Article

Monday, February 06, 2012

RE/P Files: A Quadraphonic Microphone Development

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature provides an interesting look back at quadraphonic recording. This article dates back to September of 1970. (Volume 1, Number 3). The text is presented unaltered, along with all original graphics.

As a complete oversimplification, a microphone is an instrument which measures differences in air pressure.

It is not surprising that somebody would, in light of the interest in Quadraphonic sound, experiment and perfect an instrument which would measure and transduce the differences in air pressure around a full 360 degrees - to effectively create a quadraphonic microphone.

Such a truly Quadraphonic device, developed by engineer Carl Countryman and producer Brad Miller, is in external appearance no different than the several models of standard microphones (Figure 1).

This Quadraphonic microphone has been designed and built using the case and chassis of a Neumann SM-2, into which four independent microphone heads have been built to provide full 360-degree pick-up.

The pick-up patterns (Figure 2) are cardioid, front and back, and figure-8 at the sides.

Although the obviously complicated matrixing data are proprietary, and unavailable for publication, the discussion of pickup patterns, generally, yields an understanding of how the design provides excellent separation and naturality of sounds.

Cardioid, also sometimes called unidirectional, is a heart-shaped response. It is resultant of an omnidirectional and figure-8 pickup.

The signals are superimposed on each other; at the very rear they are anti-phase, and so cancel out.

At the front they are in phase, hence the tapering hear-shaped response toward the rear.

Figure-8, or bi-directional pickup-patterns, are the result of two directional pickup patterns, one in phase and the other anti-phase.

The output at the front and the back are equal, although opposite,.

As the input signal moves to the side, the output is gradually reduced until at 90 degrees, the two patterns have, for all intent and purpose, canceled each other out.

Figure 3 shows microphone capsules as they are arranged in the microphone head.

“Front to Back” and “Left to Right” are one above the other at 90 degrees to each other.

Three demonstrations, on very spontaneous, served to convince that development of the unit is very nearly complete.

The microphone was hung in Miller’s back yard garden, surrounded by about 200 degrees of sound source emanating from a waterfall with various small tributary streams flowing from it. It presented an excellent opportunity to “hear” the complete environment; the waterfall in stereo on the two speakers in “front,” and from behind, the beautiful ambiance of the total environment and the reflected sound.

Several minutes into the demonstration, on the Southern Pacific tracks bordering on the rear of the Miller garden, a slow-moving freight train ambled by. The completeness of the sound, the way it engulfed the listening room, is difficult to describe. It was totally complete… almost frighteningly so.

Miller completed the demonstration by playing a 4-track tape of his “Mystic Moods Orchestra” on an especially adapted Sony. The machine (Figure 4) has been adapted for 4-track, in and out, and will be able to accommodate 10-inch reels of 2-inch tape.

The machine is the forerunner of a new design from the Countryman/Miller collaboration which will weigh in the vicinity of 20 pounds.

The “Mystic Moods” piece only served to further impress that Quad or Multi is certainly on the way… with an endless spectrum of sound combinations and tonal effects.

Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

 

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Sunday, January 08, 2012

Real World Gear: A Look At 2-Way “Conventional” Loudspeakers

The ubiquitous compact 2-way box gets the job done

When it comes to loudspeakers for professional audio applications, line arrays get the glory and much of the press, but traditional 2-way boxes are still the real workhorses of the business.

They remain invaluable for a range of very good reasons, with versatility that translates to “great bang for the buck” topping the list.

A quick check of manufacturer websites bear this out as well. Again, the latest line array might be the “next big thing” touted on the home page splash, but dig around and you find dozens of 2-way models.

Available in a wide range of sizes and cabinet shapes, they serve as mains, monitors, side fills, center fills, near fills, front fills and delays, and can quickly be ground stacked, flown, or placed on a stand. They can be used by themselves or with subwoofers in applications requiring additional low-end reinforcement.

Many (most) of these boxes are trapezoidal in shape, with some able to be put into tightly packed horizontal arrays and others better suited for “exploded” clusters. Still others can be stacked or flown into line and/or column arrays. Several offer a modified cabinet shape that also allow them to be placed on their side and used as floor monitors.

Smaller 2-way models incorporate a single 8- or 10-inch woofers, but the most popular models offer 12- or 15-inch woofers for additional low end performance. Usually the woofers are accompanied by a compression driver on a horn or waveguide for mid and high frequencies, although ribbon drivers have also emerged as a viable option from certain companies.

Let’s also not overlook coaxial models where the individual driver units radiate sound from the same point/axis, which, when designed properly, can offer enhanced coherence.

When evaluating conventional loudspeakers, start by defining the right box for the job – size, scale, mounting, portability, and so on. It all depends on the requirements of the application(s). Our tour of recent models that follows is intended, by design, to present the “state of the market” in terms of options.

But for each type of model presented here, understand that there are several similar models from other sources, so further homework is strongly recommended. When making “apples to apples” comparisons, here are basic factors to consider:

—Dispersion, a measurement of the pattern of MF/HF sound that emanates from the box. This is stated in degrees for the horizontal and vertical planes.

—Power Handling, which, for passive cabinets, is usually stated as an “RMS” or “continuous” rating in watts. An increasing number of these loudspeakers are now self-powered and also have onboard DSP.

—Sensitivity, stated in decibels, is a measurement of the sound level the loudspeaker can produce with a given input signal, generally measured with 1 watt input at 1 meter distance. (By the way, we’re seeing an increasing number of manufacturers who prefer to provide a Maximum SPL specification.)

—Mounting, which includes integrated flypoints as well as things like pole cups that can come in quite handy for true portable applications.

Whether placed on a stand for a speech at a groundbreaking ceremony, stacked on top of subwoofers at the local music venue, or hung from the ballroom ceiling at a corporate event, the ubiquitous conventional 2-way box gets the job done.

Go here to take our Real World Gear of the latest 2-way loudspeaker models.

Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb, and is the owner of Tech Works, a production company based in Las Vegas.

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Friday, November 18, 2011

Knowing Cone Drivers: How They Work, Understanding Key Data & Specs

What's really going on with woofers, and what are the important factors in how they perform as well as how they impact the performance of loudspeaker systems

(Editor’s Note: Eminence Speaker LLC contributed to this report.)

Cone drivers (also referred to as woofers and transducers in this article) are not overly complex. When an electrical current passes through a wire coil (the voice coil) in a magnetic field, it produces a force that varies with the current applied.

The cone, connected to the voice coil, moves in and out, creating waves of high and low air pressure. The coil and magnet assembly are the “motor structure” of the loudspeaker.

The movement is controlled by the loudspeaker’s suspension, which comprises the cone surround and the “spider”.

The surround and spider allow the coil to move freely along the axis of the magnet’s core (or “pole”) without touching the sides of the magnetic gap.

More important than knowing the details of how cone drivers work is the understanding of key data and what it means. Prior to 1970, there were no easy or affordable methods accepted as standard in the industry for obtaining comparative data about loudspeaker performance.

Recognized laboratory tests were expensive and unrealistic for the thousands of individuals needing performance information.

Standard measurement criteria were required to enable manufacturers to publish consistent data for customers to make comparisons between various loudspeakers.

Things began changing in the early 1970s, however, when several technical papers were presented to the Audio Engineering Society (AES) that resulted in the development of what we know today as Thiele-Small Parameters.

The authors of the papers – A.N. Thiele and Richard H. Small – devoted considerable effort to showing how the following parameters define the relationship between a cone driver and a particular enclosure.

These parameters can be invaluable in making choices because they can tell you far more about the transducer’s real performance than the basic benchmarks of size, maximum power rating or average sensitivity.

Let’s have a look at the parameters defined by Mr. Small and Mr. Thiele. (And note that we listed Mr. Small first this time – bet he doesn’t get that very often!)

Fs: The free-air resonant frequency of a cone driver. Simply stated, it’s the point at which the weight of the moving parts of the speaker becomes balanced with the force of the driver suspension when in motion.

If you’ve ever seen a piece of string start humming uncontrollably in the wind, you have seen the effect of reaching a resonant frequency. It’s important to know this information so that you can prevent your enclosure from ‘ringing’.

With a cone driver, the mass of the moving parts, and the stiffness of the suspension (surround and spider), are the key elements that affect the resonant frequency.

As a general rule of thumb, a lower Fs indicates a woofer that would be better for low-frequency reproduction than a woofer with a higher Fs. This is not always the case though, because other parameters affect the ultimate performance as well.

Re: DC resistance of the driver measured with an ohm meter, and often referred to as the “DCR.” This measurement will almost always be less than the driver’s nominal impedance.

Some users sometimes get concerned the Re is less than the published impedance and fear that amplifiers will be overloaded. Due to the fact that the inductance of a speaker rises with a rise in frequency, it is unlikely that the amplifier will often see the DC resistance as its load.

Le: Voice coil inductance measured in millihenries (mH). The industry standard is to measure inductance at 1 kHz. As frequencies get higher, there will be a rise in impedance above Re, because the voice coil is acting as an inductor.

Consequently, the impedance of a cone driver is not a fixed resistance, but can be represented as a curve that changes as the input frequency changes. Maximum impedance (Zmax) occurs at Fs.

Q Parameters: Qms, Qes, and Qts are measurements related to the control of a transducer’s suspension when it reaches the resonant frequency (Fs). The suspension must prevent any lateral motion that might allow the voice coil and pole to touch (this would destroy the driver). The suspension must also act like a shock absorber.

Qms is a measurement of the control coming from the driver’s mechanical suspension system (the surround and spider). View these components like springs.

Qes is a measurement of the control coming from the driver’s electrical suspension system (the voice coil and magnet). Opposing forces from the mechanical and electrical suspensions act to absorb shock.

Qts is called the “Total Q” of the driver and is derived from an equation where Qes is multiplied by Qms and the result is divided by the sum of the same.

As a general guideline, Qts of 0.4 or below indicates a transducer well suited to a vented enclosure. Qts between 0.4 and 0.7 indicates suitability for a sealed enclosure, and Qts of 0.7 or above indicates suitability for free-air or infinite baffle applications.

