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Tuesday, May 08, 2012

Project Energia: Inside Adamson’s New Multi-Phase System Project

The latest on Energia development, plus a conversation with Brock Adamson

Last September, Adamson Systems Engineering made public some of the details of Project Energia, which includes a new series of loudspeaker systems with networkable Class D amplifier modules, DSP, cable and power distribution, AVB network hardware with software integration of control, and 3-D simulation and diagnostics. The system will be under touchscreen control.

Adamson is releasing information about Energia in phases, each defined by close work with several leading sound companies who agreed to serve as beta partners.

Phases include: 1) Mechanical field testing; 2) Amplifier, power distribution and ground control field testing; and 3) Network and network hardware field testing.

Beta phase 1 actually began in July of last year, with Eighth Day Sound (U.S.), Wigwam Acoustics (UK), Fluge (Spain) and Big Daddy Productions (Indonesia) taking delivery of E15 line source loudspeakers, which were subsequently used on a variety of fall tours and large-format events around the world. Several other significant beta partners, including Sound Image, have since come aboard.

The Energia package has, at this count, four related patents pending. The E15 is built around the e-capsule, a surrounding module constructed in aircraft grades of lightweight aluminum. This skeletal structure provides an accurate and rigid frame for mounting the modular Autolock rigging system, while simultaneously housing a series of mid/high components on proprietary Co-Linear Drive Modules.

The e-capsule is flanked with two separate birch ply enclosures, each containing a proprietary Kevlar 15-inch woofer, capitalizing on the advantages of Adamson’s Advanced Cone Architecture.

Autolock is designed for a single technician to be able to set all angles on the ground, with no lifting involved. When connecting the flown section of the array to the next flyable section on the ground, the cabinets lock together automatically. Four E15 cabinets will ride in a dolly.

Briefly, the E15 is a 3-way system, with 2 x ND15 15-inch neodymium Kevlar cone drivers (2 x 8 ohms), 2 x YX7 7-inch Kevlar cone drivers (2 x 8 ohms), and 2 x 4-inch (1.5-inch-exit) Adamson NH4 compression drivers. Frequency response is 60 Hz to 18 kHz, horizontal dispersion is 90 degrees (-6 dB symmetrical), and vertical dispersion is 6 degrees (prolate-spheroidal sound chamber).

The cabinet, made of Baltic birch with textured water borne acrylic finish, measures 15.4 (h) x 51.4 (w) x 21.4 (d) inches and weighs 176 pounds.

Beta phase 2 is underway, focusing on the Class D amplification, DSP and ground control system that will provide diagnostics, control of individual bands in each E15, and more. Beta phase 3 will address the network management system, including a totally new software suite.

We recently talked with company president and CEO Brock Adamson to get further details about the concept, how it’s gone so far, and where it’s leading.

PSW: What are your observations, in general, of the current line array/loudspeaker market in sound reinforcement?

Brock Adamson: It seems that new product expectations have been lowered to the level of incremental transducer improvement. But, since the first line array entered the market, there have been enormous advances in three technology toolkits that should have a much greater affect on the evolution of the array element: engineering software, electronics and system software integration. Not enough of this is finding its way to the modern line array.

How did this drive the concepts of Project Energia?

We were motivated to put together the very best mechanical design tools found in solid modeling, finite element and boundary element analysis to expand the existing constraints of form and function of the array element. Then we looked to electronics for cost effective, lightweight power and communications. We are also developing system integration with the next generation of network and software tools such as AVB and Android.

What attributes differentiate Energia from your previous line arrays?

Well, if we reduce all that to a set of attributes, it would start with “ergonomics” and end with “total solution,” “size” and “efficiency” somewhere on the list.

Why are you rolling out Energia in stages?  When do you project that the initial product family, group, etc., will be completed and in full production?

Our strategy was established to ensure reliability at each phase of release. Energia represents a big step for our customers and for the company, particularly in the manufacturing stage. We are currently testing amplifier and power distribution hardware and the various aspects of AVB technology are just converging, with another IEEE document yet to be finalized.

How are beta partners selected?

Like most partnerships, the prerequisite is mutual understanding and common goals.

The beta test program seems to be quite thorough. What, specifically, have they (beta partners) brought to the table in terms of refining the system? Has any substantive re-design work taken place as a result of the beta partner’s input?

During Beta Phase 1 testing of the loudspeaker system, there have been a few mechanical changes, such as a revision of an aluminum extrusion profile and some packaging tweaks, but more important, is the evolution of filter presets.

Even at the outset of Beta Phase 2, consultation brought on significant changes to the power distribution system that will allow a better fit with companies ranging from small to larger shows.

Let’s focus on the E15: What is the overall scope of this loudspeaker?

To begin, we established a rigorous routine of modeling, finite and boundary element analysis, followed by rapid prototyping and acoustic measurements. This was applied to both transducers and sound chambers.

When combined with our new concept in the physical structure, we achieved the resulting improvements we were looking for.

Simply put, the system comes in a smaller lighter package with more headroom, less distortion and better coverage.

It’s faster to fly than anything on the market and it will offer advanced array processing and intelligent diagnostics.

They’ve been built around what you’re calling an “e-capsule” – can you describe that and offer further insight on the design?

The e-capsule is a rigid aluminum module that houses most of the technology. All rigging, electronics and mid/high waveguides are installed in this capsule. The woofer enclosures are then bolted on to each end. It offers a lightweight solution with the sonic benefits of wooden enclosures on the low and low mid bands. There are a number of patent applications that surround this technology.

How have the loudspeakers been designed to work with the other elements you’re developing – amplification and DSP/networking/interface?

This project is driven by the loudspeakers. The amplifiers have been closely tailored to the loudspeaker requirement, with the entire hardware and software package designed to complement the loudspeaker.

What are the notable technologies, i.e., waveguides, LF chambers, etc?

The core of the E15 is the e-capsule, with the sound chambers and drivers inside. We spent significant time refining sound chamber performance. Our Kevlar cone technology provides great transient response and in turn delivers very high resolution throughout the entire bandwidth.

Were drivers designed specifically for this loudspeaker? What are they, and are there any special aspects to them?

The YX7 midrange driver was designed specifically for this cabinet. This compression driver is more efficient than anything we have built in the past and it has very low distortion as well. As many mix engineers will tell you, most of the details are found in the mid band, and vocal headroom is crucial. This driver is designed to handle this job without question.

Please describe the rigging system and any independent certifications that it carries.

The rigging is designed by Adamson engineers and then reviewed by a German engineering firm. It meets the most rigid standards of BGV C1.

The beauty of the system is that a single technician can set all the angles on the ground before flying it. When it is lowered to the next group of cabinets, it connects automatically. This system has been met with incredible enthusiasm. 

Are there other features that enhance the portability (or other usage) of these loudspeakers?

Our dolly allows for three different packs depending on how they are arranged in transport. We wanted to offer a U.S. truck pack, European truck pack and a way to ship safely in a sea container.

Will you be using the beta partner approach with these aspects as well?

The existing beta partners will of course be carrying the flag on the introduction of power. In each phase we will have a period of beta testing. We are cautious and will only integrate the technology in a comfortable way for everyone involved.

Adamson Systems Engineering

 

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Monday, May 07, 2012

Team Effort: Keeping In Sonic Step With Blake Shelton On Tour

It may be a small crew, but one that adds value to every move it makes as a team

“We’re all from Nashville,” says Brad Baisley, who, with a name not far off from that of one of country music’s best-known superstars, seems well-suited for stating the obvious of what you’d expect of the sound crew traveling with Blake Shelton, another top artist from Music City. “Maybe we weren’t all born there, but it’s surely home, and the town’s vibe goes with us on the road.”

Imbued with laid-back poise, a down-to-earth demeanor, and a clearly no-attitudes posture, Blake Shelton is a modern poster boy for all that is Nashville past and present.

With his 11th No. 1 single “Drink on It” currently riding high on the airwaves, the three-time Grammy nominated artist migrated at equal altitude across the country on the first leg of his ongoing tour this year, rolling with added vigor and a legion of new fans brought into the fold based upon his appearance as a vocal coach on the NBC reality show “The Voice.”

“Audiences have been amazing,” Baisley, the tour’s monitor engineer relates, “both in terms of sheer number and response. I’ll be at my console with my cue wedge off between songs, and the crowd noise alone is posting 103 dB on my SPL meter.”

Out of necessity, the tour jumped in size exponentially along with the numbers on Baisley’s SPL meter, growing from two trucks to nine earlier this year, the added cartage required to meet the needs of larger venues booked to accommodate Shelton’s burgeoning fan base.

With Sound Image Nashville chosen to provide the audio inventory, beyond Baisley the crew included veteran Jeff “Pig” Parsons at front of house, system engineer Joe Calabrese, system tech Zach Mitchell, assistant rigger/stage tech Dave Shatto, and guitar tech Brett Hardin.

It may be a small crew, but one that adds value to every move it made as a team. “I have one fly guy,” Calabrese says, quickly correcting himself: “Well, make that one-and-a-half fly guys. Given the dual role Dave Shatto plays, I’ve had him half the morning and half the evening. So that’s technically one-and-a-half, then, right?”

Beyond 180

Playing mostly arenas the first three months of the year, the tour deploys JBL VerTec VT4889 large-format line arrays hung 12 deep per side in the main hang, each with three QSC Audio WideLine-10 boxes underhung for down fill.

Crown Audio is the provenance of house power, with a contingent of IT12000HD amplifiers chosen for the task. Eight VerTec VT4880 dual-18 subwoofers receive flight orders as well, with further low-end thunder by eight more VT4880s on the ground.

“Most venues are sizable for this show,” Calabrese explains. “And very few seats are blocked out. We’ve been selling 8,000 tickets on average, and going way past 180 degrees, usually to about 240. I brought along VerTec aux hangs just in case when we started, and we wound up using them 90 percent of the time.”

