Loudspeaker

Thursday, January 17, 2013

Everything Breaks: Rigging Components & Vital Safety Factors

The only question is under what load and under what conditions...

“Don’t go breaking my heart” —Elton John & Kiki Dee

Whether it’s a shackle, a hoist, a crane, or a beefy fly frame, never forget that everything breaks. The only question is under what load and under what conditions.

To address this issue, the rigging industry has adopted the term “SWL,” which stands for Safe Working Load. You might also see “WLL,” which stands for Working Load Limit. When you purchase a shackle, a pairing ring, or a length of wire rope, they’ll all carry a load rating - unless they are cheap copies of the real thing (more on this later).

Let’s start with the common anchor shackle. An anchor shackle is the bread and butter of the entertainment rigging industry.

It can be obtained in many sizes, and made from several types of material. It’s close cousin is the chain shackle, which normally carries the same ratings for a given size, but is not as commonly used (Figure 1).

The stated size of a shackle relates to the nominal diameter of the threads that secure the pin into the body of the shackle. The unthreaded section of the pin will always be larger in diameter.

This is important to know if you’re building a steel frame or fixture that the shackle will attach to. A 1/2-inch shackle will need a hole diameter of at least 5/8 of an inch or larger, to fit without jamming. 

(click to enlarge)

Loudspeaker rigging mainly utilizes shackle sizes of 1/2-inch, 5/8-inch, and 3/4-inch It’s rare to encounter any that are larger, though smaller ones are sometimes seen for light loads.

Typically, a 1/2-inch shackle will be rated at two-and-a-half tons or 5,000 pounds. A 5/8-inch shackle is usually rated at three-and-one-quarter tons (6,500 pounds). The rating is usually displayed on the shackle like this: “3¼T”.

But what does this mean precisely? Can you load them up to their stated rating? Absolutely not!

Most ratings, especially of older products, are based on a 4:1 safety margin. This was the safety standard that was used in the crane and rigging industry for many decades. Some newer parts may be rated at 5:1, or an even higher ratio, but you can’t count on it unless it’s clearly notated on the part itself, or the spool it’s wound on (for wire rope). 

Nonetheless, in our industry we have to go further. We don’t use shackles on construction sites with trained crane operators and ground-personnel - all working with a constant awareness of the possibility of a shackle or a cable failing. No.

Instead, we suspend loads over people’s heads who are paying attention to anything but the loads we suspend in the air. They’re looking at the stage, they’re dancing, they’re socializing. 

Entertainment rigging has largely adopted a 7:1 safety margin in the U.S., while in the European Union, the safety margin is mandated at 10:1, usually followed in most countries.

At a 10:1 ratio, if the failure point (also called “ultimate strength”) of a shackle or other device is 10,000 pounds, then you should never load the device to more than 1,000 pounds.

And even that rating, gentle rigger, is only valid under a static load. A dynamic load, which occurs when a chain motor starts and stops - or a gust of wind hits – can put a far greater force on the suspension equipment than that of a static load.

And then there’s material fatigue. As the sidebar explains, the shackles and the wire rope slings you routinely grab from your rigging box are all not necessarily created equally.

(click to enlarge)

Direction Of Load
Shackles are designed to be loaded in a straight line; they should never be turned sideways (Figure 2).

While this isn’t breaking news to anyone, it’s not so well known that bridling two wire rope slings, or two Spansets in the eye of a shackle, is NOT what a shackle is intended for.

Yes, you can do this and usually get away with it, but it can weaken the shackle’s ultimate strength by a considerable margin.

The weakening is a function of the angle of the bridle: the more acute the angle, the more the shackle is being pulled in multiple directions and therefore weakened. The proper method is to use a pairing ring.

(click to enlarge)

In Figure 3, the vertical arrows represent the vector force within the shackle, while the angular arrows represent the actual force when a shackle has been subjected to a semi lateral line-of-force.

Anchor shackles may look like they’re designed to take angular loads, because of the way they conveniently flare out in the top part of their eye, but they are not! They are intended for straight line-of-force loads only, or at best, very shallow bridle angles.

Some shackles, usually those of very high quality, will be imprinted with marks that represent the maximum permissible vector loading. In such cases, it may be O.K. to utilize a shackle as a pairing ring, but only when a shallow bridle angle is presented. 

Sourcing Components
Only the highest quality materials and components should be used in any entertainment rigging application.

All reputable manufacturers of rigging components employ scientists, metallurgists, and engineers to ensure that their products are designed properly, manufactured uniformly, and will reliably meet their stated load ratings. They use standardized test procedures, proven over decades, to ensure quality and reliability. 

Conversely “copy cat” manufacturers often do nothing to ensure safety. In these modern times of international trade, thousands of suppliers exist that sell cheap parts that may closely resemble the originals they’re based upon, but these suppliers sometimes have little to no regard for meeting any stated rating – if they state it at all.

Such products, which are found many places, including local hardware stores, are unsafe and can fail unexpectedly. Do not purchase them. 

If this article can convince you of anything, please let it convince you to purchase only the highest quality parts, materials, and components. The money you’ll save in the local hardware store for a length of wire rope, or a handful of shackles, will not pay for even one hour of legal representation. Are you with me so far? 

Reputable companies such as The Crosby Group have been manufacturing high quality components in the U.S. for many decades. They publish meaningful manufacturing data, and they stand behind their work. Numerous other companies, domestic and abroad,  also meet high-quality standards. Seek them out.

Establish a commercial account with a suitable supplier, especially if you intend to become a regular customer. If your usage is sporadic, consider using a distributor like McMaster-Carr, Grainger, or another industrial supply house.

Don’t be afraid to speak with the sales department and demand – yes demand – the material data sheets on each product that will let you sleep well at night. If your supplier can’t provide the information that you request, find another one.

At all costs, do not accept off-shore substitutes that may look the same as an engineered product, but might possibly perform at a far lower standard.

If you purchase your parts and materials from a manufacturer who routinely performs proof-testing of batch samples, rates them accurately, and is willing to publish comprehensive test data, then you can’t go wrong.

You’ll pay more for such parts, assuredly, but the life you save might be your own. 

Handling Forged Parts
Shackle bodies are typically forged, quenched, and tempered, and supplied with an alloy-steel bolt.

While we’re not going to get into metallurgy, suffice it to say that forged steel is very strong, but also rather brittle. Think of it as the opposite of sheet metal, which is highly ductile.

When you drop a shackle – and that’s a pretty easy thing to do – the brittle steel can become invisibly fractured from the impact of the fall, reducing its strength considerably. If you’ve ever seen a shackle explode under load, you’ll recognize the wisdom in discarding a shackle, or other forged part, that may have been inadvertently damaged.

For this reason, always carry numerous spare parts to your gig. You can’t replace the dropped shackle if there’s no others left in your rigging kit. And as “DeLoria’s Shackle Dropping Law” states: After you’ve dropped one shackle, you’re likely to drop another a few minutes later. 