Vas/Cms: Vas represents the volume of air that when compressed to one cubic meter exerts the same force as the compliance (Cms) of the suspension in a particular speaker.

Vas is one of the trickiest parameters to measure because air pressure changes relative to humidity and temperature – a precisely controlled lab environment is essential.

Cms is measured in meters per Newton, and is the force exerted by the mechanical suspension of the speaker. It is simply a measurement of its stiffness.

Considering stiffness (Cms), in conjunction with the Q parameters, gives rise to the kind of subjective decisions made by car manufacturers when tuning cars between comfort to carry a family and precision to go racing.

Think of the peaks and valleys of audio signals like a road surface, then consider that the ideal driver suspension is like car suspension that can traverse the rockiest terrain with race-car precision and sensitivity at the speed of a jet plane.

Vd: Peak Diaphragm Displacement Volume – in other words, the volume of air the cone will move. It is calculated by multiplying Xmax (voice coil overhang of the driver) by Sd (Surface area of the cone). Vd is noted in cc, and the highest Vd figure is desirable for a sub-bass transducer.

BL: Expressed in Tesla meters, this is a measurement of the motor strength of a driver. Think of this in terms of how good a “weightlifter” the transducer can be. A measured mass is applied to the cone, forcing it back, while the current required for the motor to force the mass back is measured.

The formula is mass in grams divided by the current in amperes. A high BL figure indicates a very strong transducer that moves the cone with authority.

Mms: The combination of the weight of the cone assembly plus the “driver radiation mass load.” The weight of the cone assembly is easy: it’s just the sum of the weight of the cone assembly components.

The driver radiation mass load is the confusing part. In simple terminology, it is the weight of the air (the amount calculated in Vd) that the cone will have to push.

Rms: Represents the mechanical resistance of a driver’s suspension losses. It is a measurement of the absorption qualities of the driver suspension and is stated in N*sec/m.

EBP: Calculated by dividing Fs by Qes. The EBP figure is used in many enclosure design formulas to determine if a driver is more suitable for a closed or vented design.

An EBP close to 100 usually indicates a driver that is best suited for a vented enclosure. On the contrary, an EBP closer to 50 usually indicates a speaker best suited for a closed box design.

This is merely a starting point. Many well-designed loudspeaker systems have violated this rule of thumb! Qts should also be considered.

Xmax/Xmech: Short for “maximum linear excursion.” Driver output becomes non-linear when the voice coil begins to leave the magnetic gap.

Although suspensions can create non-linearity in output, the point at which the number of turns in the gap (see BL) begins to decrease is when distortion starts to increase.

Xmax is voice coil height minus top plate thickness, divided by two, while Xmech (as expressed by Eminence) is the lowest of four potential failure condition measurements times two: Spider crashing on top plate, and/or voice coil bottoming on back plate. Voice coil coming out of gap above core; physical limitation of cone.

Take the lowest of these measurements and then multiply it by two. This gives a distance that describes the maximum mechanical movement of the cone. (For Eminence transducers, half the Xmech value represents the one-way excursion limit that if exceeded would cause permanent damage.)

Sd: This is the actual surface area of the cone, normally given in square centimeters.

Zmax: Represents the driver’s impedance at resonance.

Usable frequency range: Manufacturers use different techniques for determining this, and most are recognized as acceptable in the industry. However, they can arrive at
different results.

Technically, many drivers are used to produce frequencies in ranges where they would theoretically be of little use. As frequencies increase, the off-axis coverage of a transducer decreases relative to its diameter.

At a certain point, the coverage becomes ‘beamy’ or narrow like the beam of a flashlight.

See the chart at left – it demonstrates at what frequency this phenomenon occurs relative to the size of the transducer. If you’ve ever stood in front of a guitar amplifier or loudspeaker cabinet, then moved slightly to one side or the other and noticed a different sound, you have experienced this phenomenon.

Clearly, most two-way loudspeaker systems ignore the theory and still perform quite well.

Power handling: A transducer needs to be capable of handling the input power it’s provided. The general rule of thumb is that a power amplifier, when reproducing any program source, “provides” long term- thermal power that is approximately 1/8 its maximum rated output before clipping (rap music excluded).

This is why even UL testing for power amps is done, and listed for on the back of the amp, at 1/8 the rated output power of the amp.

Typically, a loudspeaker will handle somewhere between 6 dB to 10 dB higher peaks than its long-term-average power rating, particularly in the case of the conservative EIA-426A standard used by several manufacturers.

This means that if a loudspeaker is rated for 100 watts long-term-average power, the amp driving it should be rated between 400 and 1000 watts – if the user does not compress the source signal. Once compression is used, all bets are off.

Generally speaking, the number one contributor to a transducer’s power rating is its ability to release thermal energy. This is affected by several design choices, but most notably voice coil size, magnet size, venting, and the adhesives used in voice coil construction.

Larger coil and magnet sizes provide more area for heat to dissipate, while venting allows thermal energy to escape and cooler air to enter the motor structure. Equally important is the ability of the voice coil to handle thermal energy.

Mechanical factors must also be considered when determining power handling. A transducer might be able to handle 1000 watts from a thermal perspective, but would fail long before that level was reached from a mechanical issue such as the coil hitting the back plate, the coil coming out of the gap, the cone buckling from too much outward movement, or the
spider bottoming on the top plate.

The most common cause of such a failure would be asking the speaker to produce more low frequencies than it could mechanically produce at the rated power. Be sure to consider the suggested usable frequency range and the Xmech parameter in conjunction with the power rating to avoid such failures.

Sensitivity: One of the most useful specifications published for any transducer, it’s a representation of the efficiency and volume you can expect from a device relative to the input power.

Manufacturers follow different rules when obtaining this information – there is not an exact standard accepted by the industry. As a result, it is often the case that loudspeaker users are unable to accurately compare the sensitivities of different products.

Eminence Speaker LLC and Live Sound/ProSoundWeb Senior Technical Editor John Murray contributed this article.

Also be sure to read Real World Gear: The Latest In Loudspeaker Drivers and take our Photo Gallery Tour of the latest driver models.

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Thursday, November 17, 2011

In Profile: Karl Jackson Chief Audio Technician, United States Marine Band

Working to be transparent to the musical product

When Karl Jackson applied for the position of chief audio technician for “The President’s Own” United States Marine Band (USMB) fresh out of DePaul University in 1995, he really didn’t think he’d get the job.

Founded in 1798 by an Act of Congress, the USMB is the oldest continuously active professional musical organization in the nation, with a stated purpose of providing music for the president of the United States and the commandant of the Marine Corps.

“I thought I didn’t have a hope,” Jackson says, “but they interviewed me and within a few months I enlisted in the Marine Corps, moved to Washington, D.C., and went to work.”

Strictly speaking, it was a little more complicated than it sounds. “When a person joins ‘The President’s Own’ they have to secure a secret security clearance for access to the White House,” he notes. It’s a process involving multiple interviews with defense intelligence and, understandably, a thorough background check. “They visited every place I’d lived and talked to people I’d known.”

There’s a lot more to Jackson’s job than working high-profile government functions. The USMB plays approximately 500 shows annually and Jackson’s roles range from mixing and recording those shows, to engineering sessions featuring the USMB and some of the world’s most accomplished musicians at the Corps’ dedicated rehearsal/recording facility, to preserving and archiving USMB recordings (some of which date back to 1889).

Career Path
Although initially surprised he got the position, in many ways, Jackson’s background makes him an ideal fit. He remembers first becoming interested in technology as a child growing up in Buena Vista, CO.

“My father was an electrician, so I had a soldering iron in my hand from a very early age,” he says, laughing. “I remember helping him wire up an organ for a friend of ours when I was really young.” He also had a parallel interest in music and played trumpet throughout high school, then in college, and, for a time, semi-professionally in the Washington area.

It was actually one of Jackson’s teachers at DePaul who suggested he apply for the USMB. But it wasn’t the first time a teacher had an influence on his career path. Jackson says, recalling a school music teacher, Harold Creswell, who was crucial to his early development as a musician. “He was one of the first people to take me aside and say, ‘you know, you could be really good if you applied yourself ’.”

The message was simple, but it stuck and gave Jackson something to focus on at a time when he was just beginning to consider what direction to take long term.

Like Jackson, many of his six siblings are technically minded. “I have four brothers who are engineers, and a sister who’s a math teacher.” There’s also a family connection to the military, he adds.

His father served as a submariner after the Korean War, and two of his brothers currently serve in the Air Force and Navy. In college, however, a career in the military wasn’t on his radar. Rather, Jackson’s own instinct to serve others manifested itself in his volunteer work for organizations like Habitat for Humanity and community literacy initiatives. He also worked for a time as an intern in various Chicago area recording studios.

As it turned out, however, his degree, with its dual emphasis on the sciences – physics, electronics and mathematics – and music theory and performance, made him an ideal fit for the Marines. “Being able to sit down with a complex score, discuss it with the conductors and musicians and edit based on the score is absolutely essential in this job. You have to have strong score reading and musical chops.”

Customs & Courtesies
Now, given Jackson’s security clearance, and the Marines’ reputation as one of the world’s most proficient combat forces, you have to wonder if the job requires any additional skills – such as the ability to take out a heckler armed only with a piece of dental floss, for example?

Jackson chuckles at the suggestion, explaining he was brought into the corps through a process called “lateral entry,” which does not require combat training.

He did have to undergo instruction for the position, but primarily in the customs, courtesies and history of the Marine Corps.

Still, that begs the question: What effects do those customs and courtesies have on his job, and just how “military” are the day-to-day workings of the USMB?

“We’re at the intersection of – probably – the best concert band and the best professional military organization in the world, so both have pervasive impacts,” he says. “We’re part of the hierarchical structure you’ll find in every military, but at the same time, we have to maintain the intimate feeling, the family structure, of a musical organization where we have to treat people informally at times.”