VerTec V5 DSP presets have made a notable difference, serving as a type of “plug-in upgrade” for the arrays. For his part, Calabrese is of a mind that V5 has given new life to VerTec performance.

“Weighed against other newer options, V5 is insuring that VerTec remains a relevant player,” Calabrese states. “We’re living in an era that’s sort of a renaissance for line arrays.

“The technology is more efficient, predictable, consistent, and standardized than ever before. Things perform exactly how the software says they’ll perform – it’s getting a whole lot easier to be sure that what you see on the computer screen is going to translate directly into real life.

“Performance is reliable using mixed platforms too,” he continues. “Just take a look at this rig. I don’t know of anyone hanging WideLine under VerTec like Sound Image is doing, but it works well. With their 140-degree horizontal dispersion, WideLine is the perfect complement for the wide angle coverage we find ourselves dealing with every night.”

With processing taking place inside the amplifiers, HiQnet System Architect software facilitates connections and control in conjunction with DriveRack 4800 loudspeaker management systems from dbx.

Vocal Attributes

“Big country vocal over the top of a rock band, that’s the essence of what this show is about,” Pig Parsons says. Inside the “pigpen” – his front of house lair – he presides over an Avid VENUE Profile, a board identical to that under Baisley’s command for monitors.

“I pay a great deal of respect to how Blake sounds on his albums within my live mix,” Parsons explains. “He delivers his phrasing live pretty darn close to what’s captured in the studio. His mic technique is fantastic, he knows when to get in close to gain the advantages of proximity effect, and when to back off. Powerful singer too, with wide dynamic range – he can probably whisper louder than most people scream. Intelligibility and diction are superb as well. All of these attributes I find great for mixing.”

Shelton’s wireless handheld vocal microphone – an SKM 2000-XP transmitter topped by an e935 capsule – is, like all other mics on the tour, drawn from the Sennheiser stable. Backed by EM 2050 receivers in the racks, the wireless rig is complemented across the stage by a battery of other transducers, including hardwired e935s for all other vocals except those of a highly talented utility player equally adept at guitar, mandolin, and fiddle, who prefers to hear her voice through a supercardioid e945.

“She sings softly, so the supercardioid mic provides better rejection from the drums,” Baisley explains. “That’s a rather easy problem to solve with just the right mic. Looking at the show in its entirety,
however, one of the biggest challenges I face is that while the entire band is on IEM, Blake prefers wedges. That means I need to strike a balance onstage that allows everything to sound good in both traditional floor monitors and ears. As anyone who has been faced with this situation before knows, that can be a tricky thing.”

Baisley’s strategy to cope with this mixed bag of monitor sources is to rely upon mindful mic’ing choices and placement. One of the lead guitarists, for example, is double-mic’d with a Sennheiser
e906 located off-axis to obtain a darker sound (that wedge-users like Shelton generally prefer), along with a Sennheiser 421 centered right on the cone, which delivers a “brighter” sound that most ear-wearers enjoy.

With two available sources at his disposal, either can be delivered as needed, according to each musician’s preference.

Roaming The Stage

The band wears JH13 earpieces from JH Audio with the exception of the bass player and drummer, who use JH16s.

While the drummer and keyboard player opt to wear the wire, the rest of the band goes with wireless using Sennheiser ew 300 G3 Series systems.

Out front, Shelton absorbs the acoustical energy of proprietary Sound Image wedges, which are 2-way carbon fiber boxes loaded with a single-12 and a horn, and fueled, like the house, by Crown IT12000HD amps. With the wedges firing from both the center of the deck as well as from below up through grates,

Baisley notes that “Blake doesn’t like things extremely loud, and he keys off the house a lot, spending as much time not in front of his wedges as he does. WideLine and its 140-degree horizontal coverage came in handy once again as he roamed the stage, providing him with additional monitoring capabilities, especially midway out on the thrust.”

Baisley adds that he’s automated the entire show for the band. Even though there aren’t a lot of changes among the songs, many require a click-track to be turned up or down, an instrument to be attenuated or have its gain boosted, and so forth.

Plug-ins on his Profile console are straightforward, centering around parametric EQ supplied by Flux Epure II that he uses for overall tone shaping. Saturating plug-ins, such as Bomb Factory BF76 compression, round out his palette.

Beyond his voice, Shelton’s guitar playing starts at the source with signature acoustics from Takamine and a signature electric from Michael Kelly Guitars that features a camo finish, roaming deer track inlays, and a pair of antlers inlaid at the 12th fret. A Mesa Boogie Electro Dyne 2x12 combo is his amp of choice; when combined with a Sennheiser 421 the resulting sound leans toward classic British.

Imaging Process

For his acoustic moments, Shelton recently took possession of a pair of Aura Spectrum DIs from Fishman. Outfitted with imaging technology that brings a studio-mic’d sound to an undersaddle or soundhole pickup, Shelton’s Aura units were custom-equipped with modeling done using one of his own guitars.

“They modeled every note on the entire fingerboard in this perfect studio environment,” guitar tech Brett Hardin says, impressed by the Fishman process, “which I can now blend-in during live performances. The resulting sound is… well, consider that Blake is the kind of musician that doesn’t normally ever say anything about gear so long as it works and sounds good.

“He has, however, already stopped four times during sound checks to express how great he thinks this device is. Someone obviously did something beyond right this time. It’s rare that he’d give any piece of gear this kind of attention.”

On the other hand, attention is constantly focused on Shelton himself these days in a never-ending schedule of showmanship, fine playing, and just plain good times. Is slowing the touring pace slightly over the summer while appearing at a number of fairs a sign that everything may have become a bit too much for everyone?

“Not a chance,” Hardin says with conviction. “I can’t adequately express what this opportunity means to all of us. As a team, this gig is our career song, one we’ll never forget and that will live for us forever. The moment belongs to all of us as well as the fans, and we’re savoring every minute right in step with Blake.”

Greg DeTogne is a writer and editor who has served the pro audio industry for the past 30 years.

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Thursday, May 03, 2012

Transcending Tech: A Conversation With Ethan Winer, Author Of “The Audio Expert”

Getting down to how it really works

The Audio Expert, a new book by Ethan Winer, exhaustively covers a plethora of important technical aspects of audio. But it goes much further, discussing and explaining the relationship between audio and a wide range of closely related factors. In short, it challenges you to think, to seek a deeper understanding.

Just released by Focal Press (and available here), I received an advance copy and have had a hard time putting it down. Winer, who has worked with audio for more than 40 years, is a mix engineer musician, product designer, author (and more), and in 2009, he presented the Audio Myths seminar at the AES Convention in New York that’s still generating buzz.

I recently caught up with him to discuss the book as well as a range of other topics.

KC: What was your primary motivation for writing the book?

EW: Two reasons – one is to dispel the many myths I see repeated endlessly in audio magazines and web forums. Most aspects of audio science have been understood fully for more than 50 years. Yet some people still believe that competent wires can sound different, that typical amounts of phase shift are audible, that jitter is a problem, that digital “summing” in DAW software is somehow flawed, and so forth.

Almost daily I see posts in audio forums by people with limited funds asking if they really need to spend a lot for a microphone, preamp, converter, or external summing box to get professional results. So my goal is as much consumerism as education, to help people spend wisely.

The other reason is to explain how audio really works to those who are interested. Forty years ago, recording engineers were as much “real” engineers as they were recordists. Back then, most knew how to solder up a patch bay and align a tape recorder, and many could read schematics and do at least minor repairs. George Massenburg is a perfect example – he’s renowned for the quality of his recordings, as well as for designing the first parametric equalizer.

When I started recording professionally in the 1970s, audio magazines included technical articles and DIY plans, and manufacturers were proud of their high fidelity and provided specs for distortion and frequency response. Today, a loudspeaker review is likely to state the size of the woofer but not its frequency response, which of course is what really matters! And you almost never see distortion specs or off-axis response. If an active loudspeaker includes distortion specs, it’s usually for the power amplifiers only, not the complete system.

Many mix engineers have the talent to make music sound great, but without understanding the engineering and science behind the gear they use. I appreciate that some people don’t care at that level, but many do. In my estimation, the pro audio press has let us down in this regard, dumbing down content, and even perpetuating many of the same myths you read in hi-fi type magazines.

Your experience is more with recording than live sound. What’s the value of the information you’re providing for the live sound practitioner?

The Audio Expert is a comprehensive “reference” type book covering all aspects of audio, so there’s plenty for everyone – even interested audiophiles. It’s written for people who want to understand audio at the deepest, most technical level, but is presented using plain-English explanations and mechanical analogies with minimal math.

Besides describing how many different audio devices are used, it also explains how they work internally. The book brings together the concepts of audio science, aural perception, musical instrument physics, acoustics, and basic electronics, showing how they’re intimately related. So while I don’t address directly the challenges facing live sound engineers, there’s a huge amount of educational content. It’s definitely not a “Dummies” type book for beginners!

If you could recommend one chapter as the “must read” of the book, what is it, and why?

Perhaps most important is explaining in great detail how fidelity is defined, with included audio examples people can play on their own systems to determine at what level distortion and other artifacts are audibly damaging. This is addressed mainly in Chapters 2 and 3, though this type of information is sprinkled liberally throughout the book.

Besides the 65 demo audio files available on the book’s web site, there are also 31 videos and five audio-related software programs.

What’s the single biggest misconception or “myth” about audio?

The two biggest myths are probably that there are aspects of audio fidelity that “science” hasn’t yet learned how to measure, and that listening is a more reliable way to assess the quality of gear than measuring. I see magical thinking all the time in audio forums, but it’s easy to prove that everything affecting the fidelity of audio devices is already known.

