at an event recently where the mix was, shall we say, less than spectacular.

But it also wasn’t horrible. Let’s just say it was a little above bad and a little less than O.K.

In reality it was mainly one thing that was really grating on my nerves: every time the lead singer would really get into his mic, some level distortion would occur. 

I wanted to get up and go back to the mix position and turn the gain down on the preamp!

What I have found in situations like this is that the person mixing;
1) Has an ego problem and is not open to suggestions;
2) About 70 percent of the time achieves a reasonably good mix, but when something changes or goes awry has no ability to correct or fix the problem;
4) Knows little about the basics of sound;
5) Turns out to be a musician who wasn’t good enough to be on stage. (Sorry!)

Now before all of my musician friends jump all over me, let me say that some of the best mixes I’ve heard were put together by accomplished musicians that turned sound operators because they loved mixing.

The big issue is a lack of understanding of the basics.  I mean, come on, and gain structure 101!  If the gain structure issue was fixed in the situation I mentioned above, I would say the mix would have been much better than acceptable.

I know that it’s sexier to play with effects, work the EQ, and add dynamics… But really, if we don’t have the basics down, the only thing that will be consistent about our mixes is that they will be hit or miss on a consistent basis.

Here are two very basic things to always keep in mind:

1) Understand the essentials of gain structure. There are many articles written on this including this one right here on ProSoundWeb.

2) Use the right mic, and put it in the right position. This requires an understanding of pickup patterns, the differences between condenser and dynamic mics, and also where to place them on instruments. 

My current pet peeve is the sound operator who mics the clarinet with the mic up riding the bell! There is something about that nasally tin whistle sound that happens when close mic’ing that bell that makes my hair stand on end!

Anyway, simply doing these two things help us deliver solid mix on a consistent basis. It’s a matter of the basics!

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.

 

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Posted by Keith Clark on 06/28 at 09:19 AM
Church SoundFeaturePollBusinessEngineerMixerSound ReinforcementTechnicianAudio • (0) CommentsPermalink

Wednesday, June 22, 2011

Enveloping The Audience: The Audio Approach For Adele In Concert

“Adele’s vocals are amazing. I want the audience to forget who they are for a moment and be able to project themselves solely onto what’s occurring onstage." - Dave McDonald, front of house engineer

For the loyal fans of Adele, the time has come for the pop chanteuse’s name to be added to a pantheon of British soul divas like Amy Winehouse, Sade, Annie Lennox, and Dusty Springfield.

Currently crisscrossing North America in support of her sophomore album 21, Adele Laurie Blue Adkins indeed deserves such a distinction, based upon the sheer power of her own voice and honors such as two 2009 Grammy Awards for Best New Artist and Best Female Pop Vocal Performance for her song “Chasing Pavements.” 

Adele’s front-of-house engineer Dave McDonald knows a thing or two about chasing along on pavements himself, currently catching his sleep to the sound of steel radials humming on the asphalt while journeying to venues like Philadelphia’s Electric Factory, NYC’s Beacon Theatre, the Commodore Ballroom in Vancouver, Chicago’s Riviera, and the venerable Ryman Auditorium in Nashville, where the North American leg concludes in late June and then moves to the U.K.

The veteran Brit mix engineer is joined by monitor engineer Joe Campbell, as well as tour manager David “Zop” Yard, production manager Pat Baker, and stage techs Adam Newman and Adam Carr.

Adele’s front of house engineer Dave McDonald showcasing his Allen & Heath iLive-112 prior to a recent show in Toronto. (click to enlarge)

Intentionally small by design to offer her hardcore fans a truly intimate experience, the tour isn’t traveling with “racks and stacks,” instead opting to utilize a different rented house system at each stop. “In situations like this you can come across some real crones of audio,” McDonald observes with an appropriate dose of dry English inflection, downplaying any hint of concern about his reliance upon PA du jour. “It keeps you sharp every night. You have to be on your toes always, there’s absolutely no room for complacency.”

Traveling Light
Complacency is one thing, and confidence is quite another, and it’s indeed the latter for McDonald as he approaches his mix position at each show carrying nothing more than his Allen & Heath iLive-112 work surface. It’s connected to the house rig via a Cat 5 connector, and the mix is on.

A self-declared “Midas man” for many years, and also richly steeped in the ways of the PM5D, McDonald first put his hands on an iLive-112 a couple years back while wrangling house sound for the French electronic music duo Air and the U.K.’s Florence and the Machine.

Adele enveloping the audience on the current 21 tour.

“I walked away thinking to myself that those were the best effects I’d ever heard on a board,” McDonald says, recalling his first experiences with the iLive-112. “The work surface itself is about half the size of a PM5D, and a quarter of the weight probably.

“In these days when everyone owns their own board, yours is going to be the smallest, and you can easily just flip it out of its own case, connect to the snake, and get to work. You can even put it on a keyboard stand. When you’re done, you just fold it up, carry it away under your arm, and everyone loves you.”

A 64-input by 32-output system, the digital iLive-112 is outfitted with 28 faders in three banks with four layers offering 112 control strips.


Carrying no outboard effects at all, McDonald turns to his board for all processing, calling upon the power of a variety of plug-ins - or audio modules as Allen & Heath prefers to call them.

For Adele’s vocals, plug-ins replicating the sound of vintage ENT plates are his choice, along with delays and a host of reverbs.


Clever Things
“There are a lot of things going on onstage, so I like to create a number of different textures to properly express them,” McDonald explains.

“There are great de-essers on the board I use on the backing vocalists. There are also a number of clever little features you can’t find on anything else.

“Among them is a real-time analyzer that lets me physically see, for example, precisely how the de-essers are cutting into the vocals, or exactly how the gates on the kick drum are behaving.”

The Allen & Heath digital mixing system’s iDR-64 MixRack for monitors, working in tandem with the iLive-112 mix surface. (click to enlarge)

An extensive range of microphones from Sennheiser form the backbone of the show’s input list.

The scheme is, according to McDonald, “A personal choice really, we use all Sennheiser kit onstage.”

For Adele, McDonald reached into the Sennheiser catalog and chose a wireless SKM 2000 system with an SKM 500-965 G3 transmitter.

“Her voice has a really nice midrange,” he says, “it’s smoky - smoky rich and R&B straight from the ‘60s.

“Then the high-end is super-cutting. It’s like a much older voice than her actual years, and this mic is perfect for it. It pushes, really projects. I love Neumann 105s, but they are a bit too sparkly on top for this task.”

Sennheiser wireless systems racked up on the 21 tour, and inset, some of their companion transmitters. (click to enlarge)

Backing vocals are managed with hard-wired Sennheiser e 935s, while the drum kit starts on the top at overheads with Sennheiser e 906s, moves down to e 903s on snare, e 904s on toms, and an e 901 and e 902 jointly residing in the kick.