Jackson does refer to the mandate of the USMB as a mission, however, particularly when describing the steep learning curve he faced early on in the job. “One of the challenges was walking into a venue like Boston Symphony Hall with recording equipment and a bag of microphones and trying to figure out how to position mics optimally. It’s tough to hang mics in that room. There aren’t exactly a lot of convenient holes in the ceiling. You have to drop lines in carefully and use fish line to get them exactly where they need to go.”

Over time, he developed his own system, he says. “Through experimentation we came to the conclusion that a near coincident technique, supplemented by some flanking mics and judicious use of spot mics, gave us the best chance of not only capturing ensemble balances, but the stereo field and the location of an instrument on the stage.”

Working at some of the most iconic venues in the nation with the USMB has awoken in Jackson a keen interest in acoustics, an interest that has prompted him to contemplate how best to combine the various skills and interests he’s developed over the past 16 years.

And as he becomes eligible to retire from the USMB in a few years, he is considering pursuing a Masters degree in architectural acoustics, with an eye to transitioning into a career in performing arts facility and acoustical design at some point in the future.

Fulfilling A Mandate
When it comes to live sound, Jackson doesn’t walk into a series of similar venues, with the same gear and the same band night after night. Shows range from full concert band to smaller orchestral, jazz and contemporary country music ensembles, at venues running the gamut from school gymnasiums to Carnegie Hall to the Lincoln Memorial.

Touring is only about 5 percent of the gig and many performances requiring larger systems are outdoor summer shows and, with a few exceptions – like a series of concerts the USMB was invited to play in Switzerland at the invitation of The World Association of Symphonic Bands and Ensembles in 2001 – mostly in the continental U.S.

Put mildly, some shows and some of the musical selections performed are more complex than others, particularly when it comes to fulfilling the USMB’s mandate to introduce new music to audiences. Among them, for example, a symphony including some decidedly non-traditional components: “Shotguns going off in the concert hall and marching bands in the middle of the audience, things like that,” Jackson says.

In every case, whatever the case, when recording and mixing the band, Jackson’s approach is to stay out of the way of the music.

“There’s a lot of sound coming off the stage. My goal is to subtly reinforce that without washing it out. That’s one of our most important values,” he adds, “to be low visibility and just as transparent to the musical product as we can be. If nobody knows we’re there, we’re doing our job right.”

For events like the January 2009 inauguration of Barack Obama as the 44th U.S. president, Jackson doesn’t mix the show. His focus during inaugurations specifically is on being an intermediary between those calling the show and the front of house position and, perhaps more importantly, managing playback when necessary. That’s an absolute necessity at events the entire world is watching, where nothing can go off the rails.

“One of my most important roles is to ensure that even if every instrument in the band is frozen solid, a block of ice, that there’s still ceremonial music.”

Backups for inaugural celebrations and other large-scale events are usually recorded in the USMB rehearsal facility, and they have resulted in some standout moments for Jackson. Among them, a session featuring Yo Yo Ma, Itzhak Perlman, Anthony McGill and Gabriella Montero recording a quartet piece by composer John Williams for the inauguration.

“Hosting that session in my facility was one of those ‘is this real’ moments,” he says, as has been meeting - if only briefly - each sitting president since the mid-1990s.

Preserving A Link
He’s grateful the USMB took a chance on him early on, and is also proud of his role in preserving the musical heritage of “The President’s Own” and in contributing to it with the recordings he’s engineered. But he seems to take the most pride in being part of an organization that – for those who have served in the Corps and the armed forces in general over the course of time – is integral
in preserving a link to their past.

“That may be a tie for first place when it comes to the best part of the job – the audience reaction,” he says. “I can’t count the number of times that people have talked to me about their experiences,
or their parent’s experiences in the military, and how meaningful it is to them to feel proud about that, or to regain that pride in their service.

“Frequently, we’ll be doing a concert in a high school and the place will be packed, 2,500 people, all sitting on bleachers. We’ll play ‘The Marine’s Hymn,’ for example, and I’ll look over and see a Korean War era veteran struggle to get up from his seat and stand at attention. That is the best part of putting on a Marine Band concert; talking with folks and gaining a deeper appreciation for what they’ve contributed to our military and our country.”

Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer.

 

 

 

 

 

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Monday, November 14, 2011

Real World Gear: Microphones For Live Instrument Applications

So where do we start in this pursuit of the “best” in all aspects of instrument mic’ing?

Long ago, an old sound guy imparted to me words of wisdom about the optimum way to mic instruments: “Pick your best microphone, put it on your best stand, place it in the best spot, point it in the best direction, and hope for the best.”

That statement seems a bit trite, but it also contains truth. So where do we start in this pursuit of the “best” in all aspects of instrument mic’ing?

While it’s true that vocal mics can do double duty on instruments, few do both jobs well. Simply put, vocal mics are optimized for voice – many have a presence peak, or a boost in a band of upper midrange frequencies. This enhances the articulation and intelligibility of voices, but when applied to instruments, it can add coloration that’s detrimental.

Vocal mics also tend to roll off lower end frequencies (below the voice range) to help eliminate handling noise and stage rumble, which is not a great characteristic if you want to use the mic on instruments that produce low frequencies. General-purpose instrument mics with a flatter and wider frequency response supply a more natural and realistic sound of the instrument to the PA.

A step above the general instrument mic are those optimized for particular types of instruments. They have tailored responses and features like custom mounting clips and/or adapters to meet the needs of specific applications. The downside is that they might be so specialized that they’re ill-suited for most other uses.

Rugged and reliable, dynamic models are still the most common. They work on the principle of electromagnetic induction, similar to a loudspeaker but in reverse. Dynamics used to come in just two flavors: medium and large diaphragm, the latter intended for instruments that produce bass frequencies.

But today, a host of small-diaphragm dynamics are available to be tucked into tight spaces, nestled between drums, and clipped on to instruments.

Condenser designs have gained in popularity, with lighter diaphragms that can respond more quickly, making them very well-suited in reproducing a wide range of frequencies, particularly those at the higher end of the spectrum. The downside is that the diaphragm is not connected to a coil mass and thus require a power source (phantom power or batteries).

With both dynamics and consdensers, the size of the diaphragm is usually a good indicator of intended use. Larger ones are usually geared toward instruments that produce lower frequencies, but that too is changing, with some smaller diaphragm designs now also excelling in the lower registers.

The vast majority of instrument (and vocal) microphones are unidirectional, with pickup patterns including cardioid, supercardioid and hypercardioid. In general, they reject sounds from directions other than the front (at various degrees), and this is highly desirable given all of the noise on a typical stage as well as from the house.

Omnidirectional designs “hear” equally well in all directions, which can be problematic in noisy environments but can be effective in providing added ambience and dimension with more subtle productions, i.e., acoustic, largely unamplified. The sensitivity rating tells us the loudness of the mic’s output. The lower the negative dB number, the louder the output. For very soft instruments, choosing a mic that has a louder output may keep you from having to crank up the preamp and raising the noise floor on that channel.

The old sound guy said to pick the best, and in this Real World Gear look at instrument mics, we present a host of viable options fitting that criteria. Of course, the final choice is up to you.

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Monday, October 24, 2011

Extreme Makeover: A Look At Modern Power Amplifiers

It’s funny the difference that innovation can make...

The common “wisdom” just a few years ago was that the emergence of self-powered loudspeaker lines with enclosure-mounted amplifiers and protection circuitry tuned specifically for the loudspeaker meant the days of rack-mounted power amplifiers were numbered.

But it’s funny the difference that innovation can make. Today’s amplifier is a powerhouse, light years removed from the huge, heavy and inefficient designs of the recent past. Current designs are also far lighter, and in some cases, come in a compact package that more resembles a rack-mount processor than a device capable of generating several thousand watts of audio power.

The addition of DSP into the package hasn’t hurt either, offering convenience, space and cost savings, operating efficiencies, performance advantages and more. It also seems that some users will always prefer their amplifiers on the ground, where they can be quickly and easily accessed if there’s need for service or repair. This is particularly true for live events, where the show must go on.

The primary advancement has been to make the brutes more efficient. Greater output efficiency means less heat, and therefore, less weight, as well as more AC power making it to the loudspeakers.

The most common amplifier topology now is Class D (and variations), which uses an on-off switching method for its transistors called Pulse Width Modulation. Because its output devices are either on or off, the efficiency of the amplifier is greatly increased, and this is done without jeopardizing the integrity of the audio waveform by switching positive and negative output transistors on and off many times per waveform cycle.

This method is analog, but similar in theory to digital sampling where a 44.1 kHz sampling rate is used to accurately capture a 20 kHz signal. This rapid switching creates a square wave that is then low-pass filtered to recreate the audio waveform.

ProSoundWeb offers dozens of articles regarding amplifier classes, designs, data, testing, applications and more. In the meantime, take our Real World Gear gallery tour of audio innovation, the modern power amplifier.

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Friday, October 14, 2011

Real World Gear: Developments In Cardioid Subwoofers

The quest to keep up with the those tall columns of coupled transducers, a.k.a., line arrays

Ever since the introduction of line arrays, it has become harder for subwoofers to keep up with the efficiency of tall columns of coupled transducers in large full-range systems.

This has helped drive transducer and amplifier manufacturers to produce increasingly powerful low-frequency components. Multi-kilowatt dual-18 enclosures are now the standard.

However, while flown full-length line arrays do a superb job of getting full-range sound to the back of enormous venues, stacked subwoofers – even with their advantage of half-space coupling with the floor – must create huge amounts of sound pressure to keep pace with them.

And the omni-directional characteristic of traditional subs pushes that energy in all directions: towards the stage and its performers, and then reflecting off rear walls to produce late arrivals in the main listening areas.

Anyone who’s worked at arena shows with ground-stacked subs can attest to the low-frequency energy wasted backstage. The goal of all sound designs is even, coherent coverage at all frequencies throughout the listening area.

There are many flyable bass reflex subwoofers that are designed as companions for compact line arrays and can be integrated into their arrays and flown above or beside them.