A spectrum analyzer can display artifacts 100 dB below the music, and is highly reliable and repeatable, versus human hearing that varies from moment to moment, and is influenced by the masking effect. Many types of distortion and other artifacts can be very difficult to hear, even when they’re only 40 dB below the music.

What sources proved most valuable as you wrote and assembled 650-plus pages of significant technical information provided in the book? How did you fact check and verify?

The book actually totals 739 pages when including the three bonus chapters online. I’ve been involved with audio for many years as a recording engineer, circuit designer, and computer programmer, so I already had a solid grasp of the science. But I did learn a few things! I was fortunate to get advice from microphone expert Bruce Bartlett and loudspeaker expert Floyd Toole.

Another friend, electronics engineer John Roberts, read my entire manuscript as I wrote it, and audio expert Mike Rivers did the technical review. All of these people provided invaluable suggestions and fact checking.

How do you clearly separate what is objective in audio versus what is subjective?

Subjective preference is impossible to define, so I don’t even try. I do address some aspects of preference, such as the perceived improvement after adding acoustic treatment. But mostly I address the science of audio, and explain how audio circuits and their plug-in equivalents are used and how they work internally.

It’s impossible to “measure” the quality of a piece of music, or assess one’s enjoyment. But it’s absolutely possible to assess fidelity, even when a perfectly clean sound is not the artistic goal.

In your view, what are the differences between analog and digital audio in terms of sound quality?

First we have to define what is meant by analog and digital. “Analog” encompasses both audio hardware such as equalizers and compressors, as well as the recording mediums of magnetic tape and vinyl records.

Digital audio refers to both the recording medium and software effects. Analog gear can be very high quality, with distortion and noise low enough to not hear, and a frequency response flat enough to not matter.

Gear that meets these criteria is considered audibly transparent, such that it’s difficult to notice a change in quality after passing through the device.

Digital plug-ins have a slight advantage because their transparency is dictated entirely by the resolution of the math used to perform the needed calculations. Most modern software processes audio data using 32-bit floating point numbers. This is potentially cleaner than any real-world electronic circuit.

Of course, every computer sound card and outboard A/D/A converter has analog input and output sections, and these ultimately limit the fidelity possible. But many converters are audibly transparent. So the real answer is that both analog and digital can have acceptably high fidelity when implemented properly.

Another important factor falls outside the context of “sound quality” – intentional subtle distortion used for effect to add faux clarity to a track or complete mix, or as “glue” to make a mix sound more cohesive. A compressor with both the attack and release times set very fast also adds distortion that is useful in some contexts. These effects can be implemented effectively using either digital or analog technology.

What are the best ways to learn the essential principles of audio?

The best way to learn is by doing. It also helps to have knowledgeable friends, whether in person or a web forum. Of course, the downside of web forums is having to sort through many disparate opinions to separate fact from belief. But anyone who has basic audio software can easily try things for himself or herself.

Often I’ll see someone in an audio forum ask, for example, if they should compress before EQ or vice versa, or use EQ boost rather than cut – even though it would be trivial to just try it to find out for yourself firsthand!

If a tree falls in the forest and no one is there to hear it, does it make a sound?

Yes!

For a much more detailed and interesting answer to that last question, be sure to check out The Audio Expert, published by Focal Press (ISBN: 9780240821009) and available here.

And, go here to read an excerpt chapter entitled Audio Fidelity, Measurements, And Myths - Part 1, provided exclusively to PSW.

Keith Clark is editor in chief of ProSoundWeb and Live Sound International.

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Line 6 Now Shipping XD-V55 Digital Wireless Systems

Line 6 is now shipping the new XD-V55 digital wireless handheld, lavalier, and headset microphone systems.

Featuring microphone modeling technology, XD-V systems offer 24-bit, 10 Hz–20 kHz, compander-free performance. .

The family offers a full complement of professional features including signal encryption, dynamic filters, gain control, channel scanning and more.

Utilizing the same 4th-generation digital wireless platform as the flagship XD-V75, the XD-V55 family offers handheld, lavalier and headset systems and the compact, portable XD-V35 family includes handheld and lavalier systems.

“For performers who want wired mic audio performance and wireless freedom, the combination of Line 6 modeling and our class-leading digital wireless platform makes the latest XD-V systems the only choice,” says Steve Devino, live sound product manager at Line 6.

“Proven on countless stages and tours worldwide,” he continues, “fourth-generation Line 6 digital wireless technology ensures the best possible performance experience with crystal-clear audio, rock-solid reliability and simple, license-free operation – worldwide.”

Ensuring faithful reproduction and full-range audio clarity, XD-V systems all provide 10 Hz - 20 kHz frequency response and wide dynamic range (up to >120 dB). They do not use companders or compress the audio signal in any way, and audio quality does not degrade with distance.

XD-V systems operate in the 2.4GHz band, which is free from interference due to TV broadcast, public safety announcements, cell phone towers and other transmitting devices. Encoded DC (Digital Channel Lock) technology prevents reception of any audio interference from other 2.4 GHz devices.

XD-V handheld systems feature a selection of up to 10 models of popular vocal microphones.  Using this incredibly diverse sonic palette, vocalists can choose the perfect microphone sound to match their voice and style of performance.

For active spoken-word performers, instrumentalists or singers who require a hands-free solution, XD-V bodypack systems offer selectable EQ filter models, tailored for a wide range of vocal and instrumental applications. 

XD-V55 bodypack systems have three selectable vocal EQ filter models.

XD-V series digital wireless systems are incredibly easy to operate. Simply choose a channel on the transmitter and receiver and they lock together automatically. There is no need for RF tuning or intermodulation calculators.

XD-V55 family features: 12 channels, 300-foot range; 1/2U desktop receiver with externally mounted antennas; heavy-duty metal chassis.

Line 6

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Tuesday, May 01, 2012

US Tour Of Million Dollar Quartet Powers Up With Meyer Sound MICA And M’elodie

Having continued successful permanent productions in Chicago and New York, the Broadway musical Million Dollar Quartet (MDQ) has hit the road with a North American tour. To accommodate venues seating from
1,400 to over 3,000, sound designer Kai Harada specified a Meyer Sound system based around MICA and M’elodie line array loudspeakers.

“My goal was to give the touring crew a system with a lot of flexibility, so they could adjust coverage angles for theatres of all shapes and sizes,” explains Harada. “With the M’elodie center cluster and MICA side arrays, they have all the power they need for bigger houses, along with reliable consistency of sound—a quality at which Meyer systems always excel.”

Inspired by a 1956 recording studio jam session by Elvis Presley, Carl Perkins, Johnny Cash, and Jerry Lee Lewis, Million Dollar Quartet captures a pivotal night in the history of rock ’n’ roll. On the MDQ tour, the lion’s share of the show’s energy is delivered by a split dual center array of 20 M’elodie loudspeakers and the upper and lower side arrays comprising a total of 20 MICA loudspeakers. A left-right configuration of 600-HP subwoofers and a 700-HP at center provide low end, while a total of 16 UPM-1P and UPJunior VariO loudspeakers supplies fill and delay systems as needed. A Galileo loudspeaker management system with four Galileo 616 processors provides drive and optimization.

“This is a rock ’n’ roll show, but it was my goal to preserve the dynamic between the book scenes, the songs, and the big finale,” Harada says. “It’s important to hold a lot of punch in reserve, and this system certainly has it.”

Harada is also the sound designer for resident productions of MDQ at Chicago’s Apollo Theater and off-Broadway at New York’s New World Stages. The New York system is based around a MINA line array in the center with CQ-1 and CQ-2 loudspeakers on the sides, while the wide thrust staging in Chicago also employs CQ-1 loudspeakers with smaller UltraSeries models for delays and fills. PRG Audio supplied all three systems for the touring and resident productions.

Despite the radical differences in venues, Harada credits the Meyer Sound systems with maintaining a uniform sound. “For me, it comes down to consistency and transparency,” he says. “I can focus on bringing the audience closer to what is happening on stage with the confidence that the system won’t adversely color their experience.”

Harada also specified Meyer Sound systems for MDQ’s well-received 2010-11 run on Broadway at the Nederlander Theatre and in London at the Noël Coward Theatre in 2011-12.

The book for Million Dollar Quartet was written by Floyd Mutrux and Colin Escott. The Broadway production was nominated for three 2010 Tony Awards, with Levi Kreis (Jerry Lee Lewis) winning Best Featured Actor in a musical.

A long-time associate of Broadway sound designer Tony Meola, Kai Harada currently supervises sound for all productions of Wicked, and has designed around the world using Meyer Sound systems, including Hinterm Horizont in Berlin, and the critically acclaimed revival of Follies on Broadway.

Meyer Sound

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Vienna’s Burgtheater Relies On Audio Quality Of DPA d:fine Microphones

The Burgtheater in Vienna has added five DPA Microphones d:fine omnidirectional headsets to its existing armory of more than 70 DPA microphones. The d:fine microphones were purchased from DPA’s Austrian distributor, Studer Austria, for productions such as the prominent discussion televised panel Debating Europe, where top politicians and experts discuss the future of Europe.

The d:fine headsets are being put through the paces at the Burgtheater, which comprises three separate stages: the 1200-capacity Burgtheater itself, the 400-capacity Akademietheater and the 350-capacity Kasino. A smaller, fourth stage, known as the Vestibuel, can seat 65. Performances and presentations are staged almost every day in every theater, with nearly 700 events held per year. Most recently, the DPA d:fine headsets were called upon for political debates – which require a more personal approach to the presentation.

“We achieved excellent results using our new d:fine headsets for the latest political discussion panel and also had very positive feedback from the people wearing them,” says David Muellner, head of the Burgtheater’s sound department. “Our older headsets had elements which were not perfect for our needs. For instance, the size of each actor or politician’s neck varied, so we had to adjust the headband mount, which took time and effort - something you don’t have when the prime minister is standing in front of you!”