Acoustic guitars rely upon Avalon DIs, and for what McDonald calls a “gag piano” (a lacquered upright that looks like a traditional upright but actually houses a pair of Yamaha electric Motif keyboards), Radial J48 DIs provide the feed.

Cruise Control
At monitors, Joe Campbell presides over the stage with the help of his own Allen & Heath iLive-112.

Both Campbell and McDonald’s iLive-112s share a single controlling brain that lives in an onstage rack.

From his perch out in the house, McDonald manages gain structure for both desks, effectively “cruise controlling” Campbell’s board in this regard.

Sennheiser G3 IEMs are used widely onstage by everyone but one of two guitarists and the bass player.

Wedges from L-Acoustics provide the latter with more-of-me, and also stand in a grouping of four in front of Adele, who will occasionally pull her ears out and listen to just the monitors with all four running to obtain a different aural perspective of what’s going on.

Joe Campbell at the Allen & Heath iLive-112 he uses for monitors. (click to enlarge)

Campbell’s mixes are known for their precision and extensive scene changes built to accommodate a wide range of instrumentation and sounds emanating from the show’s more traditional instrumentation as well as a banjo, stomp box, and melodica.

“Unlike myself, Joe has a lot of work cut out for himself in terms of scene changes,” McDonald notes. “I, on the other hand, don’t believe in building and storing scenes. I like to use my own brain and hands. At front of house, we’re paid to interpret an artist’s songs, and we’re chosen for our style of interpretation.

“That’s where the art is in my job. There are many musical moments in this show that will give you goose bumps. My goal is to project these qualities in a forceful manner that isn’t aggressive. I never want the audience to feel as if it’s being shouted at. The music should envelop everyone, but not to the point they feel uncomfortable.

“Adele’s vocals are amazing,” he concludes. “I want the audience to forget who they are for a moment and be able to project themselves solely onto what’s occurring onstage. That is, after all, why we go to shows.”

Gregory A. DeTogne is a free-lance writer and publicist who has served the pro audio industry for the past 30 years.

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Posted by Keith Clark on 06/22 at 10:40 AM
Live SoundProductionFeatureProductionAudioConcertConsolesDigitalInterconnectMicrophoneMixerSound ReinforcementPermalink

Powersoft Upgrades Armonía With I/O Signal Integrity Monitor Feature

The upgrade to Armonía pro audio suite software puts more power in the hand of the audio engineer.

Powersoft has announced the available upgrade to version 2.1.803 of its Armonía Pro Audio Suite software, which includes the new I/O Signal Integrity Monitor feature.

Armonía was designed to deliver comprehensive control of all aspects of the amplifier/speaker audio chain to the audio engineer.

The I/O Signal Integrity Monitor furthers this control by providing information on input signal irregularities.

“With the release of Armonía version 2.1.803, the best audio control software in the industry just got better with the addition of the I/O Signal Integrity Monitor element,” says Ken Blecher, executive vice president of Powersoft Audio Technologies Corp.

“With Armonía, the engineer can make informed choices when setting up any touring or fixed installation employing suitably equipped Powersoft K Series and Duecanali amplifiers. The I/O Signal Integrity Monitor puts one more critical tool in the hands of the engineer.”

The I/O Signal Integrity Monitor notifies the engineer of any system irregularities due to a missing input signal or loudspeaker failure. The software enhancement was designed to work effortlessly without complicated initial measurements, yielding genuinely accurate information thanks to a programmable pilot tone and trigger threshold.

The I/O Signal Integrity Monitor will raise the alarm even if a single loudspeaker within a complex system dies, no matter if it is one of multiple lo-Z drivers wired in parallel or one of many in a hi-Z 70/100V line.

When dealing with very large and complex audio systems, the I/O Signal Integrity Monitor streamlines system setup before an event and helps relieve troubleshooting panic during a show by instantly providing the necessary information to locate and fix any problems quickly.

The I/O Signal Integrity Monitor joins a host of further enhanced features that make up the Armonía Pro Audio Suite. Powersoft’s TruePower limiter keeps the amplifier’s output power at safe levels for the speaker, depending on the varying load impedance over frequency. A Cross Power Limiter mode prevents unnatural pumping effects in a bi-amped system.

Active DampingControl provides compensation for speaker wire current losses yielding dramatically improved cone control with a virtually negative output impedance. These features join Powersoft’s patented LiveImpedance feature that displays real time load impedance at real-world power levels using the actual musical signal as the basis for the display information.

Crossover and output EQ’s can be created from FIR and IIR filters and the input EQ section allows for surgical system tuning. The smart and flexible Preset Management window makes system setup straightforward while offering system lockout options.

Armonía offers instant alerts on-screen, per e-mail and/or SMS and provides for the logging of alerts and events. The advanced grouping panel makes for easy management of complex systems, while offering sophisticated input EQ for individual channels and groups with multiple layers, various filter types and raised cosine implementation.

Armonía works with standard network protocols (Ethernet or RS485), offering easy network setup with daisy-chaining and auto-addressing with Ethernet and works with two simultaneous AES3 streams, equivalent to four audio channels over Cat-5e wiring.

The system offers redundancy per ring architecture, a dedicated on-line support forum and programmable stand-by modes for lower power consumption.

Powersoft

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Posted by admin on 06/22 at 07:15 AM
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Monday, June 20, 2011

Behringer Introduces Eurocom MA6000 Series Automixers

Designed for permanent installation in a wide range of venues requiring automixing

Behringer Eurocom MA6000 Series automixers merge quality audio with high efficiency for power savings over traditional technologies, due to switch-mode power supplies and Class-D amplifiers.

The 2 x 80-watt MA6008 and 2 x 180-watt MA6018 integrated mixer-amps are capable of driving 70/100-volt lines, and feature 8 mic/line inputs on XLR/TRS and Euroblock connectors, plus 4 pair summed RCA Aux inputs and a dedicated paging input.

The series also includes a stand-alone MA6000M mixer and the 2 x 240-Watt MA6480A power amplifier with dual switchable mic/line inputs and phantom power.

Dual bridging I/O connectors allow multiple units to be interconnected, making the MA6000 Series eminently scalable.

Equipped with remote volume control, source selection and system power, MA6000 Series automixers are designed for permanent installation in conference centers, council chambers, courtrooms, hotel meeting rooms, or any other venue where automixing is the logical solution.

The Eurocom MA6000 Series will be available in quarter 3 of 2011.

Behringer Website

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Posted by admin on 06/20 at 02:00 PM
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Sunday, June 19, 2011

Soundcraft Digital Console Range Expands Further With New Si Compact 32

Provides 40 inputs to mix, while maintaining the DSP functionality of its smaller brothers

The Soundcraft Si Compact range expands with the release of the Si Compact 32, capable of delivering 40 inputs to mix, while maintaining the DSP functionality of its smaller brothers.