Many of these enclosures, such as the Adamson dual-18 SpekTrix Sub and dual-15 Metrix Sub, JBL dual-12 VT4883, and D.A.S. Audio LX-212R, offer rigging hardware that allows the enclosures to be flown within arrays in seconds, and reversed orientation that allows cardioid arrays to easily be configured by reversing one cabinet’s direction, which inverts the polarity and adds a few milliseconds of delay.

Indeed, cardioid arrays can be constructed from well-manufactured front-loaded subs, and Steve Bush offers a concise description of the three basic types of cardioid sub arrays here.

With access to manufacturers’ prediction software (Electro-Voice LAPS, d&b audiotechnik ArrayCalc, Meyer Sound MAPP, Martin Audio Viewpoint, etc.), it’s easy to set up a scenario and play around with polarity, delay and distance to see the results.

Meanwhile, there are situations where cardioid enclosures provide a compact solution that’s quick and easy.

Companies that manufacture directional multi-element single-cabinet subwoofer enclosures remain a fairly select club. The principal is simple: one or more transducers at the rear cancel the arrival of sound from the front, in a way that also reinforces the energy going forward.

In smaller venues, a single-cabinet cardioid subwoofer enclosure allows system engineers to quickly install a product with predictable results, without a requirement for measurement or additional DSP.

While most cardioid subs are slightly larger than conventional subs, due to their rear-firing transducers, they often take less space and effort than building cardioid arrays. They’re also more fool proof and the stage-hands won’t restack them because they look wrong.

In the challenge for consistent coverage of listening areas, cardioid subs are efficient tools for extending pattern control to a system’s lowest octaves.

When used with point-source cabinets, they combine to make efficient pole-mount systems, especially for delays, where low-frequency energy better aligns with the mains.

When used as a base for ground-supported compact line arrays, they create full-range solutions with a minimal number of enclosures.

By themselves, they can focus low-frequency energy more precisely where it’s wanted.

Take our PSW Photo Gallery Tour of single-cabinet cardioid subwoofer enclosures.

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Wednesday, October 12, 2011

History Files: The Genesis Of Clair Bros To Today

The building of one of the most significant entities in modern touring sound reinforcement, and still going strong

The story of Clair Brothers starts in 1954, when a grocer decided to purchase a PA system as a Christmas gift for his two sons, Gene and Roy Clair.

“He had no knowledge of electronics or anything!” exclaimed Roy Clair of the extremely unusual present.

“I like to think my father was ‘Clair’-voyent in choosing this as a gift.”

The two brothers enjoyed using their PA to provide sound reinforcement for local dances, Easter egg hunts, etc.“The PA bug had bitten us!”

In 1963, Gene and Roy had purchased a loudspeaker re-coning business from a local music store.

This allowed them to acquire loudspeakers at the dealer level, granting the opportunity to build them for a local music store in Lancaster.

When musicians would visit from out of town to purchase loudspeakers, such as Baltimore’s Billy Joel Royal, it allowed Roy and Gene to go hear their products in use at local clubs.

“It was extremely gratifying, and I believe it was then that we realized that working with musicians would somehow be a fun career.

It was the same time we realized that having fun while making money was possible.”

F&M, a local liberal arts college in Lancaster, PA soon requested the brothers’ services to support headlining acts.

Now working in a 4,000-seat facility, one of the largest in the area, the duo would see their first brush with fame in 1966 when Dionne Warwick performed at the college.

“At the time, we had a Bogen MX-60, a few Shure microphones, and two column loudspeakers containing six 8-inch full-range loudspeakers each.

The concert went well, but looking back, we were lucky to start with an easy listening performer or things may have gone entirely different!

Timing and luck is something that has stayed with us our entire careers.”

Not long after working with Warwick, the brothers’ path would cross with Frankie Valli and the Four Seasons at the F&M venue.

Valli showed a vested interest in the duo’s Voice of the Theater A7-500 loudspeakers, particularly since the group had just performed in Miami, FL, and were denied the use of another artists’ sound reinforcement system.

“They were second on the bill to Herb Alpert and the Tijuana Brass at the Fontainebleau Hotel.

Alpert was not only a musician, but also a sound fanatic. It was no surprise that they were carrying their own sound system.

Unfortunately only Alpert would be allowed to use his system, while opening acts would have to settle for using the house PA system – even the Four Seasons’ wives and girlfriends noticed how much better Alpert sounded.

Timing and luck would strike again as our A7’s helped to make their F&M show incredible.

Valli felt they needed their own system if they were going to be successful on the road, and these two young lads were available – and cheap too!”

The brothers were working for $100 per show, including transportation, per-diem and hotels. They obviously weren’t doing it to make money at that point.

“If I remember correctly, after our first tour, in Ohio, we ended up with approximately $40 profit.”

Hardly a profitable tour, even for those days. The brothers weren’t aware of other sound companies touring like they were, but they assumed that they were one of the first to do so.

Touring with the Four Seasons and their continuing work at F&M necessitated a second sound system.

“In the beginning, I think we did a lot of begging and borrowing to do both accounts. Eventually we saved up enough and bought more A7’s.

However, with musical tastes changing as bands got progressively louder, we realized that our A7’s weren’t adequate enough anymore.

We used some of our A7-825 cabinets, and added more power by inserting two loudspeakers in the same-sized box. That seems pretty straightforward by today’s standards, but back then, it was innovation.

We had a slight advantage because we had a double-woofer, horn-loaded cabinet which was portable.

We added power with the first 300 watts per channel Crown DC300 directcoupled amplifier, purchased at the AES Show in 1968 from Clive Moore. It made us unique at the time.”

In 1968, a Cream concert at the Spectrum in Philadelphia, PA, was the now named Clair Brothers’ first large concert with 18,000 people in attendance. “Cream was big luck for us!” states Clair. “Luck and timing rides again!

Bob Kirnan, a sound and lighting technician from New York City who we met while touring with the Four Seasons, was contracted to do the show but was too busy. He recommended Clair Brothers to the show’s promoter.”

With their new Crown amplifiers, Altec Lansing cells with 288-C drivers, paired with Clair Brothers’ bass bottoms containing dual Altec woofers, they seemed to be the perfect fit for the in-the-round performance.

“Coming from Lititz, PA, we were extremely low-profile up to this point. That show in Philadelphia would soon change that…”

The Philadelphia promoter, the Electric Factory, soon started hiring Clair Brothers for shows, in addition to introducing them to many of the San Francisco bands that were successful at the time.

They also worked for the Belkin Brothers in Cleveland, OH, doing one-off shows.

“Their particular sound was instrumental in our company’s next step.

We started appearing on riders as one of the qualified sound companies for concerts, including Hanley from Boston, Kirnan from New York, McCune from San Francisco, and Swanson from Oakland.

Needless to say, Clair Brothers from Lititz didn’t get a lot of attention.”

As business started to increase, the brothers quit their day jobs and focused on building Clair Brothers full-time. They hired their first fulltime employees.

“We were lucky to have incredibly talented people from a rural area that wouldn’t normally be associated with the sound industry in larger cities.

Donald Gehman was our first employee, who did amazing things with Clair Brothers and went on to be one of the recording industry’s best engineers (R.E.M., Still, Mellancamp).

Ron Borthwick, with an EE degree from PENN State, is one of the best engineers in the industry, is still working for Clair Brothers to this day.

Dave Hendel, EE from Lehigh University, who moved on to a computer company. These were some of the few that gave Clair Brothers its start.”

The next four years, from 1968 to 1972, would see product development expand within the company.

Many “firsts” were built by Clair Brothers, including slant monitors, four-way sound systems, electronic crossovers (built by SAE), and the Elvis aluminum hanging system.

“The fourway systems contained W boxes for low end, a double-12 cone for the mid-range – built by Clair Brothers, JBL radials for the high frequency, and JBL for the super-high frequencies. Somewhere in between 1969 and 1970, Clair Brothers switched from Altec Lansing to JBL.”

The year 1970 also saw Bruce Jackson join the company, who would bring new design ideas to the company.

“We also added a lot of accounts at this time, both American and English. Blood, Sweat and Tears was a full-time account that gave us some financial stability.

We then later added Elton John, Moody Blues, Yes, Billy Joel, Bruce Springsteen, the Jacksons, etc., as accounts.”

In 1974, a large leap forward was made by the company with the creation of its S4, single-box loudspeaker system (the first all-in-one four-way box), with its hanging grid system.

Previewed on Rod Stewart’s tour that year, the S4 created industry buzz, to the point that when Mick Jagger came to Stewart’s show, Clair Brothers was hired for the Rolling Stones 1975 tour after he heard the system.

The S4 included high frequency drivers from JBL (2 x 18-inch, 4 x 10-inch, 2 x 2-inch, and 2 x 2405). Truck dimensions played an important role in the sizing of the S4, to allow them to fit two across in a standard trailer.

The S4 has lasted over 36 years, with updates as needed, allowing it to continually serve the touring industry.

The loudspeakers were even used in 2008 for the closing of the NY Mets stadium in New York City.

Clair Brothers’ patented i4 system along with its engineering digital processing continued to drive the company to the forefront of the audio industry. The Lake I/O originally designed by Clair Brothers, which was sold to Lake was a very important part of this next step in the history of innovation.

The dream that Gene and Roy started when they formed Clair Brothers is being kept alive with the second generation, namely Troy and Barry Clair.

The company has formed two divisions, with one son handling each one: Troy handles the touring side, while Barry has run the systems installation side since 1989.

Nearby Manheim, PA, is the site of the new facility has been constructed to house the systems division of the company, Clair Brothers Audio Systems, as the company has outgrown its headquarters.

The freed up space will allow the touring division to continue its own expansion.

Clair Brothers is proud of its rich history, from supporting just one show per night with the Four Seasons in 1966, to now delivering high quality sound systems to a multitude of world class acts, night after night, the world over. The story continues.