For users looking for the perfect headset, the d:fine is available in both directional and omnidirectional options, in single- or dual-ear designs, with long or short microphone booms and in black, brown, beige and the company’s distinctive lime green color. Each solution performs spectacularly in either a cabled or wireless solution.

Ideal for everything from concerts to presentations, the DPA d:fine has set the benchmark for headset microphones, with its no-compromises approach to sound, size and design. The tiny, unobtrusive capsule provides a highly-detailed signal and its comfortable yet snug ear mount features low sensitivity to wind and background noise.

“Often politicians would say, ‘Please, none of these big ugly headsets, I’d prefer to speak with a lapel mic,’” adds Muellner. “Now that we have the DPA d:fine headsets, I have reassured them that it can be mounted to be almost invisible. An added plus for us is that we don’t need to worry about adjusting the headset to people of different sizes, or self-noise such as ladies wearing earrings, which can knock against the head mount. All of these small but incredibly important details are the reason why the new d:fines are a

real breakthrough! The d:fines are so lightweight that even very skeptical actors forget that they are wearing a headset mic.”

The theater also has six DPA 4088 and eight DPA 4066 headsets microphones, which make an appearance in every production that uses wireless mics, especially for speech and singing. Around 50 DPA 4061 miniatures are used extensively with the venue’s wireless systems, while the DPA d:vote 4099V for violin and 4099P for piano instrument clip microphones are deployed for musical productions.

DPA Microphones

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Monday, April 30, 2012

Making It Work: Strategies In Optimizing A Live Club System

Ways to can work around a room and system's shortcomings to deliver high-caliber sound reinforcement

McGonigels Mucky Duck is one of those venues that bands, engineers and fans love. An Irish-style pub in downtown Houston, it’s stage is noted in the folk, jazz, Americana and World Music spheres and has played host to the likes of Joe Ely, Shake Russell, Radney Foster, Kinky Friedman, James McMurtry, Druha Trava, Sarah Jarosz, The Magpies, Iris Dement, Michelle Shocked, Leon Redbone, and hundreds of others.

The official capacity is 140, which in my opinion, would be very packed and uncomfortable. They sell out the 100 or so table seats very fast for most shows, and the remaining tickets are for standing room only. As the website eloquently puts it, “If another chair would fit it would already be there. Sorry, but you can’t bring your own chair.”

The crowd is respectful. To help those who may be visiting for the first time, there are placards on each table reminding people to be quiet, and to silence cell phones. This is one of the best things about the Mucky Duck. They respect the artists. It’s the very definition of a listening room.

The place is not without drawbacks though, and here, I’ll outline some of those shortcomings and discuss ways that visiting engineers can work around them in order to deliver high-caliber sound reinforcement.

Dialed In

Saying the stage is small is being somewhat generous – I’ve seen bigger drum risers. It’s located in a corner, which is both a blessing, in the form of a little extra real estate, and a curse, because the drums will always be too loud. Heavy theatrical drapery around the perimeter of the stage helps quiet reflected stage wash but does little to dampen the natural volume of the drums. The low ceiling doesn’t help either.

There is no house engineer, and management knows precious little except how to get the background music on. Bands are expected to bring their own mix engineer, and that person is usually the sound tech as well. The job is either easy or difficult, depending on the condition the previous engineer/tech left it in.

Luckily the level of talent booked here means that the system is usually zeroed out, and the stands and cables are neatly put away. Almost everyone leaves it a little better than they found it, which is also my philosophy – I don’t want the next guy to work any harder than he has to.

Both house and monitor systems use EAW loudspeakers, with four SM159z wedges on four separate mixes derived from front of house. Each mix has a 15-band EQ inline. With only 15 bands on the monitor sends, precise feedback taming will not happen. I bring a few XLR “wyes” in order to split the important inputs into two separate channels.

Using different channel strips – one EQ’d for the mains and one for monitors – allows much finer control over the house mix versus the monitor mix.

Main loudspeakers are FR159z, one each hung on the wall stage left and stage right, and two more toward the rear of the room providing fill. There are no subwoofers. And, there’s also no time-alignment on the front or rear loudspeakers, and no processor available to do so.

I solve this problem by bringing in a QSC DSP-4 digital processor, which I insert and use to set a 9-millisecond delay on the front loudspeakers.

When using a house system to supplement the stage volume, instead of overpowering it, delaying the mains to arrive in time with the band is the way to go, at least in my view.

It helps the PA “disappear” and leaves the impression that the band is making all the noise.

I also set a 21-millisecond delay on the rear loudspeakers. With each of them a different distance from the front, I choose a delay time that splits the difference.

The downside of this is minimal because the improvement is quite dramatic, a huge benefit, and no one notices that the rear fills arrive a few milliseconds apart.

The 2-input by 2-output DSP-4 works out great – there aren’t all that many compact DSP units that can be controlled with a laptop available at a reasonable price point on the used gear market. I also like that it’s small enough to fit in my briefcase and uses standard XLR inputs and outputs. (In fact, I like it so much that I own two.)

Stage Details

My most recent trip to the Mucky Duck was to support a performance by Max Stalling, a native Texan with a unique musical style that rolls from two-stepping dance numbers to Spanish-guitar-heavy folk music (à la Marty Robbins), with a few waltzes mixed in.

Max sings and plays the acoustic guitar, and is backed by a three-piece rhythm section comprised of Jason Steinsultz on upright bass, Jeff Howe on drums, and Bryce Clark on lead guitar, switching between mandolin, steel string and gut string acoustics, and electric guitar. Both Jason and Bryce sing harmony, and steel guitar player Hank

Early also sat in for this show, I used the band’s own Shure Beta 58s microphones for vocals, a house-supplied AKG D112 on kick, and Shure SM57s on electric guitar and steel. The upright bass and three acoustics all had band-supplied Radial Tonebone preamps and ran direct.

Just one of my AKG 451e condensers was used for drum overhead, to capture the kit as a whole. Jeff (the drummer) switches between sticks, brushes and even sometimes wrapped mallets. I will heavily compress the overhead (remember, the acoustic drum sound is still dominant in the room) so that the details on the brushes and mallets are not lost. Having at least one drum mic also lets me add reverb to this very dry room.

Max is very particular about his monitor mix. Some engineers take this as being a “prima donna” but I’ve found it to be exactly the opposite. He knows what he wants and isn’t afraid to ask. He can’t state specific frequencies, so there’s a bit of interpretation needed to get his mix the way he wants it, but once he’s comfortable, that’s pretty much that.

The tone that Max wants out from his monitor is not exactly what you want at front of house. He likes things a little dark with plenty of low mids for both his vocal and guitar. Two of the XLR wyes allowed me to split his vocal and acoustic channels so that I was able to give him exactly the tone he wanted in the monitor by using the channel strip EQ. Then I had use of the 15-band graphic for his mix to tame the little bit of feedback that tried to creep in.

Starting Quiet

After getting the monitors set, I build the front of house mix. The best way to mix in this room is to listen to what’s coming off the stage and only add what’s needed. Trying to overpower the stage volume is a losing battle.

I always start my sound check with the house PA off and just listen to what’s happening on stage, and then work to fill in the missing bits that will help make the performance “pop.” I’ve noted several times that it seems like I’m cutting too much low-mid out of the house, but that’s usually O.K. because the monitors provide all of the low-mid energy one could ever want.

I get the vocals up to a good level, over the stage volume, and only after do I work in the other instruments. Generally the drums and amplified instruments are fine coming straight off the stage. On the recent gig with Max, I needed a touch of the electric guitar and steel, but none of the bass and kick drum.

Jeff plays a kit of Slingerland Radio Kings from the 1940s. These drums are big and loud. The kick is a huge 14 inches by 26 inches and uses a ported head. (Back in the “good old days” the drums had to fend for themselves, and this set gives you all the stage volume you need!)

But here, because were already plenty loud in the house, I used the overhead high-passed around 100 Hz and compressed at a 6 to 1 ratio with about 12 to 15 dB of reduction on the loud parts to add definition and to help keep the mix cohesive all the way to the back of the room. It was also used to feed the reverb.

Same Room

The house console is a 24-channel Allen & Heath GL2400, a step above what you find in many clubs the size of the Mucky Duck. Most bands will not fully mic the drums, so 24 channels gives me plenty of room to split channels as needed.

I maximize the two available channels of compression on the venue’s dbx 1066 by inserting each on a bus and assigning several like channels to that bus. I used one compressed bus for the drums and another for the lead acoustic and a gut string acoustic that are featured prominently in the band.

There are also two Lexicon effects units – MPX110 and MX400 – on hand to add ambiance. To create a sense of space in this dry, tightly packed room, I used a trick that I’ve implemented in most of my live mixes for the past couple of years. I select a very short and transparent “room” style reverb and send the entire band to it, typically applying about a half second of decay and zero pre delay.

Then, I bring it up in the house until it can be heard clearly, then back it down to just on the edge of being noticed. If the reverb is muted, a change can be heard, but it’s not something you can pinpoint. I find that this really takes a tight mix and glues it together even more – the band is all playing in the same “room” together because they all have the same decay time.

If you’re lucky enough to have a gig in the Mucky Duck, be prepared to bring your A game. It’s bit challenging, but once the mix is dialed in you can be sure you’re mixing for a crowd that truly appreciates what you’re doing.

Tim Weaver is the owner of Weaver Imaging, an audio, lighting, and projection provider based in College Station, TX. He has been a professional sound engineer for 18 years, working across all genres.

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Friday, April 27, 2012

Size Matters: Observations On Loudspeaker Directivity

Physics hasn’t changed.... When it comes to pattern control, size still matters!

Trap boxes and line arrays get all the attention. And that’s no surprise - they’re big and loud, and dare I say it, glamorous.