The Si Compact features 14 main buses (all with dynamics, delays and BSS graphic EQs), four matrix buses, four dedicated FX buses, and four full-time Lexicon effects engines.

It also offers a range of option cards to interface with other systems, such as Aviom, CobraNet, AES/EBU and MADI.

All Si Compact models can connect via MADI to the new Compact Stagebox, adding remote connectivity to these great value digital mixers.

“We believe that this very cost-effective, powerful mixer will be a natural choice for larger install or medium tour sound and house of worship systems,” says Keith Watson, marketing director, Soundcraft Studer.

image

Soundcraft Website

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Posted by Keith Clark on 06/19 at 01:45 PM
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JBL Introduces CSM-32 And CSM-21 Commercial Series Mixers

Do not require a computer for configuration and are designed with simple analog-style controls

JBL has debuted the CSM-32 and CSM-21 Commercial Series mixers that provide an entry-level, cost-effective approach to commercial audio applications such as retail, hospitality and conferencing.

The new mixers can be configured for a range of uses such as paging, background music, and security applications, do not require a computer for configuration and are designed with simple analog-style controls. An included security plate can be placed over the controls to avoid uninvited changes to a commissioned system.

The CSM-32 features three stereo inputs and two stereo zone outputs and the CSM-21 features two stereo inputs and one stereo zone output. Both processors are stereo throughout the entire digital processing path and can be run in stereo or mono mode.

The CSM-32 offers an isolated output, specifically designed for supplying music-on-hold audio to phone systems.

The new JBL Commercial Series Mixers offer a diverse palette of processing tools, including priority override, page ducking, source and zone EQ, LevelGuard, AutoWarmth and a built-in crossover. 

LevelGuard helps maintain optimum levels through the system by automatically applying the correct amount of compression to the source signals based upon the incoming level.

AutoWarmth helps maintain musical warmth at all operating levels by automatically adjusting the tonal balance within a zone based upon the zone output level.

“Combining simple analog-style control with digital signal processing provides commercial integrators and their customers in retail, hospitality and other verticals the best of both worlds,” stated Iain Gregory, market manager for installed sound at Harman Professional Signal Processing Group. “Making sophisticated digital signal processing accessible to more users through a simple control interface means that the standard of business music systems can be redefined, which in turn enhances the overall customer experience.”

Three wall controllers are available for the CSM-32 and CSM-21 devices, which offer volume control and source selection from convenient remote locations.

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JBL Professional Website

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Posted by Keith Clark on 06/19 at 10:03 AM
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Friday, June 10, 2011

Inside The Sonic Passage: Creating A Custom Soundscape For San Antonio River Walk

An extension of the famed River Walk, where sound plays a key role in capturing the essential essence of the region

San Antonio is a city that prides itself on appreciating the local arts; the city’s museums are havens for visitors seeking to connect with the spirit of the Southwestern adventure.

When the time came to develop an area that encompasses bridges across the San Antonio River as part of a 1.5-mile extension of the River Walk, the San Antonio River Foundation realized that sound would play a key role in capturing the essential essence of the region.

Eight artists were invited by the Foundation to work on the River Walk extension – dubbed the Museum Reach – that runs from the Municipal Auditorium past the San Antonio Museum of Art (SAMA) and to the Pearl Brewery.

A stand-out sonic sculpture located along the River Walk beneath Jones Avenue Bridge near SAMA is Sonic Passage, which features the innovative work of Bill Fontana. A San Francisco-based ambient sound artist, Fontana elected to use a multi-channel sound system to project a number of unique sound sculptures that feature recordings from the local area.

“I decided to create an audio experience within three separate zones along the River Walk,” the seasoned artist explained, “based on the myriad sound of natural habitats and environments.” The soundscape combines pre-recorded sounds of local wildlife from natural habitats with live sounds captured from the river itself.

“My intention was to create a sound montage of the river for people sauntering along the River Walk,” Fontana continued. “I made a number of recordings of riverboats and local soundscapes, including river eddies and water sounds, in addition to bird sounds recorded in the Gulf of Mexico at the river’s estuary, which is very isolated in the spring and hosts a fabulous collection of birdlife. I made recordings as the estuary came alive from early morning throughout the day, sections of which I used as an evolving sound montage for the River Walk project.”

Fontana’s stand-out commissions include Spiraling Echoes, San Francisco, Speeds of Time, London, and Objective Sound, Panoramic Echoes, New York.

Click to enlarge

“This will be Fontana’s first piece in Texas and one of only a handful of permanent pieces that he has created,” said Mike Addkison, San Antonio River Foundation project director. “He’s using sequenced loudspeakers under Jones Avenue Bridge to produce a blend of continually changing, pre-recorded and live elements sampled from the headwaters of the river down to the Gulf of Mexico.

“People passing by will hear running water, birds, the turning wheel of an old mill and hundreds of other recordings of the river. In essence, what Fontana is doing is ‘washing’ people in the sounds of the San Antonio River.”

Working closely with Addkison and Big House Sound, a local audio contractor that fabricated and installed the audio system, Fontana developed a playback configuration that includes a hard-disc system to hold the pre-recorded sound elements, linked to a bank of Lab.gruppen CX20:8X modular amplifiers powering an array of Meyer Sound MM4 miniature wide-range loudspeakers mounted along the River Walk.

“The result is a natural sounding experience in a surreal atmosphere,” said Zack Edwards of Big House Sound. “The loudspeakers mounted around visitors are divided into three distinct zones: The Canal Area, The Primary Walking Area and a Secondary Walking Area. The sound is fully synchronized to provide an enveloping sequence of sound within each of these zones.”

“I personally selected the Meyer Sound loudspeakers,” Fontana added, “because of previous experience while designing my soundscape for the Champs-Élysées in Paris.” The original MM-4 miniature loudspeaker was designed specifically for an earlier project in Lyon, France. “In several of my works, I’ve come to rely on the accuracy of Meyer Sound loudspeakers. My medium is sound; I could create the most interesting piece, but if it doesn’t translate to the space, it’s worthless.”

The installer chose a pair of CX Series 20:8X 8-channel amplifiers, which deliver 250 W into 4-ohm loads from a 2U chassis.

“The C Series is absolutely bulletproof,” Edwards said. “And the amps runs happily in high ambient temperatures without air conditioning, particularly during summer months. We needed a design that can handle wide temperature swings – that was a key selection criterion. Plus the fact that the CX Series’ built-in thermal-protection mode protects the systems against major failures.”

The system has been in continual use since it was installed in the spring of 2009, with no loss of operation, the installer reports. “We had a warm summer here in San Antonio, with ambient temperatures of 100 degrees Fahrenheit; because of the forced cooling – and shielding from direct sunlight – the inside of the enclosure was maintained consistently at around 90 degrees Fahrenheit.”