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Tuesday, September 27, 2011

Real World Gear: Remote Control Microphone Preamps

Extending live digital networks

In the not-too-distant past, professional touring systems consisted of two or three remotely located consoles, tied to a combination of low-level passive or dynamic input sources (along with some active or phantom-powered inputs), connected by long lengths of multicore snakes, through a system of transformer-isolated or (yikes) passive hard-wired splitters.

The effect on low-level signals with multiple consoles and long lengths of multi-core is a reduction in signal level, higher noise floor and less bandwidth.

The most vulnerable points for distortion to enter a sound system are its electronic stages with the highest gain – microphone preamps and power amps. The character of a sound system is largely determined by the quality of these two largest gain stages.

The ultimate solution is embraced by today’s digital audio distribution and mixing systems which place their inputs close to the source, eliminating loading effects by providing quality mic-pres and digital conversion near the source, controlled from the mix position.

All the best digital consoles provide remotely controlled mic-pres, some of them beginning as digital snakes with mix engines added later.

There are numerous straight multi-channel mic-pres on the market. FireWire interfaces that are easily be attached to computer- based recording systems, but can’t provide remote control from live consoles.

Additionally there are many with ADAT I/O which can even be used with some digital snakes, like the Aphex Anaconda.

Ultimately though, control from a mix position remotely located from the microphone preamps is a requirement for live sound. Actually, any digital console’s remote stage box is a multichannel remotely-controlled microphone pre-amp.

Options for inter-operability between these systems and third-party equipment include MADI, CobraNet, EtherSound and Dante.

We expect to see more of the latter in the near future.

Eight Is Enough?
The number of inputs in multi-channel remote microphone preamps is generally eight or a multiple of that number. Not only is eight a musical number, but it is building block modularity for all consoles.

Several console manufacturers provide mix capacity in excess of the number of mic preamps that come with the console. Most Yamaha digital consoles provide a wide variety of input (and output) expansion through the use of their MY card slots.

Analog outputs provide local monitoring of a pre-amp’s signals and can save the expense of a mic splitter for a monitor desk, recording feeds or broadcast tie lines. Additional connectors in parallel with the main XLRs can be used as a low-cost passive split or provide contractors with easier terminations for installations.

Most digital consoles and snakes operate at 48 kHz, usually good enough for live sound, though there are a few notable 96 kHz products.

The ability of equipment to operate at higher sample rates helps make them future-proof or more acceptable in a recording chain. While most are able to accept external digital word clocks, better products often have superior onboard clocking, providing a benefit to the sound quality of digital consoles they’re connected to beyond the improvement from the performance of the mic-pre itself.

The console of yesterday was a unified enclosed analog system, to which various outboard electronics were added or inserted.

The console of tomorrow is control surface for a mix engine, with distributed inputs and outputs.

Not all manufacturers will have to excel at all of these products. In fact, some winners likely will specialize in one of the disciplines.

Many digital mixing systems can even benefit from additional specialized microphone preamps.

Take our Real World Gear Gallery Tour of the latest preamp models and designs.

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Wednesday, September 21, 2011

Real World Gear: The Latest In Line Source & Column Loudspeakers

With advances ongoing, it’s no wonder that column loudspeakers continue to grow in popularity

This month we look at line source column loudspeakers, which are also sometimes called “architectural” loudspeakers, presumably because their slim-profile design allows them to blend well into their surroundings.

Despite their rather scant appearance, however, many modern column loudspeakers deliver exceptional full-range performance, and this can be furthered by companion LF boxes and subwoofers.

There are two primary types of column loudspeakers – those that offer the ability to “digitally steer” the vertical beamwidth and those that don’t. But it’s not quite that simple, because several models offer some steering capability at a variety of degrees (pardon the pun).

For example, the JBL Professional CBT 70J-1 included in our following report offers switch-selectable narrow and broad vertical dispersion modes. This additional control is attractive given that a primary application for column loudspeakers is reverberant public spaces – houses of worship, auditoriums and passenger terminals.

In fact, the characteristics of line source columns – wide horizontal coverage, minimal vertical coverage above and below the enclosure and coherent sound in the vocal range – are all attractive features for these kinds of venues.

Driver spacing determines the highest frequency at which a column of identical drivers acts as a line source, while the height of the column determines the lowest frequency with directivity. As with modular line arrays, a short system might efficiently throw the midrange, but LF performance can be compromised.

Although they’ve been around for more than a half-century, column loudspeakers are popular because they offer a compromise solution for those who need efficiency in the vocal range combined with even coverage and the aforementioned slim profile. A look into line source coupling behavior and pattern control tells us that loudspeaker cones exhibit coupling behavior up to a frequency whose wavelength is half the distance between adjacent acoustic centers.

It’s common for line source columns to be combined as multiple cabinets to achieve better performance as taller systems for bigger rooms. Longer columns provide pattern control reaching to lower frequencies. A 9-foot column can provide control to 125 Hz, so combining three 3-foot columns can increase low frequency performance.

Digitally steerable models incorporate individual amplification, delay and equalization for each driver, allowing the column’s vertical coverage to be tilted down (or up) and focused for short or long throws, though its horizontal coverage remains fixed. A growing segment of the market is the development of line source systems that are highly portable, with options ranging from the much-lauded Renkus-Heinz IC Live to newer options such as HK Audio Elements.

With advances of this type ongoing it’s no wonder that column loudspeakers continue to grow in popularity.

Take our Real World Gear look at the latest available models.

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Tuesday, September 20, 2011

I/O, I/O: Spotlight On Digital Console Signal Routing & More

Let’s take a look at several proprietary options currently available

Having a roomful of high-qualify professional audio gear is great, but it’s not a system until you can connect everything together.

The hub for system interconnectivity is the mixing console, nowadays usually a digital model. While digital consoles save us the time and cables in not having to hook up racks full of EQs, effects processors, gates, and compressors, there are still a multitude of signals flowing to and from the console that all need to be connected.

Typical inputs for a concert will include microphones and direct boxes (DIs) from the stage, playback device(s), and the ever-present talkback mic at the house console.

Outputs usually include main left and right – and sometimes a center – as well as a separate subwoofer send, a feed to the house tie-in, and increasingly a feed to a recording device.

Don’t forget the split-off snake to the monitor desk and another possible split for multi-track recording. And speaking of monitoring, it’s now common to supply multiple feeds to musicians so they can mix themselves on personal monitoring systems.

Corporate events can add additional inputs, including audience Q&A mics, podium and announcer mics, master of ceremonies wireless, computer audio from presenters, musical cues and “stings,” and lavalier mics for presenters.

Outputs might include feeds to video world, remote recording or broadcast, backstage monitoring, intercoms, second language interpreters, webcasts and podcasts, sends to press mults and the ever present backup safety recording from the console.

Signals are not just in the analog domain any more. Music playback, multi-track recording, and personal monitoring are handled digitally, and digital snakes are becoming the norm as well. Instead of just having to worry about analog audio over XLRs and 1/4-inch connections, today we deal with RJ45, BNC, fiber optic, USB, Firewire, RCA, MIDI, and various multi-pins. And just because there’s an XLR or an RCA jack doesn’t mean that it’s carrying an analog signal, as both connectors can be used to carry digital.

To sort out this increasingly complex situation, many digital consoles are available with proprietary or third-party digital snake systems that help organize and simplify, in addition to facilitating multiple signal splits and complex signal routing. These systems come in configurations, offering remote stage boxes for inputs and outputs as well as expansion ports that accept optional cards (or modules) for different input/output configurations and transport protocols.

The cards are great because they allow us to configure a system more inline with our specific needs, while still allowing for changes down the road. Another benefit is the ability to upgrade current equipment to future transport standards, making current console inventory less prone to obsolescence. The only downside is that cards and modules add cost, but it’s minor and a small price to pay for all the added connectivity.

There are two primary schools of thought on digital console design – integrated and control surface. Integrated units are largely self contained, similar to analog consoles, with inputs and outputs located on the console itself. All processing and signal routing happens inside the workstation. Many of these consoles can be interfaced with remote input racks or with third-party digital snakes that allow inputs and outputs to be located at a remote location, interfacing with multiple channels of audio via a single cable.

Control surfaces, while looking like complete consoles, actually only provide user interfaces like faders, knobs, buttons and display screens. The processing is located inside a rack, usually positioned near the control surface, but in some cases it can also be located at the stage or in another remote area.

These type of systems offer a bit more flexibility, with some providing control over all mix parameters from a computer or other interface, eliminating the need for the control surface itself on some events. Some also let multiple control surfaces access the stage racks, allowing a monitor console or recording console to easily integrate into the system.

Take our tour of several proprietary options currently available.

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Monday, September 19, 2011

Everyone Goes Home Safe: Furthering The Rigging Safety Discussion

Some suggestions as well as specifics about bridling and reeving, two of the most likely culprits in inadvertently setting up potential fault conditions

I began writing this series of rigging safety articles several months ago motivated by the many dangerous practices I’ve seen over the years. (See part 1 and part 2.)

These disasters-waiting- to-happen lurk in clubs, small shows, medium shows, and sizeable temporary events.

This past summer, we have witnessed staging/rigging tragedies in Canada, Indiana, and Belgium. The common element in each of these truly unfortunate incidents are media reports of extreme weather conditions. There’s a very human tendency to believe these “acts of nature” are unavoidable, and in some respects, it’s true.

But those who promote outdoor shows, and those who stage them, have an implicit responsibility to both closely monitor worst-case weather patterns and to have decisive, immediate, and safe plans of action to do everything possible to keep everyone safe.

Further, the reality of a sudden storm – versus a slow onset – has become all too well-known in recent weeks, but regardless, the stage and all associated structures must be designed and built to withstand the worst that nature has been known to deal out in the past, plus a margin of headroom. Must be. Anything less is a gamble; a misguided dice roll that risks other people’s lives and well being.

A suggestion: if an event is held in an area known for weather conditions that cannot be mitigated by strength-of-build, such as violent storm squalls or tornados capable of leveling permanent construction, then further precautions must be undertaken. For example, spotters could be deployed a mile or two out, watching the skies and able to radio in warnings in time to get everyone off the stage and away from the towers.