But the truck rarely rolls without a complement of two-way loudspeakers sporting a 12-inch or 15-inch woofer and a horn.

Whether its monitor wedges, drum fill, front fill or just “speakers on sticks,” small 2-way boxes do many of the everyday jobs that make up a typical sound reinforcement day.

We take the performance of these boxes for granted, but they can be used to better effect if we really understand their directivity characteristics and what makes them perform the way they do. They’re often described as a 90 by 60 box or some other dubious reference.

But 90 degrees by 60 degrees at what frequency? Certainly not from DC to light.

There are four principle ingredients that govern the dispersion pattern of these loudspeakers, including the cone driver, horn, crossover and cabinet.

Let’s look at these one at a time and assess their contributions. Before we go through our list, though, let’s review some basics.

The amount of directivity any device can exert on a sound wave is directly related to the proportional sizes of the device and the sound wave.

To understand this relationship it is important to have a good grasp of how big or small a sine wave is at a given frequency.

Sound at sea level at 72 degrees Fahrenheit travels at approximately 1,130 feet per second. We express frequency or cycles (sine waves) per second as Hertz.

So if the frequency of a wave is 1 Hz, the wave is 1,130 feet long. Logically, a 10 Hz wave is 113 feet long, a 100 Hz wave is 11.3 feet long, and a 1,000 Hz wave is 1.13 feet long, etc.

While it’s not overly difficult to do the math to determine the wavelength of any given frequency, there is an old “cheat” called the rule of 5-2-1:

20 Hz = 50 feet
50 Hz = 20 feet
100 Hz = 10 feet
200 Hz = 5 feet
500 Hz = 2 feet
1,000 Hz = 1 foot
2,000 Hz = .5 foot
5,000 Hz = .2 foot
10,000 Hz = .1 foot

While not perfectly accurate, it fills the bill for “quick and dirty” calculations. Physics dictates that a source be physically large in comparison to a wavelength to exert directional control over it.

So let’s look at the low frequency directivity of a 12-inch driver in a 2-way loudspeaker with a 90-degree by 60-degree horn.

Matter Of Control
Remember that the low frequency driver’s only means of controlling the dispersion of the sound wave in a front-loaded loudspeaker are its cone diameter, and to a lesser extent, some boundary effects (we’ll discuss that later).

At 100 Hz, the driver is physically small in comparison to the 10-foot wavelength and provides almost no directivity (Figure 1).

If we increase the frequency gradually, the 12-inch driver does not suddenly exert pattern control over the sound wave when it reaches 1,000 Hz (1 foot), and is the same size as the driver itself.

Rather, it has more and more effect as the frequency gets higher and the wavelengths get shorter. (Figures 2 & 3)

In this frequency range (800 Hz as shown in Figure 3), the cone driver is actually providing approximately 90-degree horizontal dispersion.

But also realize that since this pattern is conical (the driver is round), it is not producing the specified 60-degree vertical pattern.

As the frequency increases the driver exerts more and more control until it begins to “beam” at higher frequencies.

But by the time it narrows that much, it’s above the crossover frequency.

This particular loudspeaker crosses over about a half-octave above the balloon in Figure 3.

This has an overriding effect on the polar behavior of the box, especially in the vertical domain, so we will discuss the range from 1,000 Hz to 1,500 Hz when we discuss the crossover. Now, on to the horn.

Dominate The Wavelength
There are multiple elements in a horn’s design that contribute to its ability to achieve pattern control at a given frequency.

Some of them are throat geometry, length and flare rate.

But the most obvious factor is the size of the horn mouth. The same rules apply here as to the cone driver. Size matters.

The horn mouth must be large enough to dominate the wavelength in question in order to provide complete directivity at that frequency.

So if a horn mouth is 6 inches wide by 3 inches tall it will be somewhat omnidirectional at 1,000 Hz.

It will not dominate the sound wave until the frequency reaches about 2,000 Hz in the horizontal plane and 3,000 Hz in the vertical plane.

It may provide a 90-degree by 60-degree pattern above 3,000 Hz, but almost certainly not at lower frequencies.

Cone drivers and horns by themselves are fairly predictable devices. But combining the two in close physical proximity can be quite challenging.

The first problem is physical offset. In a typical 2-way box, the devices are located one above the other ,and may also be at different depths.

Even if we use delay to correct the time alignment between the drivers on axis, any other vertical angle will skew the time arrivals from the horn and the cone driver.

Because the bandpasses and vertical dispersion patterns of the drivers necessarily overlap in the crossover region it is probable that at any vertical angle that is off axis we will be hearing contributions from both drivers out of phase.

This means there will be lobes and nulls. (Figures 4 & 5)

This particular box was crossed over at 1,350,Hz with a symmetrical Linkwitz-Riley 24 dB slope.

These lobes will vary in direction and intensity based on driver offset and pattern control, crossover slope, and overlap and alignment delay settings, but they will always occur in multiple driver boxes with physically separated sources.

If a cabinet is laid on its side we get the same phenomena in the horizontal plane. Floor wedges, anyone?

This is one reason there has been a resurgence in coaxial boxes.

Because there is no vertical offset between the sources, we only have to correct for the variation in depth between the acoustic origin of the cone and the horn driver, and that distance stays more constant with off-axis listening positions.

The trade-off is that many coaxial designs use the driver cone as the horn flare to guide the high frequencies, and while this may be fine for monitors or other near-field applications, more precise pattern control is often required for sound reinforcement duties.

Baffles, Boundaries
The final piece of the directivity puzzle is the cabinet itself and the boundary effect created by setting it on something. Fractional space loading is created when we decrease the space that a device is radiating into.

As we saw in Figure 1, low frequencies are omnidirectional, so when we set a loudspeaker on the floor, we effectively halve its radiating space at low frequencies. This produces an additional 3 dB of output (double the power) in the hemisphere that it is still exciting.

If the baffle on the cabinet is physically large enough versus a given frequency, it can act as a boundary to create half space loading. This is what is sometimes called “baffle step.”

In modern cabinets, the baffle is rarely much larger than the driver that is mounted in it, because generally, priority is given to things like weight, truck pack, handle location, flying hardware, arrayability and profile.

Technology has gone a long way towards providing a ton of output and fidelity from small packages. But physics hasn’t changed. When it comes to pattern control, size still matters!

Bruce Main has been a systems engineer and front of house mixer for more than 30 years, and has also built, owned and operated recording studios and designed and installed sound systems.

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Wednesday, April 25, 2012

Clear-Com Launches HelixNet Networked Partyline Intercom System

Clear-Com has announced the worldwide release of the HelixNet Partyline, the industry’s first networked partyline intercom system with a set of unique capabilities for achieving greater efficiency, cost-savings and flexibility from set up to operation and maintenance.

HelixNet Partyline provides digitized Clear-Com sound and central administration of the entire system (firmware upgrades and maintenance) from the Main Station with a single cable and flexible cable options, with the ability to leverage an existing cable infrastructure.

The initial release of HelixNet Partyline consists of the HMS-4X Main Station, HBP-2X HelixNet beltpack, HLI-2W2 two-wire interface module and the HLI-4W2 four-wire interface module. The system begins shipping in June of 2012.

“For over 40 years, the Pro Audio community has been using the common, three-pin XLR microphone cable to carry audio for analog partyline systems,” says Chris Barry, product manager at Clear-Com. “In order to preserve our customers’ investments in intercom systems and cabling infrastructure, we had specifically designed HelixNet Partyline to transmit four channels of digital quality audio, plus program and power for beltpacks, over a single, shielded twisted-pair cable (ex. microphone cable, Cat-5 or Cat-6 cable). This capability alone is unprecedented in the history of intercom technology.”

The HelixNet Partyline system also offers many unique features to create a much higher audio quality, increase efficiency during the set up and maintenance processes and simplify operations.

HMS-4X HelixNet Main Station and Interface Module

• High channel density and high user capacity. The sleek 1RU HMS-4X HelixNet Main Station fits into any standard 19” rack and can provide power and four channels of audio to support up to 20 digital beltpacks.

• No hum. No buzz. Unlike standard analog systems, the all-digital HelixNet system is immune to electro-magnetic interference and ground loops.

• Highly flexible and offers intuitive user operations. System settings and menus are quickly accessible. Firmware maintenance and upgrades can be achieved easily via USB ports.

• Greater connectivity with existing analog intercom systems and audio devices. The expansion bay in the Main Station allows optional HLI-2W2 two-wire and HLI-4W2 four-wire interface modules to connect easily with existing analog intercom systems and audio devices, while preserving the high audio quality that is free of hum and noise.

HBP-2X HelixNet Beltpack

• High channel density and selectable channels to save resources. The rugged, ergonomically-designed HBP-2X HelixNet Beltpack is a two-channel beltpack that can access two of any four system channels and program audio over a single cable, along with individual level control. Networked audio is distributed over a single, shielded twisted pair, keeping the number of cables required, low.

• Easy to operate and read. Optimally positioned buttons and volume knobs are easy to locate, identify and control on the beltpack. Channel labels are simple to read on the high-contrast 10-character OLED displays. Beltpacks can be set up in daisy chain or star configurations with no need for active split boxes.

• Durable and flexible. HelixNet Beltpacks are highly durable, fabricated from lightweight cast aluminum, and come with a sturdy beltclip, rubber bumpons and an integrated strap guide to offer a variety of practical mounting options.

Clear-Com
HME

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Tuesday, April 24, 2012

Church Sound: The Sound Operator’s True Purpose And Role In Worship

For those of us who support the technical side - how can we help a ministry stay on track?

What am I trying to accomplish?

It’s a question we should probably be asking ourselves, as church sound operators, on a daily basis.

In today’s rapid-paced society, where exciting new changes are mixed with fear and turmoil, the result can be a feeling of spinning in circles with no direction or purpose.