Click to enlarge

Lab.gruppen’s CX Series amplifiers employ a proprietary Class D output design, with a selectable 35 Hz high-pass filter and built-in GPIO control facilities. “The C 20:8X is capable of adapting to a wide variety of demanding load conditions,” stated Edwards.

Each channel features an individually configurable Voltage Peak Limiter (VPL) that enables the output to be optimized for any loudspeaker load. The system is fully compatible with the firm’s NomadLink network, which allows key amplifier parameters to be displayed via DeviceControl software, with remote control of channel mutes and power on/off under network control.

“It was clear from our initial discussions with Mr. Fontana that he would not compromise on the audio equipment’s quality and performance,” the installer continued. “After that phone call we knew we would be using Lab.gruppen power to drive the MM-4s. We have also started using Lab.gruppen amplifiers for our concert systems, because they sound great and are reliable; it was this kind of performance that we needed for the River Walk project, so the CX20:8X made perfect sense.”

Cables to the 16 Meyer MM-4 loudspeakers – four mounted along the canal, eight beneath the bridge and four along an additional walkway – are carried via steel conduit. In contrast to conventional low-power 70 V, transformer based systems, the MM-4 connects directly to an amplifier and is capable of producing high SPL with reduced distortion.

The loudspeaker houses a single four-inch cone driver with a 16-ohm voicecoil mounted in a sealed enclosure; it draws 150 W peak, and produces a reported 112.5 dB peak SPL. The enclosure’s black anodized, extruded aluminum serves as a heat sink to cool the driver’s voicecoil. “We specified Belden all-weather outdoor-rated cabling,” Edwards said.

The outdoor installation posed its own unique design challenges. “A major consideration was to protect the systems from weather, theft and vandalism. We decided to house all of the playback equipment and amplifier racks in a central, custom-deigned enclosure and run individual signals to each of the suspended loudspeakers,” the installer explained.

“The overall design had to cover three primary aspects: One, the enclosure needed to provide a safe haven for the equipment, safe from public access; two, it needed to be protected from the elements; and three, it needed to be temperature controlled.” Fabricated by DDB Unlimited, the rain-proofed enclosure features 19-inch rails for the rack-mounted equipment, plus a venting system that forces air through the hood.

“We suspended the enclosure some 12 feet in the air,” Edwards recalled, “using custom strapping and brackets.”

Click to enlarge

The equipment enclosure is mounted beneath the bridge out of the worst of the weather, with conduit routed away from reach of the public. “We used thick-gauge steel for the enclosure, with perforated covers that were custom designed in Austin,” the installer continued. “The Meyer MM-4 loudspeakers are a weatherized version with a water-seal kit and sealed EN3 connector for cable termination.

“Each loudspeaker is mounted within a 10- x 10- x 8-inch steel enclosure with a moveable U-bracket yoke and swivel bracket to allow them to be aimed as necessary. Within the river zone the loudspeakers are mounted between 12 and 16 feet above water level, while in the sidewalk areas they are between 8 and 10 feet in the air. It is a very robust and well-engineered system.”

The system’s front end comprises a Richmond Sound Design AudioBox AB64 system, capable of providing level adjustment and EQ of 16 discrete channels of audio routed to eight outputs; a fully loaded unit accommodates 64 I/Os. An industrial-grade Windows-based PC handles cross-point programming and replay of pre-recorded audio tracks stored on an internal hard drive.

“In addition to the 16 discrete loudspeaker channels,” Edwards said, “we provided eight input channels that handle live sources, which includes a pair of DPA 8011 hydrophones placed in the nearby canal.” The hydrophones pick up the sounds of passing boats and propellers and blend these ambient water-borne sounds with the pre-recorded audio tracks. Microphone pre-amps and phantom power for these live sound sources is handled by a rack mount Whirlwind Mix 5S 4-channel stereo console.

The playback system is fully sequenced by the artist to provide a continuously changing soundscape along the walkways. According to Edwards, the system turns on at 7 am and off at 11 pm.

“I like the built-in contact-closure input on the CX Series, which allows the amplifier to be tied directly into an AC power sequencer for powering the amp on and off; this feature can save a lot of money and makes the installation a lot cleaner.

Click to enlarge

“For the River Walk project, we used a Lowell four-step sequencer that was triggered by the city’s network clock so that all the gear is safely powered on and off daily.”

The time-controlled unit initiates pre-determined start-up and shutdown sequences.

To allow Fontana to remotely access the AB64 while programming and adjusting EQ for the three sound zones from his laptop, Big House Sound installed a wireless network. “This wireless connection enabled the AB64 and associated equipment to be controlled without opening the rack,” Edwards explained. “Bill Fontana was really excited about this setup since it allowed him to roam around while making adjustments to the replay sequence and zone levels.”

“The result is a major success,” concedes San Antonio River Foundation’s Addkison. “Our intention was to take sound and bring it into the forefront of the river experience – not just as ambience. Bill Fontana’s remarkable, ever-changing soundscape has achieved all of that, and more – we have enjoyed a very strong and positive response from audiences. The sound playback system has proved to be ultra-reliable and just what we need for this important exhibit in our fair city.”

Mel Lambert has been intimately involved with production industries on both sides of the Atlantic for more years than he cares to remember. He is principal of Media & Marketing, a Los Angeles-based consulting service for the professional audio industry.

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Posted by Keith Clark on 06/10 at 09:42 AM
AVFeatureAmplifierAVDigitalInstallationLoudspeakerMixerSound ReinforcementSystemPermalink

Rane Debuting HAL Exapndable DSP Platform At 2011 InfoComm

Seamlessly interface HAL to an application with a broad variety of peripheral devices

At the 2011 InfoComm show in Orlando, Rane (booth 615) is introducing HAL, an expert in room combining, paging and distributed audio systems.

HAL easily guides even novice users through what used to be complex tasks in just minutes.

No intricate matrix mixing or presets are required for room combining and paging, and no virtual wiring required to distribute pages and background music to multiple zones.

Seamlessly interface HAL to an application with a broad variety of peripheral devices including smart Digital Remotes, Remote Audio Devices (RADs), portable and in-rack auto mixers, bus Expansion devices, and an advanced Paging Station.

In addition, HAL and Halogen software check the status, location, CAT 5 wiring integrity, and that audio is flowing in all peripheral devices, so you know your system is properly connected and ready to go.

HAL is fully compatible and easily integrated with existing Rane products like gain-sharing auto mixers and Rane’s wide variety of RADs.

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Rane Website

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Posted by Keith Clark on 06/10 at 09:01 AM
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Wednesday, June 08, 2011

How To Use Audio Sandpaper: The Reverb Basics In Worship Audio

Reverb can be used to give a song a specific feel
This article is provided by Behind The Mixer.