Bridling & Reeving        
“Oh Lord, please don’t let me be misunderstood”—Eric Burden and The Animals

Two of the most misunderstood aspects of rigging are reeving and bridling. I’ll go into further detail about both of them next time, but must state this now: Never, ever reeve a Spanset, wire rope, or other suspension device through eyebolts, shackles, or other supporting points (Figure 4). 

Unfortunately, and for reasons that elude me, such practice is very often seen. There is never any legitimate reason to reeve suspension points when you’re rigging loudspeakers for public usage. Reeving can be useful on construction sites, when the load needs to be shifted to align with some other fixture, but should never be used in the sound reinforcement industry.

A reeved attachment will always significantly increase the load on the parts that are used in the suspension system.

(click to enlarge)

The additional load incurred from a reeved attachment can far exceed the weight of the suspended object itself. This is a simple function of the vector forces involved, which in turn are a function of the geometry of the reeve, but often may be difficult to understand due to the counter-intuitive nature of vector geometry. 

Similarly, bridling, especially at acute angles, will put far more stress on the rigging components than the actual weight of the load itself. This is also counter-intuitive, and perhaps equally difficult to comprehend. Nonetheless it is true. 

In the next installment, I’ll show exactly how reeving and tight-angle bridles magnify the load on the various components that are intended to keep your rig from plummeting to the ground. In the meantime, stay safe out there.

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 of this series here.

See our gallery for more examples of rigging component fatigue, and go to next page for sidebar “Fatigue Properties.”

SIDEBAR: Fatigue Properties
The mechanical properties of steel when a load is repeatedly applied is known as its fatigue strength.

Fatigue testing determines the ability of a material to withstand repeated applications of a load.

The load by itself may be too small to produce a failure.

There are three factors involved when considering fatigue strength:
1) The number of cycles at which a crack initiates;
2) The number of cycles at which the crack starts to grow;
3) The number of cycles at which the fitting fails.

One accepted method of fatigue rating fittings is to test them to one and a half (1-1/2 times the working load limit for 20,000 cycles, without failure.

This standard test is accepted as indicating indefinite life when used within the working load limit under normal circumstances.

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Posted by Keith Clark on 01/17 at 09:17 AM
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Wednesday, January 16, 2013

Ashly And Danley Backbone Of New System At Hope United Methodist

Hope United hired Christian Sound Installations (CSI) to start from scratch, and the integrator designed and installed a first-rate sound system that includes Ashly Audio processing and amplification and Danley Sound Labs loudspeakers, subwoofers and stage monitors.

Like many churches, Hope United Methodist Church in Trinity, Florida limped along for many years with an aging sound reinforcement system. A series of channel failures on the old analog FOH mixer was the straw that finally broke the camels back.

Hope United hired Christian Sound Installations (CSI) to start from scratch, and the integrator designed and installed a first-rate sound system that includes Ashly Audio processing and amplification and Danley Sound Labs loudspeakers, subwoofers, and stage monitors.

“Like most mid-sized churches in the current economic climate, Hope United needed to stick to a strict budget,” said Paul Garner, CSI’s owner and chief designer. “Of course, it’s my job to bring the system in at or under budget, but I’m not willing to do that by sacrificing quality, performance, or reliability.”

Part of CSI’s cost-saving plan for the 450-seat sanctuary was to install Danley loudspeakers with an exceptionally tight pattern control. As a result Garner was able to keep the energy off the walls, provide the vocal intelligibility the church leaders wanted and eliminate the cost of acoustical treatments necessary with conventional loudspeakers. 

Everything in Hope United’s sound reinforcement system is new. The stage has six Sennheiser G3 wireless microphone systems, a ClearSonic drum shield, and an Aviom personal monitoring system for the drummer.

A cutting edge Allen & Heath GLD-series digital console gives the operator the flexibility of a modern digital system with the intuitive feel of a classic analog console. CSI recessed the equipment rack into the wall adjacent to the mixer, and the back of the rack is accessible from a nearby closet. A Tascam hard drive recorder lets the church easily record all of the sermons for easy uploading to the church website.

An Ashly Protea 3.6SP processor provides all input and output conditioning for the new sound system.

“I’ve had great success with the entire Ashly processor line,” said Garner. “The bang for the buck I get with the 3.6SP and 4.8SP can’t be beat. For a straightforward installation, they have everything I need, without heaps of stuff I don’t need. Importantly, the audio path and algorithms are clean and natural sounding.”

In addition to providing crossovers, limiting, and overall EQ, Garner used the Ashly 3.6SP to signal align the house system with the stage monitors.

Two Danley SH-100s are arrayed roughly twelve feet off the center of the church angled slightly down and out, with a single Danley TH-212 subwoofer covering all of the seats. Two Danley SM-100M low-profile loudspeakers serve as ultra-high-fidelity stage monitors. With an EASE model, Garner determined that the Danley pattern control would yield the most even coverage of all the viable alternatives, but hearing a Danley system sealed the deal.

“I brought the client down to my church, where we run a Danley system,” said Garner. “The clarity and impact is so obviously beyond what a conventional system can deliver. They were convinced.”

Three Ashly KLR-2000 amplifiers power the Danley boxes. “The Ashly KLR-Series sounds excellent and has proven to be perfectly reliable,” said Garner. “We’ve put a lot of them in and had zero failures, zero callbacks, and no issues whatsoever. That means a lot in today’s market. It’s nice to deal with companies like Ashly and Danley that recognize that there’s still something to be said for quality while still working hard to keep costs in line.”

Ashly Audio
Danley Sound Labs

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Posted by Keith Clark on 01/16 at 02:28 PM
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Fostex Introduces TH-900 Professional Headphones & HP-A8C DAC/Amplifier

Sold separately but optimized to work together

Fostex has introduced the TH-900 dynamic headphones and the HP-A8C 32-bit DAC/headphone amplifier, sold separately but optimized to work together.

The TH-900 headphones include specially engineered driver unit with a 1.5 tesla magnetic circuit that allows the unit to achieve significantly wider range, as well as the company’s proprietary “Biodyna” 50mm biodynamic diaphragm for purity of sound reproduction.

The headphones are crafted with Japanese Cherry Birch housings, which offer a rigid and extremely dense texture that creates a robust acoustic performance. Style embellishments include housings finished in a traditional Japanese (“Urushi”) lacquer in brilliant Bordeaux red by a 100-year-old artisan group.

The unit’s 3m cord with a gold plated 6.3mm stereo plug is also engineered for maximum durability using 7N grade oxygen free copper (OFC). The TH-900 ships with a headphone stand.

The HP-A8C is a 32-bit Digital Audio Converter and headphone amplifier offers DSD audio playback function, as well as a high-precision electronic volume knob and TCXO clock for Asynchronous mode.