This is quite visible at many churches. It’s almost become a marketing battle as to who can do the most to keep people coming in the doors.

As a result, some are willing to compromise beliefs and convictions in favor or appearing “relevant” to mainstream society.

But shouldn’t it be the other way around? Shouldn’t the church be teaching it’s people to make their lifestyle and the “real world” relevant to their faith?

Now let’s narrow this discussion down to those of us who support the technical side of ministry. How can we help a ministry stay on track?

Imagine making a trip by boat from England to New York City. The course is set and you’re on your way, but somehow, the direction of the boat is just a couple of degrees off.

Where do you end up? Most certainly not where you wanted to go.

Working in ministry can be the same way. The problem is that sometimes we don’t realize we’re off-track until we arrive at the wrong place. I’ve come to observe that church sound and technical volunteers (as well as their paid counterparts) face constant danger of going off course.

Who Knows What
Why is this so? Simply, it’s very easy to get caught up in new gear, technologies and theories.

These types of things are always changing, and there are hundreds of different opinions floating around about all of it.

And it may be hard to admit, but it’s true: these are the things that most certainly are not the secret to success in audio ministry.

Don’t get me wrong - quality equipment, properly applied, is essential. Returning to my trip analogy, it sure would be a tough journey to New York if you tried to make it on a surfboard. The same goes for sound systems.

But our job is to be educated about show knows what they’re talking about, and who can provide us with the right tools - not the latest and coolest tools.

Success in all endeavors is greatly determined by whom we associate with. Choose wisely, not just technically or economically. Unfortunately, those who may purport to give advice about sound sometimes doesn’t even know what they don’t know.

We must understand our role within ministry as a whole, and be completely committed to it with excellence. Christianity suffers today because it is horribly misunderstood, mostly due to the fact that it’s poorly communicated.

We all agree that church should always be a venue in which to communicate the truth of Christianity.

But if one attends church and the message is not clearly communicated, we are off track.

Those delivering the message have the obvious responsibility of making sure it is consistently correct.

These folks, however, are dependent on the sound operator to make sure the message is delivered clearly to the congregation.

By the way, when I use the word “message,” I’m not simply referring to the sermon.” Worship and music themselves are also part of the overall message.

Our role is to make sure that all of this is accurately amplified, helping the congregation join in the worship experience. A colleague once compared running sound for a rock concert to running sound in church as the difference between mixing for an audience and mixing with an audience.

Therefore, every time we are at work at the console, or plugging in a microphone on the platform, we must ask ourselves what is our role, and what are we really trying to accomplish?

Role Playing
A high level of value must be placed on attitude. All too often, I’ve witnessed a frightening level of alienation between worship leaders, musicians, pastors and the “tech folks.”

Many view the role of sound operator as one that should be invisible. However, tho those of us actually working with sound, that role is far from invisible.

The best way to change this problem is to change our attitude, always look at at things from everyone’s perspective, rather than just our own.

At the same time, sound operators should never view their role as secondary or insignificant, regardless of how anyone else sees it. But if the worship leader calls it a support role, that’s OK, because is a support role to and for that individual.

It doesn’t make any difference who gets the credit, or who gets talked about at Sunday dinner. What the sound operator does matters, even if others don’t recognize it.

Our most important mission remains the same. Every time we take our position of supporting worship at our church, the only question we need to ask is simple: “What am I trying to accomplish?”

The answer will always show us the way.

Rob Stam has served as an AV system designer and installer for more than 20 years, and has been active as both a musician and sound operator as a church member.

 

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Friday, April 13, 2012

Why Not Wye? When Combining Two Signals Into One Is Not A Good Idea

Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right.

 

Wye-connectors (or “Y”-connectors, if you prefer) should never have been created.

Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right.

A wye-connector used to split a signal into two lines is being used properly; a wye-connector used to mix two signals into one is being abused and may even damage the equipment involved.

Here is the rule: Outputs are low impedance and must only be connected to high impedance inputs—never, never tie two outputs directly together—never.

If you do, then each output tries to drive the very low impedance of the other, forcing both outputs into current-limit and possible damage. As a minimum, severe signal loss results.

“Monoing” Low End
One of the most common examples of tying two outputs together is in “monoing” the low end of multiway active crossover systems. This combined signal is then used to drive a subwoofer system.

Since low frequencies below about 100 Hz have such long wavelengths (several feet), it is very difficult to tell where they are coming from (like some of your friends). They are just there—everywhere.

Due to this phenomenon, a single subwoofer system is a popular cost-effective way to add low frequency energy to small systems.

So the question arises as how best to do the monoing, or summing, of the two signals? It is done very easily by tying the two low frequency outputs of your crossovers together using the resistive networks described below.

You do not do it with a wye-cord.

Summing Boxes
Figure 1 shows the required network for sources with unbalanced outputs. Two resistors tie each input together to the junction of a third resistor, which connects to signal common. This is routed to the single output jack.

The resistor values can vary about those shown over a wide range and not change things much. As designed, the input impedance is about 1k ohms and the line driving output impedance is around 250 ohms.

The output impedance is small enough that long lines may still be driven, even though this is a passive box. The input impedance is really quite low and requires 600 ohm line-driving capability from the crossover, but this should not create problems for modern active crossover units.

The rings are tied to each other, as are the sleeves; however, the rings and sleeves are not tied together. Floating the output in this manner makes the box compatible with either balanced or unbalanced systems.

It also makes the box ambidextrous: It is now compatible with either unbalanced (mono, 1-wire) or balanced (stereo, 2-wire) 1/4-inch cables.

Using mono cables shorts the ring to the sleeve and the box acts as a normal unbalanced system; while using stereo cables takes full advantage of the floating benefits.

Stereo-to-Mono Summing Box
Figure 2 shows a network for combining a stereo input to a mono output. The input and output are either a 1/4-inch TRS, or a mini 1/8-inch TRS jack. The comments regarding values for Figure 1 apply equally here.

Balanced Summing Boxes
Figures 3 and 4 show wiring and parts for creating a balanced summing box. The design is a natural extension of that appearing in Figure 1.

Here both the tip (pin 2, positive) and the ring (pin 3, negative) tie together through the resistive networks shown.

Use at least 1 percent matched resistors. Any mismatch between like-valued resistors degrades the common-mode rejection capability of the system.

Termites In The Woodpile
Life is wonderful and then you stub your toe. The corner of the dresser lurking in the night of this Note has to do with applications where you want to sum two outputs together and you want to continue to use each of these outputs separately.

In other words, if all you want to do is sum two outputs together and use only the summed results (the usual application), skip this section.

The problem arising from using all three outputs (the two original and the new summed output) is one of channel separation, or crosstalk. If the driving unit truly has zero output impedance, then channel separation is not degraded by using this summing box.

However, when dealing with real-world units you deal with finite output impedances (ranging from a low of 47 ohms to a high of 600 ohms).

Even a low output impedance of 47 ohms produces a startling channel separation spec of only 27 dB, i.e., the unwanted channel is only 27 dB below the desired signal. (Technical details: the unwanted channel, driving through the summing network, looks like 1011.3 ohms driving the 47 ohms output impedance of the desired channel, producing 27 dB of crosstalk.)

Now 27 dB isn’t as bad as first imagined. To put this into perspective, remember that even the best of the old phono cartridges had channel separation specs of about this same magnitude.

Therefore stereo separation is maintained at about the same level as a high-quality hi-fi home system of the 1970s.

For professional systems this may not be enough. If a trade-off is acceptable, things can be improved.

If you scale all the resistors up by a factor of 10, then channel separation improves from 27 dB to 46 dB.

As always though, this improvement is not free. The price is paid in reduced line driving capability.

The box now has high output impedance, which prevents driving long lines. Driving a maximum of 3000 pF capacitance is the realistic limit. This amounts to only 60 feet of 50 pF/foot cable, a reasonable figure.

So if your system can stand a limitation of driving less than 60 feet, scaling the resistors is an option for increased channel separation.

Presented with permission from Rane Corporation.

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Thursday, April 12, 2012

Audio-Technica Introduces New System 8 Wireless Systems

Audio-Technica has introduced new System 8 VHF wireless systems, available in handheld, headworn, guitar, lavalier and bodypack configurations. The single-channel, fixed-frequency systems are available in three traveling VHF frequencies (169.505, 170.245 and 171.905).

Features include an advanced dipole antenna system for extended operating range; Power, RF and AF Peak indicators; volume control, 1/4-inch output jack and user-adjustable squelch; rugged unidirectional dynamic element on handheld microphone/transmitter; variable microphone trim control and multi-color battery/power indicator (with AA operation on transmitters); and professional locking connector on the UniPak bodypack transmitter.

Audio-Technica’s seven System 8 configurations will be available April 2012 with pricing as follows:

ATW-801 basic system (includes ATW-R800 receiver and ATW-T801 UniPak transmitter)
U.S. MSRP: $149.95

ATW-801/G guitar system (includes ATW-R800 receiver and ATW-T801 UniPak transmitter with AT-GcW guitar/instrument input cable)
U.S. MSRP: $174.95

ATW-801/H headworn microphone system (includes ATW-R800 receiver and ATW-T801 UniPak transmitter with PRO 8HEcW headworn microphone)
U.S. MSRP: $249.95

ATW-801/H92 headworn microphone system (includes ATW-R800 receiver and ATW-T801 UniPak transmitter with PRO 92cW headworn microphone)
U.S. MSRP: $324.95

ATW-801/H92-TH headworn microphone system (includes ATW-R800 receiver and ATW-T801 UniPak transmitter with PRO 92cW-TH headworn microphone [beige])
U.S. MSRP: $324.95

ATW-801/L lavalier microphone system (includes ATW-R800 receiver and ATW-T801 UniPak transmitter with omnidirectional lavalier microphone)
U.S. MSRP: $224.95

ATW-802 handheld microphone system (includes ATW-R800 receiver and ATW-T802 handheld dynamic unidirectional microphone/transmitter. Includes AT8456a Quiet-Flex stand clamp)
U.S. MSRP: $174.95

Audio-Technica

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Don’t Overlook These Aspects Of The House System/Tech Package

The small things can be the very things that make for smooth running, enjoyable shows

As we left off last time, our hero was fighting for truth, justice… Oops - wrong story.