 
Reverb is a beautiful effect that gives us the ability to bring a fuller sound to an instrument. 

Reverb can be used to spatially place an instrument in a mix such as sitting in the background. 

We can also kill our mix with reverb if we use too much or we use the wrong type of reverb. 

Here are three standard types of reverb and how they can be used…

First, let me just get this out of the way. When a singer talks between songs such as saying a prayer or reading a section of scripture, PLEASE TURN OFF THE REVERB! It’s not the place for reverb and it is distracting to the congregation.

What is a good definition of reverb? Reverb is an effect which modifies an existing sound so it appears to take on the audio characteristics of a unique space. The reverb in a bathroom is not the same as the reverb in a grand hall, but both sounds produce distinct reverberations.

Recording studios can use reverb to make a song sound like it was recorded in an intimate space or a huge concert venue. 

The way we use it in the live environment is similar but a little different.  For example, I could place a lot of reverb on a singer’s voice so as they sound like they are singing in a opera hall. 

However, if I were to do so in a very small church then the congregation will hear the sound and find it doesn’t fit.  In short, their ears tell them it’s a lie. That is to say, what can be acceptable in a recording isn’t always acceptable in the live environment.

In the live environment, we can use reverb to add spatial distant in a mix. 

For example, you can make backup singers “sit back” in the mix by giving them some reverb.

The more reverb, the more distance.

We can use reverb for instrumental separation. Straight on instruments with no reverb have an all-out up front sound. Adding a little reverb to a guitar or a singer and it’s like taking sandpaper to a freshly built chair - it makes it smooth. 

Reverb can be used to give a song a specific feel. The more reverb, the more airy and light. Not always, but when speaking about audio, there are very few absolutes.

There are three common types of reverb; room, hall, and plate.

Room: Everything about room reverb is small. It has the characteristics of a small room. It adds a little depth and a little space. Also, it’s a short time period of reverb.

Hall: “WELCOME TO THE VELODROME-OME-OME-OME-OME!” Hall reverb lasts a longer period of time and carries more reflection. It carries a larger fuller sound. The smallest of halls is still bigger than the largest of rooms. 

Plate: Plate reverb is the sound a plate makes when you accidentally drop it in the kitchen and the sound echoes throughout the house…OK, maybe not.

Plate reverb does not emulate any specific space. It is created through sound vibrating a metal plate at the end of a tube. This metal plate vibrates rapidly, and this “reverbed” sound carries a lot of early reflection. You might even say it has a heavy feeling. 

Plate reverb is popular with drums. A benefit of plate reverb is it gives the thicker sound you might associate with a hall reverb but for a shorter period of time.

The next time you start your mixing, remember that reverb carries a lot of power that’s just waiting to be unleashed.

Oops, I almost forgot…when mixing, EQ your instruments before you add any reverb. Going back to the sandpaper analogy, you always use sandpaper last.

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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Posted by Keith Clark on 06/08 at 04:33 PM
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Radial Engineering Releases Workhorse WR8 500 Series Rack

Radial Workhorse WR8 is an 8-slot 500 series mixer with an optional WM8 8 channel summing mixer.

Radial Engineering Ltd. is pleased to announce the availability of the stand-alone Workhorse WR8 500 series rack and optional WM8 8 channel summing mixer.

The Radial Workhorse WR8 is an 8-slot 500 series mixer that is 100% compatible with all standard 500 series modules. Individual XLR inputs and outputs are complimented with parallel 1/4” TRS connectors and Pro-Tools compatible 25 pin D-Subs.

This enables any of the 8 module slots to be easily integrated within the framework of today’s popular digital workstation environment. A special feed function also enables modules to be connected in series without having to hard patch using a cable. Should the user wish to add the optional Workhorse 8 channel mixer section, a step-by-step mixer installation slide show can be found on our website by clicking on the WR8 rack photo.

According to Radial sales manager Steve McKay Radial began shipping the Workhorse into the North American market and then expand into Europe.

“After shipping hundrers, we are pleased to say the that Workhorse launch has been tremendously successful,” adds McKay. “The time has come to make the Workhorse available in a rack-only format for those who may not require the built-in 8 channel mixer. This not only brings the cost down of the Workhorse rack, but also enables the user to add the mixer at a later date, should the need arise.”

Both the Workhorse WR-8 rack and the WM-8 mixer retail for $799 USD and are now available through authorized vendors.

Radial Engineering

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Posted by admin on 06/08 at 12:30 PM
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Maximizing Your Church Sound Mixing Console With A Logical Approach

Figuring out a simple road map that illustrates how the signal gets from the input to various outputs

Let’s take a poll: how many of you use a road map to find the most efficient route to get where you’re going?

Wow, a couple of folks actually raised a hand! It may seem odd, but one of the best ways to understand how to operate a mixing console is to learn its signal flow.

Figure out a simple road map that illustrates how the signal gets from the input to various outputs.

Without this understanding, then the likely approach is your one of, “Well, if I twiddle these three knobs just so, push this fader up to here, and never ever let the master fader go past this indelibly engraved red mark that someone has carved into the face of the console, then maybe it’ll work today!”

There really is a better way, and it’s the concept of signal flow logic. Let’s get down to work.

Figure 1, below shows the input strip from a hypothetical but typical console. We’ll talk about each of those controls as we make our way through the signal flow.

On the surface, one might think that the sound flows through each control in sequence from top to bottom. Actually, manufacturers have arranged the controls in a way that they think will be the most efficient way to operate the console.

Figure 1: Hypothetical of a typical input strip (click to enlarge)

For example, controls that you would for the most part “set and forget” are placed at the top of the input strip, well out of your reach, such as, say, the microphone trim.

Controls for immediate and frequent access are placed right at the user’s fingertips, like the channel faders.

As we move on to the signal flow diagram later, it’s a good idea refer back to Figure 1 from time to time. It helps tie the signal flow into the related controls.

Applied To All
The signal flow diagram (Figure 2, below) is again from a hypothetical console. This arrangement of the controls is fairly common, and once one understands this concept, it can be applied to all consoles.

The diagram reads from left to right, top to bottom. So we start at the upper left corner at the mic input - a very good thing to have on a console! The first component that the signal sees is the input transformer.

Now, the reality is that on most consoles we work with today, especially on lower cost consoles, the input transformer is actually replaced by a less expensive electronic circuit that accomplishes much the same task.

Figure 2: Signal flow diagram (click to enlarge)

The purpose of this input stage on each channel is to receive the balanced signal from the microphone and to cancel any extraneous noises.

The strength of that mic signal is quite low, and needs to be brought up significantly to operate with a quiet signal-to-noise ratio throughout the rest of the console and beyond. That gain increase is accomplished with the mic preamp.