Additional features of the HP-A8C:

• Complies with USB Audio Class 2.0 up to 32bit/192kHz (24 bit on Windows)
• Large capacity toroidal power transformer
• Built-in SD (SDHC) card drive for epoch-making DSD file (DSF format) reproduction in addition
to the firmware update
• Selectable between the internal clock and the external clock (except for USB and SD card)
• Built-in up-sampling function of x2 and x4
• Digital Filter selection between the conventional “sharp roll-off” and the “minimum delay” developed by Asahi Kasei to eliminate pre-echo
• Direct Out mode by passing the volume control circuit
• Variable headphone amplifier’s gain from 0 dB to -12 dB by 0.5 dB step for perfect match with any type of headphones
• Various inputs including USB, AES/EBU, Coaxial, Optical (x2) and analog RCA
• A dedicated Remote Controller is supplied

Style features such as the enclosure with a glass front panel and sleek black side panels and striking accents like the pure white Organic LED display enhance the aesthetic value.

The TH-900 and HP-A8C are separately priced at $1999.99 each and are available now at select authorized US Fostex retailers. Go to American Musican And Sound (here) for a list of retailers.

Fostex
American Music And Sound

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Posted by Keith Clark on 01/16 at 12:47 PM
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Protecting Loudspeakers: Using Limiters To Enhance LF While Keeping Things Under Control

Most loudspeaker systems have a limited excursion capability. The voice coil and cone assembly can travel only so far

While it’s nothing new to place a limiter immediately in front of a power amplifier to keep it from clipping, there is more that can be accomplished with a limiter to squeeze a bit more low-frequency performance out of a loudspeaker while keeping it within its safe operating area.

Most loudspeaker systems have a limited excursion capability.

The voice coil and cone assembly can travel only so far before the motor strength significantly decreases, the suspension (surround and/or spider) reaches its limits or both.

This excursion limit is generally known as Xmax. With knowledge of the Xmax of a given driver and the displacement response of this driver in a given enclosure it is possible use a limiter to provide protection from over excursion.

Most low frequency loudspeaker designs have increasing displacement with decreasing frequency.

This keeps the SPL output reasonably flat down to the cut-off frequency of the system. Below the cut-off frequency the SPL decreases while the displacement continues to increase.

In Figure 1 (below) we see the displacement response for a 15-inch loudspeaker driver in a vented enclosure.

The y-axis is labeled in mPa (millipascals) but the units are really mm (millimeters). EASERA displayed the RMS value of signals so the measurement data have been multiplied by 1.414 so that the display will correspond to the peak displacement of the cone as a function of frequency.

This measurement was made with an input of 4.0 Vrms (5.66 Vpeak). The dip in the curve at 54 Hz is due to the vented enclosure.

In the frequency region of vent tuning the displacement of the driver decreases and will reach a minimum at vent resonance.

For a linear system the displacement response will be directly proportional to the input voltage. This means that the displacement curve can be scaled up or down on the Y-axis as the input voltage is increase or decreased.

For example, we can see that at 100 Hz the displacement is 0.28 mm with an input of 5.66 Vpeak. If the input is 11.31 Vpeak the displacement at 100 Hz will be 0.56 mm.

Figure 1: Peak displacement of a woofer in a vented enclosure (red) and with a 40 Hz high pass filter (blue) with 4.0 Vrms swept sine input. Click to enlarge.

Doubling the voltage doubles the displacement. At relatively high input voltage the system may no longer remain linear due to power compression (voicecoil heating) as well as excursion limitations (both motor and suspension limits).

The former is a thermal issue related solely to the RMS voltage of the input signal. The latter is related to the peak voltage of the input signal.

Hopefully the program input signal to the loudspeaker system during normal use has a sufficiently high crest factor that Vrms and Vpeak are separated by at least a factor of four (12 dB). The measurements detailed here use a swept sine with a crest factor of 1.414 (3 dB).

Measurement preconditioning to warm up the loudspeaker voice coil can help to account for thermal compression at higher input voltages.

A reasonably high voltage test signal can help to minimize the effects of displacement compression when the displacement response is scaled up.

The loudspeaker driver shown in Figure 1 has a manufacturer’s rated Xmax of 5.0 mm. Usually the displacement can be allowed about 15-20 percent beyond the rated value before audible distortion becomes problematic.

For the purposes of this article we will stick to the manufacturer’s rated value. If we scale this 4.0 Vrms (5.66 Vpeak) measurement by a factor of 5 we can see that the 5.0 mm displacement limit will be reached at approximately 21 Hz with an input of 28.3 Vpeak. Assuming a program signal with a crest factor of 12 dB, this equates to 7.1 Vrms.

To put this in perspective, this signal would just be clipping the voltage rails of a 50 W amplifier (into 8 ohms). However, the amp would only be delivering 6.25 W into an 8-ohm load due to the crest factor of the signal.

The point here is that we need to limit the excursion in the low frequency region so we can drive the loudspeaker with higher voltage yielding greater SPL.

It is relatively common practice to use a high pass filter to accomplish this. Applying a second order (12 dB/octave) Butterworth high pass filter at 40 Hz results in the displacement response also shown in Figure 1.

The displacement maximum is now 0.45 mm at 36 Hz for an input of 5.66 Vpeak. This will allow for a maximum of 62.9 Vpeak to reach 5.0 mm at 36 Hz.

In keeping with our reality check perspective above this would just be clipping a 250 W amp. A program signal with a 12 dB crest factor would yield a 15.7 Vrms input (30.9 W into 8 ohms). This is still relatively low.

In order to decrease the displacement in the frequency region below vent tuning so that it is no greater than the displacement maxima above vent tuning (at approximately 70 Hz) requires a second order high pass filter at 50 Hz. This is high enough in frequency to begin to attenuate useable low frequency output from this system.

Another drawback of using this high pass filter is that it will add more phase shift (and group delay) to the output of the loudspeaker. This is a vented design so it already has a fourth order high pass response.

Now let’s get back to limiters. This article is supposed to be about limiters isn’t it?

Figure 2: Limiter schematic showing the side chain signal flow. Click to enlarge.

For the loudspeaker displacement problem detailed above it would really be nice to be able to limit the displacement without using a high pass filter, or at least one set a lot lower in frequency.

But to do this a limiter can’t just limit at a single set level. It must be able to limit lower frequencies at progressively lower levels.

Ideally we would like for the limiter to function so that below some frequency, say 45 Hz or so, the displacement response becomes a flat horizontal line. To do this we must alter the signal that is presented to the limiter’s side chain.

For those readers not familiar, a side chain is the portion of the limiter that computes the gain reduction that needs to occur so that the signal going through the limiter does not exceed the limiters threshold value. This is shown schematically in Figure 2.

Typically the side chain receives the same signal present at the input of the limiter. However, it is possible to send the side chain a different signal if the limiter has an external side chain input. This is sometimes called the key input.

So how do we set all of this up? Let’s say were going to use a 1,000 W amplifier (into 8 ohms) with our loudspeaker system. This amplifier size was chosen so that for a signal with a 12 dB crest factor the maximum output from the amplifier will be 22.4 Vrms (89.4 Vpeak).