Our story here is to continue the discussion of creating a truly useful technical information package for a visiting acts and sound personnel, is a bit less glamorous.

But it need not be overly complicated, either.

Previously (here), we looked at the basic format of the tech package, and talked about the first five of 10 essential items it should offer. Now, let’s lay out the second five.

Snakes and cabling. The snake routing information sheet for the venue where I work, a theatre in a large city, looks like a parts list for a 747. I’m not sure when this system infrastructure was installed, but I can tell it took a good bit of thought and labor.

The snake line sheet in the tech package is essential for everyday operation and can be helpful to house crew and traveling crew alike. Be sure to include all the oddball lines, such as those to the spot booth, stage manager offices, and places unique to your particular venue, like the bar next door or the trap door operator position under the stage.

This sheet can do double duty by serving as a place to note malfunctioning or suspect lines as well as lines that have been commandeered as temporary replacements. Permanent lines can be noted, like those for a “stage god” mic or for emergency announcements.

Even the main sound system drive lines and data lines should be noted. This can save many a frustrating moment should something fail at an inconvenient moment (as if there is a good time for a failure of anything in a sound system).

I’ve heard it suggested that keeping crucial information like this in one’s head is a type of job insurance, the thinking being that the “keeper of secrets” will be the only one able to save the day. Let me dispel this myth here and now. If the working relationship is destined to end, the fact that the poor sap getting the job next will have a tough time is not job security.

Rather, having this information on record can never hurt, and really, can only help. It might even save that job one day, not to mention a show.

Console layout. The house console at our venue is used for every type of show imaginable. From serving as a relatively simple drive signal router for shows with nearly self-contained sound systems to having every single channel chock full of orchestra inputs for a ballet, the board does it all.

I’ve found it easy to more or less permanently assign some console input strips to a single function each, no matter what the show may be. This page in the info package not only lists the permanent channels, but also has blank spaces where an input list can be created for the empty channels.

The page should be correlated with the snake line sheet to eliminate inconsistency. Any problem channels or input strips can be noted to assure that maintenance is performed. In fact, I keep several blank copies in the folder so that input lists can be created on the spot, if needed.

Backstage paging. This seldom-praised portion of the house audio infrastructure is just as essential to many performances as the sexiest loudspeaker array in the house rig. If the actors or crew cannot hear the paging system telling them their cues and calls, chaos can ensue and further, it can rain down upon your poor head. Stage managers are not enemies to make if it can be avoided.

A list of every loudspeaker in the system, as well as its location and operating condition, serves dual purposes. Visiting engineers can quickly identify areas without coverage that may need to be supplemented and the house crew can do periodic checks of the system with a handy checklist.

Touring crew will often ask for a thorough check of program and paging in the dressing rooms, if they value their paychecks.

Actors have a habit of turning off those speakers equipped with volume pots and then throwing tantrums when a call to the stage is missed.

A list of the paging system that includes the details of the installation pathways can aid in troubleshooting.

I once had a paging system partially fail in between shows, but because only one “trunk” of this 70-volt system had failed, I was able to determine that some workmen installing a CCTV system had evidently drilled in just the wrong place in one wall.

I wish I could claim that my paging system list helped save the day. The truth is that this incident is the reason for the detailed chart I have today. Lesson learned.

System and microphone inventory. This is the most straightforward part of any tech package. A simple list of the microphones and any other audio equipment in the inventory, listed with columns for serial numbers, repair status information, and any other details deemed useful will suffice.

As with the backstage paging sheet, this page can be useful for taking inventory as well as for informational purposes. Providing it to visiting engineers along with a copy of the console input list will really speed up the process of creating an input list on those shows for which the house is providing production.

Product data. I’ve downloaded and printed product information from manufacturer web sites for some of the newer and lesser-known pieces of equipment in our sound system. The idea here is for anyone who is unfamiliar with a specific speaker or processor in the rig to be able to flip to the back of the folder and review details like power handling or frequency response of speakers.

These are the details that have no other place in the kind of “quick glance” overview found elsewhere in the package. On-line versions of a tech package could easily include a set of links to manufactures web pages for most of the items in an inventory.

Fellow house engineers out there, as well as you touring folks - I hope that this information is useful. It’s about communication, and the more we do it, the less we all goof up, and the better we all look (and most importantly, function in our jobs).

The day-to-day operation of a theatre or rock club is probably one of the least glamorous aspect of our industry, perhaps only second to last on the list, ahead of the fine art of amplifier dust screen cleaning.

But the small things a good house sound person and the crew do when no one is looking can be the very things that make for smooth running, enjoyable shows for all concerned.

Next time, the fine art of dust screen cleaning! (Just kidding.)

Mark E.P. Woods served as head of audio for a large performance theater, and now is the technical director theatre/convention center.

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Wednesday, April 11, 2012

Properly Cleaning Mixing Console Faders

Cleaning a fader is not brain surgery, but it takes practice and a lot of care. Here's how to go about it - successfully.

Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.

I’ve cleaned a lot of faders over the years and suppose I’ve gotten a little bold when it comes to tearing a fader apart and giving it a good bath.

And I’ve learned the hard way just how much punishment a fader can take before it breaks. 

In some cases, a certain amount of brute force is required to crack open a fader, but then a certain amount of gentle finesse is needed to clean its individual parts.

I recommend practicing with old junk faders – without experience, it’s all too easy to ruin a good one. 

Cleaning a fader is not brain surgery, but it takes practice and a lot of care.

Before we getting into a total fader rebuild, let’s talk about quick cleaning. Much of the time, if the gear has been well cared for and the faders are not too dirty, then a little routine maintenance is all that is likely required. 

Besides, keeping faders clean is always a good idea, preventing dirt from becoming embedded deeper inside where it can cause more wear and tear.

Keep in mind: with relatively new faders in particular, do as little as possible in order not to undo the original lubrication. And overall, don’t go any further with this process than you feel you need to.

Level 1
The first step is to use compressed air to blow as much dirt as possible out of the fader.

There is usually “dust bunnies” in the fader that will come out easily, and this might be all that needs to be achieved in terms of cleaning.

Move the fader carriage to one end and blow air into the slot aiming away from the carriage so that dust can escape through the slot. Then move the carriage to the opposite end and blow air aiming the opposite way.

Skip this step and compound the laziness by spraying some off-the-shelf cleaner-lube into the fader, and it’s likely that the dust bunnies will be matted down and stick in the corners.

A fader might seem to work better for a while, but this won’t last and might lead to the need for a more substantial (and time consuming) cleaning effort.

Note that the compressed air must be clean and dry.

I do a lot of cleaning, so I’ve invested in a $100 air compressor and then added an air filter / dryer unit for about $40. To this I’ve added a dryer cartridge that contains silica beads for about $5.

If an air compressor isn’t available, cans of aero-duster will work, but they don’t last long.

If the plan is to clean a couple/few consoles, an air compressor is a worthwhile investment, and it helps do the job right because you don’t need to be worried about running out of air.

In addition, the compressor will offer higher pressure.

Most canned air provides about 60 psi, with this dropping as the can is used.

With the compressor, I’m able to set pressure at a consistent 80 psi, which works very well. (And I found out the hard way that 100 psi will blow some faders and switches apart!)

Level 2
If the initial “blowing out” process didn’t offer the desired results, it’s time to move on to use of chemical contact cleaner.

Some faders have lubricating grease applied by the manufacturer, while others employ a self-lubricating Teflon-type of plastic.

If used sparingly, chemical contact cleaner shouldn’t impact the self-lubricating type, but it will invariably wash away lubricating grease.

The goal is to avoid adding any more lubrication than is absolutely necessary - dust tends to fall away from dry surfaces, but it sticks to oily surfaces.

After spraying contact cleaner, exercise the fader and then quickly blow out the excess cleaner.

This helps to spread the cleaner over the entire fader surface, while the excess cleaner carries away additional loosened dirt.

I’ve tried several types of contact cleaner since canned Freon was banned from the market.

There are a lot of good choices – my preference is Contact Cleaner II made by Techspray. It’s about $30 per can and worth the price. Note that I also invested another $30 for a screw-on trigger nozzle so that I can be precise and cut waste.

Level 3
The fader is still feeling a little rough? Time to try a little lubrication. The key word is “little” – use as little as possible.

Did I mention not to use too much lubrication? Third time’s the charm – lubrication collects dust, so don’t overdo it! 

Depending on the type of fader, I use a precision dropper to place just a few drops of lubrication in the fader, or give it just a quick squirt.

Exercise the fader and then blow away the excess with compressed air. Again with the compressed air?

Seriously, this helps spread the lubrication into a thin film and gets rid of any excess.

I’ve had good results with a spray lubrication called Tefrawn, made by Rawn. It’s Teflon-based and beneficial to the self-lubricating type plastics noted earlier.

Also, it smells like bananas, not that it matters!) Caig also offers products of this type. 

Lesson learned the hard way: some oils react with plastic, causing it to break down. If there’s any doubt, test it out on a spare fader first before applying.

Also, certain faders use thicker grease that results in a “smoother” feel, and these may actually feel too loose after lubrication.

If this proves bothersome, use silicon or petroleum grease (but not bacon grease!). I’ve found this step to be more trouble than it’s worth - if “feel” is that important, buy new faders.

Time to reiterate:  “Air > Cleaner > Air > Lubrication > Air” About 30 seconds of effort for each fader.