At the least, a competent individual should be assigned to monitor all available weather reports. He /she must have the authority, and the clarity of purpose, to stop the show at the first sign of trouble. A false alarm or two is far better than waiting until it’s too late.

In addition, fall zones should be established around the stage, and any support towers and cordoned off, so that in the event of structural failure, no one will be in the danger zone when the structure collapses.

Safety guy wires (wires that run from ground anchors to the top of the stage structure) can greatly help to stabilize a structure when it’s subjected to wind load - and even seismic loading - though they also produce a vector force of their own pulling outwards and downwards: the structure therefore must be strong enough to handle the added load of the safety wires.

Safety wires represent a lot of extra work and expense - but nowhere near the cost of a single human life. It’s also possible to add extra tension to the stabilizing wires in the rear so that if the structure does fail, it will topple rearwards, away from audience members, rather than forwards and towards them.

Further Steps
Load cells - small devices that provide tensile information electronically to a remote readout - are relatively inexpensive these days. They can be built into the stabilizing wires and into other key load points of a stage, or a sound tower or lighting tower. The increasing force from a freshening storm can be monitored electronically from a safe distance.

A rapid-deployment retractable roof system can be readily designed to remove the wind load on the huge fabric surface in seconds, in the event of a storm that sneaks up suddenly. Using the same mechanical arrangement as a ball-bearing sail track on a sailboat, and a few electric cable-spools, a roof could be retracted in seconds.

Sure, the equipment on stage is going to get wet. That’s O.K. if lives are saved. (And anyway, that’s what tarps are for.) There’s likely to be plenty of time to deploy tarpaulins, but only after the threat of the roof collapsing has been reduced by removing the roof “sail” area.

The bottom line is that a wide range of solutions need to be considered, adopted, and implemented.

The alternative, which I fear we’re close to, is the cancellation of all future outdoor events until safety guidelines are drafted by a government agency, and a method of policing the guidelines is put in place. That could take years.

So more than ever, I believe that we desperately need to establish a safety charter that governs all 50 of the United States.

If we don’t, it will be done for us and we probably won’t like the results.

Fault Conditions
Now let’s pick up where we left off last month: bridling and reeving. This is more than incidental knowledge because these two practices, while often misunderstood, are two of the most likely culprits for inadvertently setting up potential fault conditions.

Bridling doesn’t refer to horses, as you may have guessed. It refers to angular loads. When a load is suspended by two or more suspension points that are not directly in line with the mass of the load, the resultant rig is said to be bridled.

As you can see in Figure 1, the vector force of a bridle can create loads within the rigging parts that far exceed the actual load of the object itself. This does not mean that the 100-pound object used in these four examples will ever weigh more than 100 pounds. Of course not.

But the load on the rigging parts can greatly exceed 100 pounds due to the vector loading of those parts (which might be wire rope, Spansets, slings, chain, or other materials).

Because this is counterintuitive, I like to show the effect in real time by using three 30-pound fish scales - which I do when teaching classes in person. I’ll actually rig up a 10-pound load with a sliding spreader bar. I put one fish scale on the top-most hang point – which is always going to read 10 pounds – and then put the other two scales on each of the two bridle parts of the rigging.

Then, as the angles of the bridle are adjusted, everyone can see how the load on each of the bridle parts varies. It’s particularly dramatic when the angles are acute in either compression or expansion, and the fish scales quickly max out at 30 pounds - then suddenly the light bulb goes off!

As the bridle angle approaches 90 degrees (full horizontal) the loading on each bridle part mathematically approaches infinite. In real life that doesn’t happen because of material elasticity and other factors. But it’s worth knowing that the true force in a bridle part can exceed the actual mass of the suspended object by a very large factor.

Greater Forces
It’s equally important to understand that an acute bridle angle places a considerable compression (or expansion) force on the object of mass, as a function of the angle of the bridle.

While this may be no problem for an industrial spreader beam that came at “no extra charge” with your recent 90-ton crane purchase, it often is a considerable problem when you suspend a loudspeaker, or loudspeaker array, using an acute bridle arrangement.

Many suspension frames are simply not designed or built to be loaded at sharp angles, and especially with far greater forces than the actual weight they were intended to support.

Not all frame designers and manufacturers are adept at imagining the full gamut of strange (and inappropriate) rigging arrangements that their frames may become subjected to.

If a loudspeaker uses simple eyebolts as a means of attachment (this is very common in small systems), even a shallow bridle angle will subject the eyebolts to forces they were not designed to accommodate.

Eyebolts – even the good ones – are intended primarily for a straight line-of-force load that aligns with their threads. (Figure 2) They should never be loaded laterally.

Though they can tolerate loading in the direction of their eye, as the angle decreases from a straight line of force, their load rating in turn decreases dramatically.

Other devices such as rigging frames, bumpers, etc, can exhibit similar limitations as eyebolts, but here the issues and properties are product specific.

One product may be able to tolerate an acute bridle, while another may not. It depends on the design geometry of the device being rigged, so there is no simple or single answer.

That said, the additional stress that’s put on the wire rope, the shackles, the Spansets and the other parts of the “hang” will all be magnified by using an acute bridle – even if the rigging frame or bumper itself can handle the load. Best idea: always use shallow bridles!

Rule of Life (this is like a “Rule of Thumb” but more so): do not bridle past a 30-degree angle (0 degrees being a straight line pull). See Figure 3. Doing so will place a significant compression force on the loudspeaker (or other object) that you’re intending to suspend.

Situations may arise in which you must bridle tighter than 30 degrees. In such case, the shackles should be oriented towards the line-of-force so that they pivot easily towards the load point. If there are eyebolts in the rig, they will need to become expensive swivel eyes, rather than inexpensive static eyebolts.

With sharp bridle angles, considerable care must be exercised to insure that no parts are overloaded.

Stress Effects Of Reeving
Reeving is when a Spanset, nylon strap, wire rope, or other suspension part that’s already fastened to an attachment point is then passed through a shackle, an eyebolt, a handle, or some other fitting, and then is fastened to a third attachment point.

It’s common to see reeving in all manner of rigs ranging from small club systems to large PA hangs.

For reasons unknown, people just like to do this. I’ve overheard talk in which the reckless reever believes it’s actually safer to do so. I did it once or twice myself, before I understood the stress effects.

What happens with reeving is a lot like a 90-degree bridle. The force on the reeved part, as well as the shackles and the other fittings associated with the reeve, is magnified many times over to the point where it can damage otherwise perfectly constructed materials and weldments in a fly frame. The compression force on the object, be it a loudspeaker, a rigging frame, or other, will be many times that of the object’s actual weight.

Additionally, the load can rotate or shift within the reeve, increasing the stress yet again when the load shifts – very possibly leading to sudden failure. Therefore don’t do it!

The only arguable reason might be to “use up” an excessive length of a Spanset or wire rope sling, in order to shorten the distance to the upper attachment point(s) when headroom is limited. Sorry, that’s not an acceptable reason. Obtain the proper length of wire rope slings or Spansets. Again, do not reeve!

Now, just so I don’t get letters from irate construction workers, reeving does have a place in the construction industry. If you reeve the attachment fittings on a 100-ton water valve, you can use your crew to shift the valve within the cradle created by the reeve to align it with the 80-inch pipe you’re fitting it to.

But please note that when this takes place, it does not occur over people’s heads. And it usually doesn’t last for more than a few moments of work in a restricted-access construction zone with trained personnel. Neither of the aforementioned conditions typically prevail in an entertainment technology rigging setting.

Straight & Lateral
A frequently asked question: do pairing rings always come in pairs (like earrings)? No. You can buy them one at a time. In the previous installment, I harped on the fact that shackles are not pairing rings (Figure 4); in other words you should not use a shackle to join two bridle parts together, unless the bridle angle is very shallow and the shackle is rated for limited “pairing” use.

So instead, you’ll want to carry a selection of pairing rings in your rigging kit. A pairing ring is intended to take lateral forces in addition to a straight line force. Like anything else, it too has its limitations. You can’t pull it in all directions at once; if you need to do that, you’ll want a Weldless Circular Ring, such as the one shown above, which is also easy to obtain from any good industrial supplier.

And when using either a pairing ring or a circular ring, simply attach the shackles to the ring. In both cases, you’ve avoided placing lateral loads on the shackles themselves. Now it’s all good.

While we’ve conquered many things once deemed impossible but now takenf or granted – such as air transport, wireless communication, modern PA systems of amazing quality, and much, much more - let not our quest for pure technology overshadow our intrinsic responsibility to our community.

Be it a rock concert or a political rally, we need to be able to say, “Everyone gets to go home safely.”

Ken DeLoria is the founder and former owner of Apogee Sound, a manufacturer of loudspeakers and many associated rigging accessories. For more than 30 years, he has been a sound engineer, a hands-on rigger, and a safety supervisor at numerous events and permanent installations.

(See part 1 and part 2.)

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Friday, September 16, 2011

Consoles & Mixers For Churches, In Context

Options are plentiful and growing in this dynamic marketplace

Over the past decade, we’ve seen the proliferation of digital consoles in both live and recorded sound.

This is not to say that analog consoles are going away at all, particularly with respect to house of worship systems.

In fact, it’s a very safe bet that the vast majority of church systems are still headed by an analog console.

Hector La Torre, Managing Partner and producer of the national HOW-TO Sound Workshops, notes that while digital consoles are a major topic of discussion at his organization’s audio education seminars presented to more than 1,500 church sound personnel annually - largely volunteers - throughout the U.S., most churches are still hesitant to dip their toes into the digital technology stream.

“There are two primary factors with respect to the bulk of the church sound market - cost and complexity,” La Torre explains.

“Digital consoles have largely been out of the price range of all but the largest churches, and while digital mixers are not necessarily more complicated to operate, keep in mind that about 95 percent of church sound system operators are volunteers who have limited experience, and that’s who is being asked to take on a new learning curve.

“Although most churches and volunteers who take on digital consoles find that they become more efficient and proficient at their job, some still hesitate because of the initial learning curve and overall lack of knowledge of the technology.