The input sensitivity required to fully drive this amplifier with a sine wave is 2.0 Vrms (2.83 Vpeak).

To keep the amplifier from clipping we need to set the peak threshold of our limiter to +9.0 dBV. This corresponds to the peak voltage of the amplifier input sensitivity.

When the amplifier output is at 89.4 Vpeak the loudspeaker displacement response would be close to that shown in Figure 3 if it had no inherent displacement limitation. We can see that the displacement would exceed 5.0 mm below 49 Hz and from 62-88 Hz.

This indicates we have to increase the signal level presented to the limiter side chain in these frequency regions.

The gain computer of the limiter will then see higher signal levels at these frequencies.

This will cause the limiter to reduce the signal level only when these frequencies are high enough to warrant it. The rest of the time the signal is unaltered.

Figure 3: Scaled peak displacement of a woofer in a vented enclosure with 63.2 Vrms swept sine input. Click to enlarge.

The displacement response is now displayed in dBV (corresponding to the RMS voltage output of the LP201 accelerometer preamp) in Figure 4.

We use the RMS value instead of the peak value here because we will be comparing this curve to measurements of some filters.

The EASERA measurements displayed measurements based on RMS values unless otherwise scaled. Since our measurement signal is a sine wave this all works out nicely.

Coincidentally, -10 dBV corresponds almost exactly to the 5.0 mm peak displacement shown in Figure 3. Anything above -10 dBV needs to be limited to -10 dBV in order for the displacement not to exceed 5.0 mm. Several filters are set up to yield the transfer function also shown in Figure 4.

The signal that will be presented to the side chain of the limiter will pass through these filters.

Figure 4: RMS displacement of a woofer in a vented enclosure (blue) and filter transfer function for limiter side chain (red). Click to enlarge.

The signal flow for this is shown in Figure 5 (below). The filters used are a second order shelving filter (low pass and mixer), a first order shelving filter and a 3-band parametric EQ.

 

These filters will increase the level of the signal in a manner corresponding directly to the desired frequency dependent limiting in order to control the woofer’s displacement.

The only thing remaining is to set the threshold of the limiter and measure the displacement of the woofer with the new limiter processing in place.

Since I want to show only the effects of the limiter and not any potential non-linearity in the suspension or motor of the woofer in decreasing the woofer’s displacement, I will not test at 89.4 Vpeak.

I do this also because I like my neighbors, I think they like me (although I may be delusional) and I would like things to stay this way (I think).

So instead I will test at 4.0 Vrms (5.66 Vpeak) and scale the displacement as if it was tested at the higher voltage. To accomplish this all I need to do is adjust the threshold of the limiter so that it limits at 5.66 Vpeak.

This should be done either with the side chain filtering bypassed or at a frequency much higher than would be affected by these filters.

Because we’re now using a limiter, the system under test is no longer linear, time invariant (LTI).

For this reason FFT measurement techniques, such as those in EASERA and other similar measurement systems, can no longer be used to accurately measure this system. Instead, a steady state sine wave at a fixed frequency is used. The results are recorded and plotted in an Excel spreadsheet.

Figure 5: Signal flow schematic and parameters for filters feeding the limiter side chain. Click to enlarge.

Measurement points for different frequencies are at 1/12 octave spacing. The results are shown in Figure 6.

Here we can see that with the limiter bypassed the displacement is almost identical to that shown in Figure 3.

This is expected. With the limiter engaged the displacement does not exceed 5.0 mm except for a small region around 80 Hz.

This is due to the filters feeding the limiter side chain not being an exact match to the displacement curve in this frequency region. This can also be seen in Figure 4.

One of the additional benefits of this type of low frequency excursion limiting, as opposed to a high pass filter, is that boost EQ can be applied at low frequencies if desired.

Figure 6: Scaled peak displacement of a woofer in a vented enclosure with a nominal 63.2 Vrms sine input; limiter bypassed (blue) and limiter engaged (red). Click to enlarge.

At lower signal levels this EQ may subjectively improve the sound of some loudspeaker systems.

The limiter we have implemented will reduce this EQ boost when the signal level is sufficiently high that it would cause the amp to clip or the driver displacement to exceed its recommended range.

In essence, this limiter will remove the EQ momentarily, only at the very highest signal levels containing low frequency energy, and then return the EQ when these high level peaks have decreased below the threshold.

When implemented correctly, this can be very transparent and yield good results.

Many thanks must go to Ray Rayburn, formerly with Peak Audio and currently with K2 Audio, for his valuable insight and patient discussions with a very green engineer many, many years ago on this topic.

Charlie Hughes has worked at Peavey Electronics and Altec Lansing. He currently heads up Excelsior Audio Design & Services; a consultation, design and measurement services company based near Charlotte, NC. Charlie is a member of the AES, ASA, CEA and NSCA. He is an active member of several AES and CEA standards committees.

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Posted by admin on 01/16 at 08:39 AM
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Tuesday, January 15, 2013

Chainsaws & Scalpels: What’s Really Going On With That Equalizer?

Matching EQ types to EQ tasks

In the beginning there were volume knobs, and that was good. Then came bass and treble controls, and that was better still.

But more was wanted, so more was created and thus the equalizer was born. Over time, equalizers have been shaped and honed into the various formats that are ubiquitous today.

Graphic, parametric, semi-parametric, paragraphic, and more – such as Lake’s rather brilliant Mesa filters – entered the market years ago and we can’t seem to get enough of them. At last count, retailer Sweetwater has 70 different models in it’s catalog, with GC Pro offering an even larger portfolio.

How do we sort them out? It starts by matching EQ types to EQ tasks. The use of equalization falls pretty squarely into four primary processes – let’s call them layers – that they’re expected to perform. It’s important to approach these layers in a logical order.

Room. The first layer should always be correcting low-frequency room resonance and any other room-related issues. Usually, this is best accomplished by means of a loudspeaker management tool that includes a generous bank of user-adjustable EQ filters, as most now do. This layer is the foundation that everything else is built upon.

Loudspeakers. The second layer is likely to be far less linear than a console, power amplifier, cables, most microphones, and so on. Even top-shelf loudspeaker systems will rarely be the flattest ruler in the tool box, so this is the layer in which we solve frequency and phase response problems, fix misalignment in the time domain, and integrate the mains with the subwoofers.

Fortunately, modern DSP-based loudspeaker controllers offer a wide selection of corrective tools, enabling precision alignment. Such tools include an extensive range of filter types, incremental delay, protective limiters, and more.

It’s important to note that if a given loudspeaker system is seriously deficient and nowhere near flat, it needs to be addressed first, then the room, then the loudspeaker system one more time. Don’t mistake room resonance for loudspeaker performance problems.