Note that there is a related control called the mic trim (or mic gain). This is a gain adjustment for the mic preamp that allows you to adjust for differing signal strengths coming into each channel.

If there isn’t a control labeled mic gain, it simply means that the manufacturer has chosen to keep the price of the console down by replacing that variable resistor (called a potentiometer) with a fixed resistor.

The manufacturer figures that you’re going to plug in mic “A” and mic “B”, and you’ll place them on these various instruments or voices; then a ballpark gain setting is chosen that should serve the needs of all of the inputs. (That fixed resistor costs about one-tenth as much as the pot - multiply that times one for each channel.)

However, the trade-off for lower cost is flexibility. For example, you might decide that you get the best sound if you mic a certain instrument in a certain way. If the incoming signal is too hot or too soft, you could use that sensitivity control to adjust for the difference.

Without that control, creativity is required, i.e., what’s another way to drop the strength of the signal coming into the console on a particular channel? One could move the mic farther from the instrument.

The main path of the signal goes next to the equalizer section of the channel. This is really just a glorified volume control which allows you to turn the sound up or down in various frequency bands. Your intent should be to improve the tonal quality of a particular voice or instrument.

Remember to be subtle with EQ adjustments. A small EQ change can make a huge improvement in the quality of a person’s voice. But there are times when altering EQ is simply not needed. Part of a mixer’s job is to know the difference, and then to not twiddle the EQ knobs just because they’re there.

Hip Terminology
The signal then flows on to the channel fader. Fader is the hip term - I want all of you to be hip, so stop using the term “slider”.

Here’s where most of the action of the mix happens. The fader provides the means to make adjustments in the balance of each voice or instrument during the song or sketch.

Faders generally have a logarithmic taper; that is to say, the rate that the volume changes will vary along the throw of the fader.

For example, a 1/4 of an inch move of the fader near the top of its travel may account for a 5 dB change in level, whereas the same 1/4 of an inch move of the fader near the bottom of its travel may account for a 20 dB change in level.

As a result, the mix will be smoother and quieter if the faders are operated in the upper one-third of their travel for the most part. This deals with gain structure; hang on, we’ll get to that point soon.

Now our signal path is at the summing point. The Greek Sigma symbol with a circle around it is a common notation for a “summing” or “combining” point.

The signal flow diagram here shows just one channel of the console, but let’s say that we’ve really been talking about a 16-channel console. Each channel is identical up to this point, and then they all combine or sum

Signal then goes to the master fader where the overall level of all the signals is controlled with one fader. Signal then goes through the output stage of the console circuitry, and then to a connector/cable on the back of the console, where it’s fed to the input of the power amplifier(s), for example.

That’s the main path through any console. No matter how expensive, that’s it. But we still need some more information on how the controls interrelate, so look back at the signal flow diagram and note that right after the mic preamp, but before the EQ, there is a pickoff point. Think of it as a side road on your map.

The path drops down to a control I’ve labeled auxiliary 1. Follow the signal path down, and you’ll see that it comes in contact with the aux 1 bus. This is basically a piece of wire that runs from one side of the console to the other.

Signal then flows on to the right to the aux 1 master control, which controls the overall level of all the signals fed to the aux 1 bus.

The Fade Route
Let’s imagine for a moment that we’re using aux 1 to feed a set of stage monitors. We’re looking at channel 5 that is the lead vocal mic, and aux 1 feeds the lead vocalist’s stage monitors.

If you were to make an adjustment to the EQ on channel 5, would that change be heard in the vocalist’s monitor mix? No, because the feed for the monitor mix is picked off before the EQ.

If making adjustment to the lead vocalist’s channel fader, would that change be heard in the vocal monitors? No. We’re using what is known as a “pre-fade” auxiliary send, because the signal is picked off before the fader. It also happens that the signal is picked off before the EQ as well, but the proper term is “pre-fade”.

But what if we make a change to the mic trim on channel 5? Yes, because it’s upstream in the main path, changing the setting of the mic trim would affect the signal to both the main path and the auxiliary send.

This relationship is usually the preferred setup for mixing monitors. Most vocalists and musicians prefer that once their monitor mix has been established during soundcheck and rehearsal, that their mix not change during the course of the worship set (or show) unless they specifically request it. Unexpected changes in the mix can be very distracting.

Just past the fader is another pickoff point that goes to a control labeled auxiliary 2. As before, this also drops down to an aux #2 bus and its related master control. In this case, let’s imagine that the aux 2 output on the back of the console is patched to the input of a reverb unit sitting in an equipment rack next to the console.

In order to hear the reverb, the output of the reverb device is connected to a line level input on the console called the auxiliary return. We establish our dry-to-wet ratio by adjusting the send on aux 2 of channel 5, and of course the return level of the reverb unit is adjusted with the aux return.

The “dry” sound is the direct signal from, in this case, the vocal mic, and the “wet” sound is the reverb effect.

Draw & Redraw
Now that the reverb sounds the way we like… If an adjustment is made to the lead vocal fader, does this affect the signal fed to the reverb device? Of course it does. What if you make a change to the EQ on channel 5? Yes, that change will also be reflected in the reverb sound.

Does adjusting the master fader affect the feed to the reverb device? No, because the feed to aux 2 is picked off before the master fader. What if the aux 1 send on channel 5 is changed - will that affect the signal fed to the reverb unit?

No. Even though aux 1 is picked off upstream from the aux 2 feed, it’s still just a pickoff point and will not affect the main signal path. Realize that the aux 2 send described here is commonly called a “post-fade” send.

It’s typical to use a post-fade send to feed effects devices because doing so allows us to easily maintain that dry-to-wet ratio. It is generally a more musical sound for the loudness of the reverb to track with the direct sound.

If you push the fader up, the direct sound increases, as does the feed to the reverb unit, so the dry-to-wet ratio is maintained. If you pull the fader down, the reverb sound diminishes as well.

The owner’s manual for your console will include a block diagram. These drawings are often awkward to read, so redraw it! Not only will it be easier to read and simpler to understand, but the point of signal flow is reinforced understanding of its flow.

All you have to do is get past the manufacturer’s silkscreen on the console. For example, what one manufacturer calls “solo” another labels the same control “cue”, and still another labels it “PFL”; what one manufacturer calls “monitor send” another labels “foldback”.

But it doesn’t matter if the console cost $600 or $60,000. The reality is that they both possess the same basic signal flow.

Curt Taipale heads up Church Soundcheck, a thriving community dedicated to helping technical worship personnel, and he also provides expert systems design and consulting services with Taipale Media Systems.