Total Rebuild
Some of the more expensive faders are designed to be easily taken apart for cleaning.

If less expensive faders can’t be cleaned using the steps already outlined, it may not be cost-effective to go any further.

Consider replacement, but if it’s an emergency, keep in mind that you’ll be dealing with tiny parts that are easy to break and lose.

A total fader rebuild should take only about 5 to 10 minutes, after going to the trouble of taking apart the console to get to the fader.

If I take a module out for repair, I go ahead and clean its fader at the same time. Otherwise, I do fader rebuilding as part of a larger console-cleaning project.

There are several different types of fader construction. Higher-cost faders are literally a “snap” to take apart; that is, they have a “snap together” design.

The main parts of a typical fader include the element that carries audio on conductive tracks, the carriage that holds wipers against the tracks, and the rails that guide the carriage.

Be extremely careful with the wipers - they’re easy to damage, and once bent, the fader is toast.

After opening the fader, first blow away the loose dust. There might be dirt wedged in at the point where the carriage and rails meet, so use a dental pick to loosen this up, and then blow it out.

Use a strip of clean cloth dipped in isopropyl alcohol to clean the rails, pass the strip under and around each rail. 

Gently wipe the surface of the conductive element with a clean cloth dipped in alcohol or contact cleaner. Be gentle, and do NOT go under the carriage with the cloth. This can damage the wipers!

Apply just a few drops of lubrication to the rails and exercise the fader. Blow away any excess lubrication with and reassemble the fader

And that’s it. With a little practice and patience, anyone can make old faders feel like new again!

Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.

Alex Welti is vice president of research for Creation Audio Labs, a service facility in the southeastern U.S. He served for a decade as service manager of Soundcraft, and prior to that, worked as a technical supervisor for Westlake Audio.

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Church Sound: Preventing Hum And Radio Frequency Interference (RFI) In A System

Approaches to keeping audio clean

 
You patch in a piece of audio equipment, and there it is: HUM! This annoying sound is a common occurrence in sound systems.

Hum is an unwanted 60 Hz tone—50 Hz in Europe—maybe with harmonics. If the harmonics are especially strong, the hum becomes an edgy buzz.

A sound system also might be plagued by RFI (Radio Frequency Interference). It’s heard as buzzing, clicks, radio programs, or “hash” in the audio signal. RFI is caused by CB transmitters, computers, lightning, radar, radio and TV transmitters, industrial machines, cell phones, auto ignitions, stage lighting, and other sources.

This article looks at some causes and cures of hum and RFI. By following these suggestions, you can keep your audio clean.

HUM AND CABLES

One cause of hum is audio cables picking up magnetic and electrostatic hum fields radiated by power wiring in the walls of a room. Magnetic hum fields can couple by magnetic induction to audio cables, and electrostatic hum fields can couple capacitively to audio cables. Magnetic hum fields are directional and electrostatic hum fields are not.

Most audio cables are made of one or two insulated conductors (wires) surrounded by a fine-wire mesh shield that reduces electrostatically induced hum. The shield drains induced hum signals to ground when the cable is plugged in. Outside the shield is a plastic or rubber insulating jacket.

Cables are either balanced or unbalanced. A balanced line is a cable that uses two conductors to carry the signal, surrounded by a shield (Figure 1).

On each end of the cable is an XLR (3-pin pro audio) connector or TRS (tip-ring-sleeve) phone plug.

Each conductor has equal impedance to ground, and they are twisted together so they occupy about the same position in space on the average. Hum fields from power wiring radiate into each conductor equally, generating equal hum signals on the two conductors (more so if they are a twisted pair).

Those two hum signals cancel out at the input of your mixer, because it senses the difference in voltage between those two conductors—which is zero volts if the two hum signals are equal. That’s why balanced cables tend to pick up little or no hum.

An unbalanced line has a single conductor surrounded by a shield (Figure 2).

At each end of the cable is a phone plug or RCA (phono) plug. The central conductor and the shield both carry the signal. They are at different impedances to ground, so they pick up different amounts of hum from nearby power wiring.

There’s a relatively big hum signal between hot and ground that results in more hum than you get with a balanced line of the same length.

Some hanging mics have long unbalanced cables, and some cables used between pieces of equipment are unbalanced. An unbalanced line less than 10 feet long usually does not pick up enough hum to be a problem.

Wherever you can, use balanced cables going into balanced equipment. Keep unbalanced cables as short as possible (but long enough so that you can service them). Check inside cable connectors to make sure that the shield and conductors are soldered to the connector terminals.

Route mic cables and patch cords away from power cords; separate them vertically where they cross. This prevents the power cords from inducing hum into the mic cables.

Also keep audio equipment and cables away from computer monitors, power amplifiers, lighting dimmers and power transformers. Keep mic cables and mic electronics well separated from lighting equipment in the grids.

Better yet, don’t use hanging mics. A few stage-floor mics and wireless mics will suffice.

GROUND LOOPS

Another major cause of hum is a ground loop: a circuit made of ground wires. It can occur when two pieces of equipment are connected to the building’s safety ground through their power cords, and also are connected to each other through a cable shield (Figure 3).

The ground voltage may be slightly different at each piece of equipment, so a 50- or 60-Hz hum signal flows between the components along the cable shield. It becomes audible as hum. Also, the cable shield/safety ground loops acts like a big antenna, picking up radiated hum fields from power wiring.

For example, suppose your mixer’s power cord is plugged into a nearby AC outlet. The system power amplifiers are plugged into outlets on stage. So the mixer and amps are probably fed by two different circuit breakers at two different ground voltages. When you connect an audio cable between the mixer and power amps, you create a ground loop and hear hum.

To prevent ground loops, plug all audio equipment into outlet strips powered by the same breaker. (Make sure the breaker can handle the current requirements).

Run a thick AC extension cord from the stage outlets to the mixer, and plug the mixer’s power cord into that extension cord.

That way, the separated equipment chassis will tend to be at the same ground voltage—there will be very little voltage difference between chassis to generate a hum signal in the shield.

Caution: Some people try to prevent ground loops by putting a 3-to-2 safety ground lifter on the AC power cords. NEVER DO THAT. It creates a serious safety hazard.

If the chassis of a component becomes accidentally shorted to a hot conductor in its power cord, and someone touches that chassis, the AC current will flow through that person rather than to the safety ground.

Lift the shield in the receiving end of the signal cable instead, and plug all equipment into 3-pin grounded AC outlets.

Let’s explain the signal ground lift in more detail. The hum current in a ground loop flows in the audio cable shield, and can induce a hum signal in the signal conductors. You can cut the audio cable shield at one end to stop the flow of hum current.

The shield is still grounded at the other end of the cable, and the signal still flows through the two audio leads inside the cable.

So, to break up a ground loop, disconnect the cable shield from pin 1 in line-level balanced cables at the male XLR end (Figure 4).

You can either cut the shield, or plug in an inline audio cable ground-lift adapter.

Removing the shield connection at one end of the audio cable makes the connection sensitive to radio-frequency interference (RFI).

So solder a 100 pF capacitor between the shield and XLR pin 1 (Figure 5).

This effectively shorts RFI to ground, but is an open circuit for hum frequencies.

Some engineers create a partial ground lift by placing a 100 ohm resistor between the cable shield and male XLR pin 1 (Figure 6).

This limits the current passing through the cable shield but still provides a good ground connection.

Label the XLR connector “GND LIFT” so you don’t use the cable where it’s not needed. For example, mic cables must have the shield tied to pin 1 on both ends of the cable. The ground lift is only for line-level cables.

Here’s another way to prevent a ground loop when connecting two balanced or unbalanced devices. Connect between them a 1:1 isolation transformer or hum eliminator, such as a Jensen Iso-Max CI-2RR or Ebtech Hum Eliminator.

OTHER TIPS

Even if your system is wired properly, hum or RFI may appear when you make a connection. Follow these tips to stop the problem:

• Unplug all equipment from each other. Start by listening just to the powered PA loudspeakers. Connect a component to the system one at a time, and see when the hum starts.

• Remove audio cables from your devices and listen to each device by itself. It may be defective.

• Partly turn down the volume on your power amp, and feed it a higher-level signal from your mixer (0 VU maximum).

• If you are using a mic snake, be sure that its stage box is not touching metal.

• Do not wire XLR pin 1 to the connector-shell lug because the shell can cause a ground loop if it touches grounded metal. If you are sure that the shell won’t touch metal, wire XLR pin 1 to the shell lug to prevent RFI.

• Try another mic. Some dynamic mics have hum-bucking windings.

• If you hear hum or buzz from an electric guitar, have the player move to a different location or aim in a different direction. Magnetic hum fields are directional, and moving or rotating the guitar pickup can reduce the coupling to those fields.

• If the hum is coming from a direct box, flip its ground-lift switch.

• Turn down the high-frequency EQ on a buzzing bass guitar signal.

• If you think that a speaker cable, mic cable or patch cord is picking up RFI, wrap the cable several times around an RFI choke (available at Radio Shack or other electronics supply houses). Put the choke near the device that is receiving audio.

• Install high-quality RFI filters in the AC power outlets. The cheap types available from local electronics shops are generally ineffective.

• Connect cable shields directly to the equipment chassis instead of to XLR pin 1, or in addition to pin 1. Some equipment is designed this way to prevent the “pin 1 problem”. The cable shield should be grounded directly to the chassis—not connected instead to a ground terminal on a circuit board inside the chassis.

• Periodically clean connector contacts with Caig Labs DeoxIT, or at least unplug and plug them in several times.

By following all these tips, you can greatly reduce the likelihood of hum and RFI in your audio system. Good luck!

Bruce Bartlett is a microphone engineer (www.bartlettmics.com), recording engineer, live sound engineer, and audio journalist.

 

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