Instruments, Microphones & Vocals. The third layer is tailoring the sound of the vocals and instruments to support the style of music in consideration of what the artist wants to achieve. We add filters to suppress feedback, make that piano, sax, or Sousaphone sound like it’s meant to sound (and/or whatever sound is appropriate for the mix), and shape the overall aural “feel” of the show.

Vocal tonality should flatter the vocalists, instrument tonality should flatter the instrumentalists, and everyone needs enough gain-before-feedback to be clearly heard and understood. If the event includes track playback, then tracks should be given the attention they might need to satisfy the desired production values.

This layer can usually be handled by channel EQ on the console, but outboard devices – graphic or parametric, depending on what you’re comfortable using – can be very useful for taming difficult sources such as podium mics, lavaliers, strings, grand pianos, vibes, xylophones, poorly recorded tracks, and any other source that requires more sonic shaping than the channel strip is able to provide. 

Preferences. When the other layers are firmly in place, you may wish to make tonal changes that are purely preferential at the start of sound check. This may be a one-time thing, or it may vary from moment to moment.

Perhaps the guitarist demands an unusual tonality for a certain solo part that defines his signature sound and needs help to obtain it. Maybe the vocalist performs in a wide range of styles and wants to come across differently from one tune to the next (think David Bowie). Or it might be as simple as occasionally compensating for changes made on stage, such as swapping out a guitar or horn.

In any case, preferential EQ changes should ideally be programmable from console presets, or if that’s not available, then programmable from outboard equalizer(s) so that the baseline EQ can always be found again and reverted to. Preferential EQ should never compromise the system’s overall linearity.

Arbitrary Matters
Someone very clever, a long time ago, figured out that room resonant characteristics tend to fall somewhere around one-third of an octave in width. Not always, but fairly often. This makes it seem like those 31 bands on your graphic equalizer can take us anywhere we want to go in terms of room correction.

But room volumes and reflective characteristics vary considerably from one architectural marvel to another (and from one trashy club to another).

And this is the important part: room resonant modes have not yet formed a committee to agree that they will always resonate at the ISO center frequencies that are the backbone of the graphic equalizer.

While the foregoing is intended as a joke, it’s absolutely true that ISO centers are arbitrary and the room you’re working in is completely ignorant of them.

So what happens when you identify a room resonant mode with a band center of 140 Hz, very narrow in width but high in amplitude, and the only way you can cut it is to adjust the 125 Hz or 160 Hz filter on your graphic equalizer? Well, it might perfunctorily help the room problem…but it probably will not. Instead, it is more likely to exacerbate the issue (Figure 1).

Figure 1: The top graph (A) depicts room resonance centered at 140 Hz. In the middle graph (B), we see the response of a graphic equalizer attenuating 125 Hz and 160 Hz. In the bottom graph (C), it’s clear that the two filters only served to exacerbate the room resonance problem instead of solving it. (click to enlarge)

Think about how a 1/3-octave RTA and a companion 1/3-octave graphic EQ are like audio venetian blinds. Look through a partially closed blind and you see an image, but you don’t see it all. The brain fills in the gaps. It’s easy to tell that the girl outside the tour bus is a girl. But can you precisely see her form, her clothing, all of her features? You’re viewing a limited amount of visual information and your brain does it’s best to fill in what’s missing.

A similar effect occurs when measuring a sound system with a 1/3-octave RTA and then tune the system with a 1/3-octave graphic EQ. Unfortunately, the end result is not going to be the same as looking at the girl through the venetian blind. I can’t emphasize enough the importance of viewing a high resolution response trace (at least one-twelfth octave), and then using an equalizer that has at least the same (or better) resolution, if the results are to be precise.

Room Resonance

Anechoic chambers are built to have almost zero resonant modes because they absorb the sonic energy rather than reflect it, but virtually all other rooms (and for that matter any vessel that contains a volume of air without equivalent absorption) will resonate in response to the air mass that’s being excited by sonic energy. It’s why flutes work, it’s why saxophones work, it’s why pipe organs work…and it’s also why large theatres (and sports arenas) tend to have a lot of acoustic “mud” in the low frequencies.

Resonance is much more pronounced in the low-frequency region than the high-frequency region, except in special cases such as tent structures in which the LF energy is absorbed by the flexibility of the tent walls. In very large rooms (Radio City Music Hall, for example), it’s not uncommon to see LF peaks that are so high in magnitude that it takes ganging two analog filters on top of each to provide the required level of attenuation.

These days, that’s largely become a non-issue because most DSP-based equalizers have a very wide range of cut capability (often -40 dB), but it was a genuine problem in the “good ol’ days” when most analog filters maxed out at -15 dB.

While a well-designed graphic equalizer can be a great tool for shaping the sonic qualities of an individual instrument (I once spent two hours with an 11-band graphic to get a problem snare drum to sound “just right” on an album project), is it also the right tool for tuning a sound system in a resonant room? Numerous professionals make it their first choice. Some believe it’s not even possible to tune a system with a parametric equalizer. Let’s look at this more closely, because some basic engineering concepts can help you make the best choices.

Peaks & Valleys
Resonant problems can actually be improved rather easily, but only if you can accurately identify the amplitude and Q (the ratio of the center frequency divided by the bandwidth) of the resonance.

A high-resolution analyzer is essential for this task. You’re not going to see the precise characteristics of the various resonant modes on a 1/3-octave RTA (real-time analyzer); it’s simply too coarse.

Conversely, an FFT (Fast Fourier Transform) is ideal for measuring the response of a loudspeaker system in a room, as long as it has very fine resolution in the lowest five octaves of the audible spectrum ~20 Hz to ~640 Hz, because that’s where you’ll detect and defeat room resonance.

Of course, a room can’t actually be stopped from resonating with an equalizer alone, but the energy from the loudspeakers can be attenuated at the resonant frequencies. Doing so will radically improve the clarity, intelligibility, and musicality of the event.

Typically, there will be one dominant LF peak, followed by several harmonically related peaks (Figure 2). The frequencies of these peaks are a direct function of how much air volume is in the room. Other peaks, at other frequencies, may also be present in one or more compartmentalized acoustic areas, such as under or over a balcony. These areas need to be examined individually, but only after the main system has been flattened.

Figure 2: The top graph (A) depicts the three primary room resonant modes; note that they are harmonically related. The middle graph (B) shows the response of the parametric filters, accurately adjusted to attenuate the resonant peaks. The bottom graph (C) shows the final result of the correction.(click to enlarge)

Every resonant peak that is identified needs an attenuating filter that precisely cancels-out that peak. The filter should be set dead-on, on the center frequency of the room peak and as narrow in Q as possible. The goal is to surgically cancel the room peaks. Larger rooms usually require four, five, or more filters to remove the primary room resonant frequencies.

Typical amplitudes of room resonant peaks range from just +3 or +4 dB in relatively small rooms to as much as +20 dB (or more) in large rooms. Don’t be afraid to use the full extent of an equalizer’s capability to cancel the peaks. Cutting filters (attenuation) are safe, generally speaking, and will not harm the phase response of the system; in fact, if the amplitude, Q, and band-center are accurately dialed-in, the measurable result will be an improvement in the phase response of the system in those spectral regions where the cutting filters have been applied.