More articles by Curt Taipale on PSW:
Staying Focused - A Path To Excellence In Operating Your Church Sound System
Choosing The Right Console For Your Church Sound System
The Powerful Affect Of Digital Effects In Your System
Who Defines “Good” Sound At Your Church?
Install Your Own Church Sound System? Here Are Some Cautionary Tales
Humor Files: Unintended Amendments To The Laws Of Physics

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Posted by Keith Clark on 06/08 at 12:12 PM
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Audio-Technica AT-MX381 SmartMixer Joins Crestron Integrated Partner Program (IPP)

Contractors and system integrators can now control A-T 8-channel automatic mixer with Crestron products

The Audio-Technica AT-MX381 SmartMixer 8-channel automatic mixer with optional computer control (via RS232) now features full integration with Crestron Electronics control products, through a dedicated AT-MX381 module developed as part of Crestron’s Integrated Parter Program (IPP).

This compatibility allows contractors and system integrators to install the AT-MX381 to operate seamlessly with other hardware in an open-platform Crestron systems environment. Such a system may consist of audio/video, security, HVAC, lighting and other controlled devices that may be incorporated into commercial configurations.

Equipped with optional computer control, the AT-MX381 SmartMixer is the latest addition to Audio-Technica’s family of automatic mixers. This microprocessor-controlled automatic mixer is designed to provide automatic mixing functions for installed sound, house of worship, broadcast and conference applications.

Mark Donovan, Audio-Technica sales engineer – installed sound, states, “Audio-Technica is committed to supporting systems integrators by providing them with the tools necessary to help streamline their projects, saving valuable time in the process. With that in mind, we are proud to join forces with Crestron in order to simplify the process of integrating the AT-MX381 with Crestron’s control systems.”

“Crestron is happy to welcome Audio-Technica to our Integrated Partner Program,” says Dominick Accurso, DMC-D manager - integrated partner program. “We have also just released the AT-MX381 module on our site and it is now available for free download.”

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Audio-Technica Website
Crestron Website

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Posted by Keith Clark on 06/08 at 06:29 AM
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Behind The Glass: Producer/Engineer Joe Chiccarelli On Being A Sonic “Chameleon”

A conversation with Producer/Engineer Joe Chiccarelli, who first landed a job as an assistant engineer at Cherokee Studios and soon found himself engineering Frank Zappa on "Sheik Yerbouti" and "Joe's Garage Acts I, II & III" in 1979, and who has gone on to work with a tremendously diverse group of artists. An excerpt from Howard Massey’s "Behind The Glass Volume II," which features more than 40 all-new, exclusive in-depth interviews with many of the world’s top producers and engineers.

Joe Chiccarelli is a chameleon.

Not literally, of course. But unlike many producers whose sonic stamp is immediately recognizable (a Roy Thomas Baker or a John Shanks, for example), you’d be hard pressed to identify a Joe Chiccarelli “sound.”

It’s hard to believe that the same individual who produced the rough-and-ready White Stripes’ Icky Thump was also responsible for the ephemeral, moody ambience of the Shins’ Wincing the Night Away or the smooth, slick jazz tones of Kurt Elling’s Night Moves.

But not only was it the same guy, it was a body of work that netted him a 2008 Grammy nomination for Producer of the Year.

Chat with the soft-spoken, self-effacing Chiccarelli for just a few minutes and it becomes apparent why artists in so many different genres gravitate to him.

“Honestly, I don’t think I’m confident enough in my abilities to have a sound and a strong direction,” he admits disarmingly. “It’s more important to me to study the song and the artist and figure out what’s strong about them and then help the record be the best it can be.”

Originally from Boston, Chiccarelli relocated to Los Angeles in the late 1970s after playing in a series of rock bands. Always interested in the technical aspects of music-making, he landed a job as an assistant engineer at Cherokee studios, but his big break didn’t come until the day that Frank Zappa’s regular engineer was held up in London with visa difficulties.

As low man on the totem pole, the 20-year-old Chiccarelli was given the assignment to work with the notoriously difficult and demanding artist. Seven albums later, he had a career.

Since then, Chiccarelli has worked with an astonishingly diversified group of artists, including Tori Amos, Oingo Boingo, Black Watch, American Music Club, and My Morning Jacket. And every album he works on, it seems, sounds totally different from every other album he’s ever worked on.

“When people ask me, ‘What’s your approach to producing records?’” Chiccarelli says laughingly, “my answer is, ‘Well, what day is this?’ But on a creative level I think I would be dead if I just made the same record over and over again. The personal challenge for me is to try to make something that’s unique to that artist.” Clearly, he’s succeeding.

What do you think it was that Frank Zappa saw in you that made him want to continue to work with you?

I think it was because I was very much an open book. At the time, my only experience was in making good, clean contemporary pop records, while Frank’s whole thing was to try the most outrageous things possible in order to make the music interesting and dynamic and over the top.

It was a new place for me, but I was very willing to go there. Perhaps he just viewed me as someone who hadn’t done a lot of records and so wouldn’t be as set in his ways or closed to new ideas.

Frank was all about breaking rules and challenging the norm. I learned pretty quickly during my first few days with him that you just didn’t say no. [laughs] He really had a great sense of the big picture.

Before I even had a chance to make a statement or try to do things my way, I realized that this was a guy who could see five steps down the line, so I had to learn to trust him and know that in the end it would be okay.

A lot of producers and engineers I’ve talked to have stressed how important it is to be ready when your big break comes. Looking back with hindsight, what preparations had you made to be ready for that moment?

To be honest, I didn’t know where the Frank thing would lead. I was fortunate in that I fell in with an artist who was a workaholic and went from one album to the next.

But I didn’t know at the time that this was going to be a break; I thought it would be a very transitory thing, that I would work with Frank for however many weeks and then go back to Cherokee and resume my assisting gig.

In terms of preparation, I’m not so sure that I did anything specific, but the one thing I tell people who want to become an engineer or producer is, “learn everything.” Not just engineering and music, but also learn about art, poetry, literature, psychology. The job really involves a lot of things, and it changes from project to project.

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As someone who appreciates good sound, do you ever find it frustrating to work with an artist like Jack White, someone who’s into rough edges? Do you ever find yourself thinking, “if only we could work on this mix a little more we could get it sounding so much better?”

Yes, and there are many times where I will say something just like that: “Give me another half hour and let me fix this and fix that.” But the thing that makes rough mixes good is that you just kind of go for it, as opposed to laboring over it and making sure that every corner is polished and every little detail is in place.

That’s why they often find their way onto records, and that’s one of the things I respect about Jack: he’s so much about spontaneity and honesty - the reality of something - that he doesn’t want to spend a lot of time on sounds, on mixes, on anything.

Jack is a big fan of old-school recording; he’s the kind of guy who thinks that nothing’s sounded good since 1972. [laughs]

But if you go back and listen to a lot of the music from the ’60s and ’70s, the thing that it’s got more than anything else is a feel and an emotion. So I actually think Jack is correct in that things sometimes just get polished to death.