“A board for a higher end professional application like a tour or a performing arts center is usually an upgraded version from what you’ll find being used in your average church service,” he adds.

“And in the mainstream of the professional audio marketplace, new technology often wins out over cost issues, but it’s pretty much the opposite with the majority of churches.”

“That’s why education is the key to the future of digital consoles in worship. If church folks don’t know or understand a technology, they won’t adopt it.”

Both aspects are changing, with manufacturers now increasingly introducing digital models in line with church budgetary needs while maintaining functionality and feature sets to meet all but specialized applications.

Affordable digital consoles cited by La Torre are the Yamaha LS9 Series, as well as models from Tascam and Soundcraft, and he’s also talked with a number of other manufacturers who indicate they’re quickly moving in the same direction.

On The Upswing
Some context about the church market is in order. There are an estimated 450,000-plus churches in the U.S. alone, and three-fourths (and likely more) of that number is comprised of venues offering seating for 500 or less, with the norm in this range being 250-300 seats.

While we read about sophisticated church sound systems (that often include one or more digital consoles) on a regular basis, these are often deployed at larger venues ranging in scope up to the “megachurch” realm.

The production needs at most churches are not nearly as ambitious, budgets follow that scale, and the “technical staff ” is made up of a few volunteers who might spend their weekdays selling insurance, driving a truck, teaching school and so on.

Within this context, there are countless analog consoles and mixers that are at least 15 years old still working great and meeting expectations, and when a church is seeking a new board, their mindset has still tended to analog.

Acceptable Result
In general, worship services are generally one of two categories - traditional or blended/contemporary.

Let’s look at traditional first. Often there will be a pulpit microphone, an altar mic, a lectern mic, a wireless mic on the pastor, and perhaps a feed from a digital piano.

Chances are this type of system has been in place for quite a while, and the mixer is sometimes mounted in the rack and offers only volume control for the various inputs. The rack is stuck out of the way in a closet somewhere - “set it and forget it.”

If this mixer needs to be replaced, a similar analog rack-mount mixer with level controls only and a good equalizer (properly tuned) can create a reasonably acceptable result.

The downside is that there is no individual EQ for the various input devices, so all channels will have to compromise with the EQ needs of the other channels.

For those seeking more capability but wishing to stay in the rack realm, a digital rack-mount mixer is an option.

Once installed, the system contractor hooks up a laptop to the sound system, and via system software, uses the digital output equalization to give the sanctuary as close to a flat response as the loudspeakers and the room itself will allow.

After that, the contractor can set the digital EQ for each individual input device to get the most pleasing possible result for that specific channel. No compromises are necessary.

Further, a simple touch-panel can be provided at a remote location, allowing for several types of adjustments by the user without the possibility of damaging the loudspeakers or causing feedback.

Of course, another choice is to replace the rack-mount mixer with a small console/mixer that offers EQ on all individual input channels.

This can also present accessibility advantages, as well as locating the operator in the sound coverage field to make adjustments.

And, most of these models usually offer more than eight channels at attractive price points, making them well-suited to meet future system expansion.

Changing Rapidly
Some traditional services - up through virtually all blended/contemporary services - need a true mixing console, hopefully located in the primary listening area and manned by a competent system operator.

The size of the console is determined by the number of channels needed plus future growth considerations, and even a few more for good measure.

Both analog and digital consoles/mixers definitely track with the rule of “you get what you pay for” - particularly in terms of analog, there are some inexpensive models that are surprisingly decent, and there are some absurdly inexpensive models that should be absolutely avoided.

The system operator is another key to the selection process. A skilled, experienced operator can make beneficial use of a more sophisticated, feature-laden console.

The control surfaces of early digital consoles/mixers were not very intuitive for the majority of church operators.

As noted, that’s changing rapidly, with most modern models offering user-friendly interfaces that track well with analog intuitions.

As a result, any operator with a decent amount of analog experience can now successfully perform basic digital mixing. This is probably even more true for younger operators, who have practically grown up using touch screens to scroll through menus and so on.

Digital console platforms offer several advantages for church sound applications.

The dedicated offering of EQ and dynamics to every single channel (in addition to scads of other operations) mean more highly tailored and adaptive sound, as well as saving the space and cost required for outboard gear providing the same capabilities.

Further, there’s a lot of value in the ability to save preset mix “scenes” for instant recall so that specific settings may be accessed instantly at the touch of a button.

This is quite useful for contemporary services that can feature dozens of performers and talkers appearing in rapid succession, as well as at churches where the first service is traditional, the second service is blended and the third service is full-blown contemporary.

It can also be a significant benefit to the pastor who needs to conduct a funeral service or a simple wedding ceremony mid-week without benefit of a system operator.

Times they are a-changin’ with mixing consoles, and it’s all good for churches.

Digital is coming on for very good reasons, but there’s still a plethora of options for those who prefer an analog workflow and layout.

Jon Baumgartner is a veteran system designer for Sound Solutions in Eastern Iowa, and Keith Clark is editor-in-chief of ProSoundWeb and Live Sound International.

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RE/P Files: Recording Forum, The Trend Toward Self-Production, Part I

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature is an amazing look back at "the growing trend of self-production" as discussed by Phil Spector, Elliot Mazer, Gabriel Meckler, Dave Hassinger, Denny Cordell, Ian Anderson, and Leon Russell. This discussion dates back to April / May of 1970. (Volume 1, Number 1). The text is presented unaltered, along with all original graphics. A pdf for a print-out of the original article is located on page 3.

I think the question could best be answered by the analogy: “What’s wrong with wetting your finger to measure wind velocity?”

In truth, the VU meter does a great job of indicating the average level of a constant state signal.

However the VU meter becomes unpredictable when attempting to measure the actual levels involved in a transient passage or in a complex waveform.

Since music is composed of complex waveforms and transient passages a problem in accurate monitoring does exist.

Engineers have been aware of the problem for years and have learned to cope with it.

Engineers had learned that they could depend on instantaneous peak levels to be about 6 to 8 dB higher than that indicated on the VU meter when the program material was orchestral or vocal music.

When engineers were dealing with ordinary performance mixed orchestral or vocal music, on one or two tracks, they simply allowed themselves 10 dB of “headroom” in the electronics, and recorded at a level of 6 dB below the 3% distortion point of the tape.

If a cleaner recording was required, the record level was reduced by another 4 to 6 dB thereby reducing the 3% distortion peaks.

With only one or two tracks, noise was not the problem it is today. Further, the public ear had been conditioned to accept the transient distortion and the tape noise, even if these had not been masked by the noise levels and distortions produced within the available playback equipment.

Today-it’s been happening for about the last five years—we find that our crutches have been kicked out from under us. Each of the following series of events has contributed to the unsuitability of the VU meter as an accurate monitor of levels in modern recording practice.

Multi Track Recording
Today, we are monitoring levels of individual instruments more often than we are monitoring ordinary performance mixed material. Consequently, peak levels do not follow traditional 6 to 8 dB reading error. Take, for instance, a tambourine.

Traditionally, it was buried in the mix someplace and its transient peaks were low enough not to be a serious problem. Now take the same tambourine and put it on a track of its own. Try to record it at “O” level and you will find yourself in big trouble.

Depending on the particular instrument, a specific microphone and the amount of equalization used, you will probably find your VU meter reading to be between 12 and 16 dB off. The only procedure you can follow is to make a calculated guess as to the proper VU reading and go.

If you have guessed too high, the result will be transient distortion and a lack of presence. If the recorded level is too low, you have an excessive amount of noise, which leads to the next modern-day metering problem.

Noise
Noise has suddenly become a paramount consideration in the commercial recording industry.

Consider the fact that in most mixdown sessions, you are combining the tape noise of as many as 24 tracks, possibly more if “ping ponging” of tracks has been done.

If an acceptable overall noise level is to be achieved, it is an absolute necessity that each track receive as high a level of signal as possible . . . yet not high enough to cause distortion.

This, of course, is the original function of the VU meter.

Unfortunately, the VU meter is not capable of this function with enough accuracy for the complex or transient signals of today’s recording.

Program Content
As has been shown above, there is obviously a very fine line between proper and improper recording levels for complex and transient nature waveforms.

If we are going to come up with recordings that satisfy today’s critical listener, then we are going to have to hit that very fine line with every track. An analysis of a typical 1970, 16 track, group session will well illustrate the point:

Track No.
1 - electric base
2 - very tight transient drum set
3 - super-equalized acoustic guitars
4 - organ
5 - headless tambourine
6 - sandpaper blocks
7 - lead vocal
8 - background vocal group
9 - double of track 8
10 - handclaps and vocal percussion
11 - electronic music synthesizer track
12 - overdubbed drum accents
13 - maracas and more handclaps
14 - horns
15 - orchestra bells and vibes
16 - piano accents and fuzz guitar lines

Now, the group wants a really clean sound, and the producer wants everything very “hot” (because if you don’t record hot you’ll get a lot of noise on mixdown).

So they all crowd around you like a football team and “help” you watch your meters! (Of course you need help because the meters are strung out over eight feet of control board and your field of vision only covers thirty inches.)

At this point, the inadequacy of VU meters really hits home. You are faced with a real killer of a situation consisting of:

- very transient and delicate program material
- a critical and discerning production staff
- a tight margin of required accuracy
- a distracting environment

In the midst of all of this you sit trying to compute the error factor of 16 meters whose individual errors range from 4 to 16dB, and whose smiling faces require three men and a lap dog to monitor.

I think that these considerations would graphically answer the question of “What’s wrong with the good old VU meter?

Times have changed, requirements are different, and our new technology enables us to provide ourselves with better methods of doing things.

What Is The Answer?
In order to arrive at the ultimate metering methods for use in modern recording systems, I think we should look closely at the requirements.

The basic requirement is the same as it has always been…to enable us to record in the optimum region between noise and distortion.

This can best be accomplished by placing the recorded signal as near the distortion point as possible without crossing that fine line into audible degradation.