On rare occasions it’s acceptable to use an accentuation filter (boost) to “fill-in” a hole in the system’s response, but this should be done carefully, and with enough time to critically listen to the results with familiar tracks.

In most cases, some aspect of the loudspeaker system’s response, not a room-related issue, is being corrected. Rooms don’t often exhibit acoustic cancellations that need to be boosted, but if the architectural elements are complex enough, or the building materials unusual enough, then it’s possible.

The Loneliest Number
When there is only one measurement microphone available on a project, move it around a lot. If there’s more than one, time can be saved and accuracy improved by instantly comparing one part of the room to another.

In either case, take note of the differences in each room location in respect to others. Don’t adjust EQ for one “sweet spot.” A typical approach is to measure where the console is located, as that’s the quickest and easiest thing to do. But it’s not the optimal approach. Moving the measurement mic around the room, as time permits, will lead to a far better outcome.

It’s vitally important to not use the full system when taking measurements and making EQ corrections. If the room is symmetrical (and most are) start by measuring and EQ’ing only one side of the PA then transfer those filter parameters to the other side (assuming left/right sources).  Otherwise you’re measuring the acoustic addition and subtraction of the various L/R PA elements interacting with each other, rather than identifying the true room resonance.

Of course, after all elements of the system are initially adjusted, listen (and measure) the entire system as a whole. Inevitably the LF region will be more pronounced and dominant than when only measuring one side of the system. This is an excellent time to use an LF shelving filter to gently reduce the LF build-up, rather than to add more attenuation to the surgical attenuation filters.

An LF shelving filter can easily be trimmed on the fly by ear, with no dire consequences to worry about. A narrow Q surgical LF attenuation filter cannot.

Making Magic
Sooner or later, audience members will enter the room, thereby displacing a measure of air volume with their relatively solid human bodies, and thus raising the resonant frequency centers. This is a simple effect of physics: the room now has less air volume so it resonates at higher frequencies, just as a shorter pipe on a pipe organ produces a higher frequency than a longer pipe.

Acoustically, the presence of the patrons will be very noticeable in most rooms, both by ear and by measurement. If you have the means to re-measure the system’s response with the audience in place, then you can make some real magic! Products such as Rational Acoustics Smaart, Meyer Sound SIM, SATlive, and others, foster accurate measurements using walk-in music or even the opening strains of the performance itself.

What you’ll see is that the resonant modes, so carefully identified when the room was empty, have now shifted upward in frequency. Sometimes it’s a lot, others only a few Hz. But keep in mind that a few hertz in the lower frequencies is actually a considerable part of the LF spectrum; e.g., 20 Hz to 40 Hz is a full octave, but that octave contains only 20 actual Hz of LF information (in terms of integers).

Due to the surgical cuts instead of broadband swipes, the audible difference that you (and your audience) will hear when you touch up those precise cuts, causing them to once again fall on the new band-centers, can be astounding. Shifting a filter upwards by a fraction of an octave can have a profound effect? Hearing the results, particularly when the program material comprises natural instruments, makes it obvious that the resultant accuracy is well worth the time and effort it took to get there.

At this point it should be pretty clear that this type of precision tuning can’t be accomplished with anything other than a parametric equalizer, and one that has 1 Hz or 2 Hz of resolution in the LF section. Not all do – some jump around quite a bit in the LF region – so choose your weapon wisely, Mr. Bond.

Ken DeLoria is senior technical editor for Live Sound International and ProSoundWeb has had a diverse career in pro audio over more than 30 years, including being the founder and owner of Apogee Sound.

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Posted by Keith Clark on 01/15 at 12:29 PM
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Martin Audio Appoints Dominic Jacobson As New EMEA Export Manager

Will streamline company’s continuing expansion into global markets

Martin Audio has appointed Dominic Jacobson to the new role of EMEA export account manager in order to streamline the company’s continuing expansion into global markets.

Jacobson will be applying his extensive experience and skills acquired in the film, broadcast TV and music production industries to the live sound and contracting markets, where he will be responsible for the EMEA territories.

“I’m relishing the challenge of promoting a brand that is so highly regarded in each of its core market sectors,” he says. “I am particularly looking forward to growing the business in certain parts of the world such as Africa, where there is scope for furthering Martin Audio’s market share.”

The new export manager will report to director of sales Simon Bull, who confirms, “Dominic’s appointment will provide much-needed assistance through a range of EMEA territories and enable us to apply greater focus to some burgeoning countries.

“He is already well established with audio professionals in many of the territories that he will be visiting and has a strong technical aptitude that makes him well suited to this role.”

Having also assumed a pan-EMEA role as sales manager with Euphonix/Avid, and worked as a music producer and DJ for 17 years, including residencies at Ministry of Sound and Fabric where he had first hand experience with premium Martin Audio systems, Jacobson brings an enviable pedigree to the company.

Martin Audio

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Posted by Keith Clark on 01/15 at 12:11 PM
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Community R-Series Selected For Athletic Hall At Ocean County College In NJ

New distributed system is comprised of 20 R.5COAX two-way loudspeakers

The Athletics Hall and Gymnasium At Ocean County College (New Jersey), a multi-use facility used for games, practices, classes, and other events, was recently treated to a new sound system to replace its aging 1970s era system.

The design-build project was handled by Blackwood, NJ-based JD Sound and Video, and includes Community Professional R-Series loudspeakers and iBOX subwoofers.

“The original system, which was installed in the early 1970s, featured some of Community’s first hand-molded horns,” explains JD Sound’s Joe DiSabatino. “When the college decided to put in a new system, they went around listening to other schools’ systems and were once again convinced that Community had the best solution.”

The distributed system is comprised of 20 R.5COAX two-way loudspeakers, painted black to blend in with the venue’s dark beamed ceiling.

The R.5s are hung in pairs, with one loudspeaker covering a portion of the bleachers and the other directed to the main floor. A pair of iBOX i215LVS subwoofers hung left and right along a central beam provide powerful bottom end.

The system is powered by Crown Audio CTS and CDI amplification, with a Rane RPM2 DSP unit providing system drive and processing.

“The room has a folding wall that can divide it in half,” says DiSabatino, “and the system can be configured to route any sound source to either side of the system for maximum flexibility.”

While the project itself was relatively straightforward, the time-frame was anything but, says DiSabatino. “Once they accepted our bid, we had a two week window before classes resumed to get the entire system installed. Community had just introduced the R.5 in black, and they were still manufacturing some of the cabinets as we began the installation.

“Then we had to evacuate one day due to a minor earthquake, and another day due to a water main break. We were literally fighting the clock, and it was then that we really appreciated how easy the system was to install and set up.”