With the White Stripes, my basic role is to capture the performance and protect the energy and the magic that Jack and Meg have. And they’re a pretty powerful combination, I have to tell you.

I’ve recorded Jack now with three or four different drummers, but there’s a chemistry between him and Meg that’s unique. They’re so respectful with one another, and they work hard, and they push each other. Whatever people say about her abilities, it’s immaterial, because there’s something that she does that lets him do something very special.

Do you prefer to record digitally or to tape?

It really depends on the project. When I feel confident that the band has got it down in terms of performances and things will probably be just a matter of a few takes, then I’ll do it in analog.

With the White Stripes, we recorded to 16-track analog, which was Jack’s preference. But if it’s a situation where there’s still some uncertainty as to arrangements and structures, then I would choose the digital approach.

Having the Shins project done in Pro Tools was a godsend, because I was able to say to [singer/songwriter] James [Mercer] something like, “You know, it would be wonderful if the chorus happened again at the end,” or “Let’s put a whole new section in the middle with different textures, and let me show you real quickly how it could go.” Working digitally gave him lots of options.

For example, there’s a track on the album called Sea Legs where the chorus only happens twice in the song, and that was slated for release as a single. But for radio, sometimes that doesn’t really work.

So we tried doing the song with a more traditional pop structure, where there are three choruses and it ends on the chorus, and it worked, but we all felt that it was a little too normal-sounding. So we opted to go off on this crazy, quirky, almost Latin jam thing because it sounded really exciting when the song took a big left turn, and that’s the version we used for the album.

But when it came time to prepare the track for a single release, we went back and used the file that had a shortened jam section and a third chorus at the end.

Both analog and digital work fine, and they each have their strengths and weaknesses, and their own distinctive tonality. To me, it’s like having another microphone or compressor to choose from.

But even when I record digitally, my goal is still to get the sounds the way I want them on the way in. I’ve always taken that approach, and everyone I ever learned from back when I was just starting out took that approach.

In those days, you were limited track-wise, so my attitude was, every time you put up the faders to do a rough mix, that was your record, or at least it was 90 percent the way you wanted it. I viewed mixing as a process of balancing and refining, not reinventing, and that’s still my attitude.

What do you think it is that makes a song great?

In any kind of pop song, you want to be able to tune in and tune out at the same time. In other words, you want it to engulf you and captivate you every second of the way, but you also want it to take over your body in the sense that you don’t want to have to work too hard; you want to be able to turn off and just kind of sing along.

I think great songs work that way, in that you can view them from afar or be really inside them, just like a great painting or a great movie.

What do you think is the most important quality in a successful producer?

I think the more you are a fan of the music and are moved by it, the better the job you will do with it. And if you are really in love with the music, you will protect the artist’s integrity at all costs, and that’s all-important.

Of course, you do need to know a little of the technical side of making records as well as the musical side of it, but mostly you need to be well-rounded as a person. I’m always inspired by people that create works that are long-lasting, in any art form.

I think that what we do can sometimes be a very ephemeral thing, and I’m always awestruck by the Bob Ezrins and the George Martins in this business - people who have made records that will indeed last for a long, long time.

But I often try to gain my inspiration from art forms other than pop music - painting, or filmmaking, or novels, or great architecture: something that’s been around a hundred years, created by some guy who really broke all the rules.

If I go to a museum on a Sunday and I get motivated by some new young painter or sculptor, that’s more fuel for me to go into my medium and try to do the best that I can do.

Suggested Listening:
Frank Zappa: Joe’s Garage, Zappa, 1979
My Morning Jacket: Evil Urges, ATO, 2008
The White Stripes: Icky Thump, Warner Bros., 2007
The Shins: Wincing the Night Away, Sub Pop, 2007
Kurt Elling: Night Moves, Concord, 2007
American Music Club: San Francisco, Reprise, 1994

To acquire “Behind The Glass: Volume II” from Backbeat Books, click over to www.musicdispatch.com. NOTE: ProSoundWeb readers can enter promotional code NY9 when checking out to receive an additional 20% off the retail price plus free shipping (offer valid to U.S. residents, applies only to media mail shipping, additional charges may apply for expedited mailing services).

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Posted by Keith Clark on 06/08 at 06:16 AM
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Tuesday, May 31, 2011

Church Sound: Staying Aware Of What’s Happening With Personal Monitor Mixes

Just because the musician has control it doesn’t necessarily keep monitors from being a problem

With the advent of personal monitor mixing for musicians, the front of house guy is theoretically free from any monitor issues. 

However, that’s if:
1) The system is set up properly;
2) The musicians know how to use the stage system;
3) You don’t mess with the channel gain (If direct outs a channel are used).

Remember, though, that just because the musician has control it doesn’t necessarily keep monitors from being a problem.

The church where I mix has personal monitor mixers available on the stage.

Our musicians are top quality, and are there consistently (we don’t have much turnover). So 99 percent of the time monitors are never an issue. However, that remaining 1 percent can be frustrating!

I should note here that we do two distinct services every Sunday - they are very different from each other in terms of program and production elements.

This past Sunday during sound check for the first service, the worship leader stopped mid-song and said very emphatically, “Can’t you hear that!!!!????”  I responded, “Hear what?”

He then proceeded to play a note right around A 440. As he played it I could hear feedback wanting to take off.

On stage we had multiple choir microphones, a full rhythm section, a string player, a sax and a clarinet.

I immediately thought the problem must be the choir mics, so I dialed out the general area of 440 Hz on the EQ.

This seemed to help, sort of, but things still sounded “off,” and anything the musicians played in the key of F seemed to exacerbate the problem.

.

As I sat at the booth a little perplexed, the worship leader accidentally bumped into a microphone that we use for lead singers in our contemporary services.

This mic, sitting to one side of the stage, was folded down on it’s boom so that it sat parallel to the floor, and it was also pointed right at the back of one of the monitors on stage.

When the worship leader nudged this mic, he turned to the booth and asked why it was even on.

Bingo! It immediately occurred to me that someone on stage, probably trying to turn up the vocals in his/her monitor, was turning up that mic by mistake. 

I went to mute the channel of the offending microphone - ah, but the direct out on that channel is pre-mute.

So I figured I had a few choices:
1) Turn the gain all the way down (but then I would have to remember where it was and return it back to the right spot);
2) Disconnect the mic from the personal monitoring system (however, same problem as above, I would have to remember…);
3) Shut off the phantom power to this mic (it’s a Shure Beta 87 condenser).

I opted for choice number 3, and also put a large piece of tape on the button to remind me to re-engage it after the first service was done.

Problem solved, but a great reminder of the perils of using personal monitor mixers on stage with open wedges. You never know with musicians…

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.

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Posted by Keith Clark on 05/31 at 01:06 PM
Church SoundFeaturePollMixerMonitoringSound ReinforcementStage • (2) CommentsPermalink
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