Distortion, obviously, is an instantaneous function of the waveform. As such, the only way to accurately control it is to be able to meter the instantaneous (or peak) levels of signal.

Once we have this sort of metering, we can establish an accurate correlation between meter reading and amount of the distortion introduced.

In order that the meter be capable of measuring high frequency transients properly, it is desirable that its attack time be on the order of 100 microseconds or less. This attack, or upwards deflection must then be followed by a slower release time (or downward deflection), for two reasons.

First, so that the human eye will be capable of following the meter movement easily. Secondly, so that the meter will tend to integrate a rapid passage of high frequency peaks into a readable display. This would call for release times on the order of 25 to 100 milliseconds.

Judging from the current state of the art, it would appear that such a meter would, in all probability, be one utilizing a segmented light display.

To accomplish the desired degree of accuracy in such a meter, the number of segments should be in excess of fifteen. Otherwise the dB difference between increments will tend to be too great for accurate readout.

Additionally, there is a question as to the desirability of a single spot of light. Tests conducted in a recording studio environment have shown a single moving point of light display to be much more readable and considerably less fatiguing to the eye, than a widening line of light. It has been further observed that each increment of light must be either “on” or “off.”

That is to say that no segment should be allowed to be half on or dimly lit while its companion is in “on” mode. Should this not be the case, readability suffers and eye fatigue tends to increase.

It is definitely within the realm of our industry to develop such a device. When the hardware has been developed, the progressive engineer will have little choice but to change his thinking from VU metering to “Peak Level Metering.”

This, of course, will require the establishment of a new set of ground rules in the area of studio level measurements and will almost certainly result in a large number of “Bah Humbugs.”

We do seem to recall, some years ago, several chorus’ of “Bah Humbug” when someone wrote an article entitled “WHAT’S WRONG WITH THE GOOD OLD VACUUM TUBE? . . .

 
Downloadable Media
Original Article (pdf)
Original Cover (pdf)

Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970

Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

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Wednesday, September 14, 2011

In Profile: Maryland Sound’s Bob Goldstein, Doing Really Big Really Well

“Doing shows is our first, primary love. There’s nothing better than a great sounding show.” - Bob Goldstein

It’s fair to say that Bob Goldstein loves a challenge, and the bigger, the better.

“We do really, really big, really, really well. Everyone puts in the effort. There’s no quit,” he says.

Consequently, when Goldstein talks about his life and career, he does so by relating anecdotes that focus on the challenges – personal and professional – that he and MSI have proudly met over time.

Some are a result of the incredibly complex jobs he delights in taking on, while others are the product of a love affair with the arts that dates back to the very beginning of his career, from when he first began playing bass guitar in local Baltimore-based bands in 1961.

That love has always fueled his work in audio, he explains, as well as an abiding interest in history, art and architecture that has, literally, caused him to lose a fair bit of sleep over the years. “As long as I was on the road, I never did a tour bus. I’d always make a point of driving through places, going to museums and historical sites. It was tough because I never slept, but I liked driving the truck because it allowed me to stop and see things.”

As a result, Goldstein has a wealth of stories well worth telling. Stories that range from good, like the company building its first MSI “Super-board” from scratch for an early Andy Gibb tour in 30 days, to downright ugly, like an early misadventure with electricity that prompted him to rethink his pursuit of a career in music.

“When I was 15 I picked up an electrical cord. I had a bit of moisture in my left hand and the thing exploded and burned all the skin off of it. I had a gig the next night, and it’s hard to fret a bass with the skin burnt off your hand. So I realized right there, this is a really fragile existence.”

It was that realization that would ultimately lead him to a life on the road and the founding of MSI, a company that has gained a reputation for taking on jobs considered virtually impossible, and doing so with a degree of calm that has often prompted clients and competitors to shake their heads in disbelief.

Taking It On
Anyone responsible for sound reinforcement in large outdoor venues faces challenges, including huge storms and often punishing timelines. But some that MSI has faced trump weather and scheduling issues in a big way: chief among them negotiating security while trying to provide sound for more than 2 million people - the largest audience to attend a civic event served by an outdoor system in the history of the U.S – for the January 2009 inauguration of U.S. president Barrack Obama. (Go here to see our Photo Gallery of the system for the inauguration.)

Then there’s the Times Square New Year’s Eve extravaganza, which had never had sound reinforcement prior to 1998-99. It was an event other companies approached for the job considered impossible, but also suggested organizers speak to a “crazy company from Baltimore,” who might just take it on.

In all, there were 150 conditions governing what type of infrastructure could and could not be used. “

No generators, structures, or wires, nothing you could climb on, nothing with fuel. And no testing time; you have to set it up and it has to work perfectly,” Goldstein says.

The solution: individual negotiations with owners of surrounding buildings to get clearance to drill holes in those buildings and crane-mount loudspeakers that would tap into existing A/C infrastructure to power a unique microwave system to transmit signal. And although the system has changed since then, MSI has been contracted to provide sound for the event every year.

Goldstein’s passion for audio manifested itself initially while he was growing up in Baltimore. “When I was 11 my grandmother bought me a bass guitar. There were no bass amps that were worth anything back then, so I built my own and taught myself what sounded good and what didn’t.”

He went on to gain a reputation as a bass guy, he adds. Correspondingly, MSI always set the bar very high for bass devices, providing some of the most powerful of the time and prompting a member of the Commodores to comment, “A lot of guys got bass, but MSI’s got the thunder.”

Early Touring
By age 16, Goldstein was working as a musician and as a sales manager for a local electronics company, a combination that led to his first job as an engineer/DJ, a regular weekend gig in the summer of 1966 at Baltimore’s inner harbor.

Working from midnight to 6 am, he soon got to know the staff and owners of numerous local nightclubs, and it led to a house gig at Club Venus, host to numerous well-known acts of the time. “That was when I started Maryland Sound,” he adds.

When Frankie Valli and The Four Seasons came through in 1968, Clair Brothers asked Goldstein to tour as their mix engineer. Studying architecture at the University of Maryland at the time, Goldstein jumped at the chance, becoming one of the first touring engineers for the company.

Although he wanted to be an architect, he also wanted to be a sound engineer, and there was no formal training available for audio. “I took engineering and music composition courses, but you couldn’t really put a useful group of classes together.” Instead, he opted to continue his education in sound on the road. His first lesson: expect the unexpected.

“I had a bunch of Altec Voice of the Theatre (loudspeakers), power amps, and a mixing console made up of Altec rack mounts strapped together with a treble and bass control on every five mics.” All packed, he adds, “in my brand-new blue Dodge Maxi Van.”

After his first show, at Temple University in Ambler, PA, Goldstein got into his van and headed for the next gig.

Just as he was coming out of the Lehigh Tunnel in his sparkling new ride, around 3 am, he felt something hit the front of the van.

“I hear BAM, pull over to the side of the road, and there’s blood and guts all over the side of my new van. And there, in the middle of the road, about 150 yards back, is this headless Black Angus steer. He must have lifted his head just as I hit. If I’d been driving a foot and a half to the left I’d probably be dead.”

Goldstein found the head wedged into his front bumper, but what first occurred to him wasn’t how to pull it free, it was what to do with the rest of the beast.

“My family would buy a quarter of a steer for, I think, $1,800 to $2,000 back then. I thought, my god, this thing’s worth 8 grand, so I’m trying to figure out if I can strap it on the van. You’re not thinking, right? Then I realize I’m going to be out on the road for six months and it’s going to stink by tomorrow, so I gave up.

“My first day on the road,” he adds, laughing, “and I said, ‘well, this is going to be an interesting life’.”

Building A Reputation
Two years into the gig, when Clair Brothers asked him to take on another project, Goldstein felt he couldn’t leave Frankie Valli. “You’re talking about a show that has a lot of cues. You can’t just put someone else in there without training them on what Frankie and the band liked, and I didn’t like the idea of leaving them high and dry.”

He still considers the band his favorite. “It was where I started. When I played music that was what I played - doo-wop, Motown and R&B.”

He also felt he was an integral part of delivering the music to the audience, a role he takes enduring pride in. And although he did tour with many acts over time, he continued working with The Four Seasons until 1985, at times doing upwards of 200 shows a year.

Building MSI from the road before the advent of cell phones and email was nearly an impossible task in itself, but Goldstein explains, “You do a good job and people start asking ‘can you help with this or that?’ You start hiring people, sending them out and they do a good job. Then you hire more people, they do a good job, and you get a good reputation.”

As MSI grew it integrated other companies to expand its capabilities and geographical reach, among them Northwest Sound and Stanal Sound, and since, has branched out into virtually every segment of the industry. But concert sound remains Goldstein’s greatest pleasure.

“Doing shows is our first, primary love. There’s nothing better than a great sounding show.” Having said that, Goldstein stopped doing sound personally in 1985 to focus on his company. “I didn’t mix for 20 years. Some could argue I should have stayed on the road instead of tinkering with what was working at the office,” he says, laughing again.

Pushing Limits
A self-admitted perfectionist, high production values and great musicianship are what Goldstein has always valued most highly in an act; artists who push their own limits and thus force him to push his. “Great voices,” he explains, like Whitney Houston, Frankie Valli, Minnie Ripperton, Mariah Carey, Daryl Hall and Josh Groban – who he began mixing when he felt able to delegate the day-to-day running of MSI to others.

“I went back to mix Josh Groban in 2004, and what I’d forgotten about when I was in the office for all that time was that addiction you have for the drug of the audience. When they go nuts it’s not for you, but part of it is, and it’s a great feeling knowing you translated the music well.”

As for the future, Goldstein sees MSI working on more and more television, broadcast and corporate applications and continuing to push the limits of possibility at ever-larger events and concerts. Again, he stresses, the bigger, the better.

“Tell us there’s something we can’t do. We thrive on that. We like doing big. Anytime somebody’s got a big project, send it on this way. We don’t mind doing small, but we love really, really, big. If you can think of something that will draw 15 million people, give us a call. We would really enjoy that.”

Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer.

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