Despite those minor obstacles, DiSabatino reports that the system has been performing even better than anticipated. “They recently hosted a formal gala in the room, completely transforming it into a nice looking ballroom. The system sounded great, and coverage was just as good at low levels as it does pumping well over 90 dB for the games and pep rallies.”

Community Professional

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Posted by Keith Clark on 01/15 at 11:06 AM
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Monday, January 14, 2013

Argosy Debuts New Line Of Spire Speaker Stands For Studio Monitors

Safe, sturdy solution for monitor isolation

Argosy Console has debuted the new line of Bioacoustics-enhanced Spire Speaker Stands, which provide an attractive and stable solution for studio monitor isolation.

Two models are offered in the line—the Spire 420i with a height of 42 inches, and the Spire 360i, which is 36 inches tall.

The Spire Stands, which are comprised of 5/4-inch powder-coated substrate for stability and resilience, offer a modern design with die-cut side panels and beveled edges to present high-tech look and feel.

The patented Bioacoustics technology is designed to keep all movement on-axis to move in the direction of the loudspeaker cones’ travel while resisting movement in other directions. The result is that when any monitor is placed on a Spire 420i or 360i, there is a tightening of the low end as well as improved stereo imaging.

Dimensions of both Spire Stands are 18 x 18 inches at the base, 7.75 x 9.9 inches at the top, and again, with heights of 36 inches (360i) and 42 inches (420i). The Spire 360i is available for $379 per pair and the 420i is available for $399 per pair.

The new Spire Speaker Stands will be on display at the upcoming Winter NAMM show at both the Bioacoustics Booth (Hall E, 1631) as well as the Waves Booth (Hall A, 6824.)

Argosy

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Posted by Keith Clark on 01/14 at 06:05 PM
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Celestion Unveiling New 18-Inch Cast-Frame LF Drivers At Upcoming Winter NAMM

Extends next generation CF range of high-end ferrite magnet drivers

Celestion will be unveiling new additions to its next-generation CF Series of high-end ferrite magnet drivers at the upcoming Winter NAMM show in Anaheim (booth 4676).

The new 18-inch, cast aluminum frame, ferrite magnet, low-frequency drivers are well-suited to use in high-end, professional sound reinforcement.

The CF1830E is designed to be a multi-purpose loudspeaker driver for reflex, scoop bass or horn-loaded subwoofers as well as a bass unit in a large multi-way system. The CF1830E hass a 3-inch/75mm multi-layer voice coil and delivers 700 watts RMS (AES standard) power handling and 94 dB sensitivity.

This model also incorporates Celestion’s DMM (Dual Magnet Motor) technology, where a secondary magnet is utilized to increase overall motor force (Bl) without the need for any increase in the size (and more importantly weight) of the magnet assembly.

The new CF1840JD is designed for reflex subwoofer applications, with a 4-inch/100mm multi-layer voice coil. It delivers 1,000 watts RMS (AES standard) and 96 dB sensitivity. Twin demodulation rings reduce flux modulation, further lowering electromagnetic distortion.

Both models feature BAV (Balanced Airflow Venting) to provide enhanced magnet assembly cooling, further enhancing performance by minimizing thermal compression.

All CF Series loudspeakers are designed using specialist FEA (Finite Element Analysis) modelling techniques, enabling Celestion’s development team to rapidly achieve genuine increases in performance, such as a highly symmetrical cone movement. This results in exceptionally low harmonic distortion and a more efficient performance.

Developed at Celestion’s headquarters in Ipswich, England, The CF series incorporates a unique and powerful industrial design, created in partnership with Allen Design Associates. More than a simple cosmetic makeover, this aesthetic is intrinsic to the performance of the loudspeaker itself.

Celestion

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Posted by Keith Clark on 01/14 at 12:41 PM
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Andy Flint Named Senior Manager, Portable PA Marketing For JBL Professional

Will lead the development of new marketing and product strategies for this segment

Andy Flint has been named senior manager, Portable PA market for Harman Professional’s Loudspeaker Group and its JBL Professional products, and will lead the development of new marketing and product strategies for this segment.

He will report directly to Mark Gander, head of marketing for the Loudspeaker Group and its JBL Professional brand.

Flint had previously been with the Harman Pro Amplifier division for the past eight years, where he was responsible for managing product strategy, pricing structures, brand image and large account relationships for the professional audio retail market, and he played an integral role with Crown Audio.

“Andy’s extensive portable PA experience and strong industry relationships make him an ideal match for the role of Senior Manager, Portable PA Marketing,” Gander says. “We are confident that he will be a tremendous asset as he takes on this new position.”

“I’m looking forward to the prospect of working with a group of talented professionals in the Loudspeaker Group and am equally excited to promote JBL Pro, the industry’s leading range of loudspeakers in the portable PA category,” Flint adds.

He holds a BA in Marketing and an MBA in International Management and Marketing from the University of Wisconsin.

Harman Professional
JBL Professional

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Posted by Keith Clark on 01/14 at 12:19 PM
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Friday, January 11, 2013

High Standards: Concert Sound For Joe Jackson And The Bigger Band

Audio approaches for a classic artist going in a new direction with a large ensemble

Joe Jackson is best known to many for a string of breakthrough pop-rock/New Wave hits that began with 1979’s “Is She Really Going Out With Him?” but the British singer, songwriter, and composer (and five-time Grammy Award nominee) has explored a variety of musical styles over his stellar career.

His latest tour is no exception; a series of live shows in the U.S. and Europe in late 2012 featured both reinterpretations of his own material and selections from his latest record, a tribute to jazz great Duke Ellington that’s aptly titled “The Duke.” Jackson performed with his largest band to date, a seven-piece ensemble appropriately named The Bigger Band, featuring jazz violinist Regina Carter.

Ellington’s own (sometimes radical) reinterpretations of his own music heavily informed Jackson’s unconventional treatment of the material on record and on stage, using instruments including banjo, tuba, synthesizers and accordion rather than more traditional big band instrumentation. Doing so was a challenge, which Jackson notes in a recent bio: “When I started this, it felt daunting – like, how am I gonna pull this off?”

When monitor engineer Paul Froula first heard the recording, he echoed that sentiment. “Before pre-production I was thinking, ‘How much of this are we wanting to reproduce live?’ But it was a nice challenge.”

The three-week rehearsal process provided the opportunity to pull things together and work out the kinks. One highlight, Froula notes, was working with a mixture of musicians who were comfortable with in-ear monitors and others who were using them for the first time. “It was a nice change, as opposed to a band that’s completely accustomed to in-ears.”

Joe Jackson and The Bigger Band performing on the recent tour. All performers on IEM made for clean stage. (click to enlarge)

Simple Integration
One of the primary factors driving the design of both monitor and house systems was size and scalability, with the tour stops at venues ranging from large theatres to smaller clubs. “Essentially we had to fit everything we carried into a trailer behind a bus,” notes house mix engineer George Cowan.

Froula based his monitor system approach on a Roland V-Mixer M-480 console. While a variety of smaller format digital consoles have hit the market of late, he’s most familiar with the M-480, having used it recently with Todd Rundgren and others.

“We carried backline, band gear and our own FOH/monitor consoles on this tour,” he notes. “I’ll use this desk where space is at a premium and performance can’t be compromised. I’m comfortable with the workflow, and, considering the number of players and mixes I had to manage, things needed to be where I wanted them on the control surface.

Paul Froula with the Roland M-480 console that proved a great fit for his work on monitors. (click to enlarge)

“Items like pulling up EQs, effects and my defined presets demanded a well-designed layout,” Froula continues. “The desk has an onboard RTA so I could isolate a few things visually – particularly Joe’s vocals. He has a fantastic ear, and it really helped to be able to work with George out front and his (Rational Acoustics) Smaart rig to get the sound he wanted in his ears.”

The ability to control the M-480 remotely with an iPad also came in handy, particularly for optimizing microphone positions and gain settings on band members Sue Hadjopoulos’ percussion and Nate Smith’s drum rigs.

“They’re next to each other on the same riser, separated by a Clearsonics drum shield, and between them there are 17 open microphones, so there was a lot of bleed potential. Not everyone has the iPad control app down, but the Roland app has almost every feature I needed, so it allowed me to be in two places at once.

“Standing at the congas monitoring a mix in the IEMs while simultaneously making console adjustments was a real time saver. Plus, I can tweak mixes while beside band members for those who like to be involved in that way.”

The M-480 was paired with the Roland S-3208 and S-1608 modular digital snakes to meet a 48 input and output requirement.

“The snake heads are part of the REAC networking system that drive the heart of the A/D conversion. It all combines for very simple integration, and sounds great. I really wouldn’t have it any other way for this situation.” Froula says.

He also recorded each show to a Roland R-1000 48-track recorder that lived handily on the REAC network.

“It plugs right into the console’s REAC bus,” he explains, “allowing me to multi-track 48 channels with two Cat-5 cables and a hard drive, providing for easy DAW integration as well as the ability to do virtual sound checks, if necessary.”

Studio Influences
In keeping his front-of-house footprint compact, Cowan handled all processing using a combination of the onboard capabilities offered by his Yamaha DM2000VCM console and a variety of Universal Audio plug-ins hosted in Pro Tools 9 on his MacBook Pro.

His choice of the DM2000VCM was “a no-brainer,” he says, something his production company, A Major Productions, had on hand and that allowed him to travel with very little in the way of external processing. “It’s well equipped with plugins, and the Waves 96 DSP Card provides additional choices for compression, EQ and de-ssing, but, primarily, I’m using Universal Audio plug-ins for reverb and delay,” he says.

Front of house engineer George Cowan at his Yamaha DM2000VCM console. (click to enlarge)

This preference is informed by his long-time work as a recording engineer, and including at upstate New York’s Bearsville Studios. Since taking on live duties for artists like Natalie Merchant, and later, in 2003, with Jackson, he now splits his time between recording and live sound.

“UAD products have always been counted among the list of top gear, especially for studio recording,” Cowan explains, “but I wouldn’t put their hardware in a rack on the truck every night. Now they’ve moved into the digital realm so I can take the tools that I’m used to, and that do what I expect of them, and put them in my live arsenal.

“I park the UAD plug-ins on Pro Tools channels, put the channels in record ready so I have a through port and digital I/O between the console and the laptop using an M-Audio Lightbridge. It’s a lot like setting up a rack. You don’t see many people doing this. It’s not a typical way of using Pro Tools, but has proven to be stable.”

A look at the miking approach for Nate Smith’s kit. (click to enlarge)

Typically at each venue, Cowan requested a high-quality stereo, 3 or 4-way main loudspeaker system with commensurate subwoofers, controlled by a configurable, digital electronic crossover/loudspeaker management system.

The goal was consistently delivery of sound pressure levels of 112 dB, A-weighted, throughout each venue.

“Essentially, I start from the top down, providing a list of acceptable PA systems,” he adds, noting that top-flight brands including d&b audiotechnik, Meyer Sound, L-Acoustics,
Martin Audio, Adamson Systems, NEXO, Electro-Voice and JBL all make his cut.

Still, he’s not locked into using one manufacturer’s product exclusively over another: “It depends on what’s available, what the venue’s needs are and if additional loudspeakers are required for over/under balcony support. I’m not going to demand a specific system if they already have an adequate one.”

Making Decisions
Jackson prefers a clean, quiet stage – all in-ears, no wedges.

In response, Froula deployed beyerdynamic IMS-900 RX/TE-9000 TX RF systems feeding Future Sonics MG6 Pro in-ear monitors, a choice based on their armature-free single driver design.

“That makes a big difference in delivering natural sounding mids and highs, along with huge low-end reproduction,” he states.

While Allison Cornell (keyboards/viola/banjo/multilingual vocals) and Hadjopoulos – both veterans of previous Jackson tours – and Jackson himself knew exactly what they wanted, there was a little more interpretation going on with the rest of the band.

“They weren’t as familiar with what they wanted, versus what they needed, versus what they thought they wanted,” Froula says, “but once they started performing, the in-ears became a part of who they were on stage, which is the way it should be. And Joe, he was looking over the band smiling, pleased that everyone was doing what came naturally and the improvisation breaks kept flowing.”

In addition to percussion, drums, keyboards, violin, viola and Adam Rogers’ electric and acoustic guitars, Jesse Murphy plays electric and upright bass and Tuba, Jackson plays accordion and Cornell plays banjo. Ensuring the band could do what came naturally, and that the eclectic mix of instruments sounded natural both in the player’s IEMs and in the house, played a part in the choice of microphones.

Radial JDI boxes provided the stereo keyboard feeds, with mics on violin and banjo. (click to enlarge)

Froula’s predominant selections are from beyerdynamic. “As a rule, for drums, I go with Opus models because they sound really good and have an innovative mounting system that clamps on drums very nicely and allows for a full-sized XLR coming off the barrel connector. They’re clean, reliable, and I’ve used them on drums for the better part of 10 years.”

Both drums and percussion received similar treatment: Opus 88s and 87s on snare, toms, timbale and bongos, and beyerdynamic MC 930s for hi-hat, overheads and various “toys and tinkles.” Cowan went with a Sennheiser e 602 on Nate Smith’s kick – part of a vintage kit that has it’s own peculiarities, Froula notes. Additionally, a Shure SM98 was applied for Hadjopoulos’ glockenspiel.

More beyerdynamic could be found on other instruments: Opus 62s on banjo and on Jackson’s accordion, TG-X 60s for backing vocals and MCE 86s for house left/right ambient mics. Violin was handed with a combination of an MCE 10 clip-on mic and an LR Baggs direct box, but the viola was LR Baggs direct only.

“Allison’s choice,” Froula explains. “The way she configures the DI sounds great. As far as tone and EQ, she’s got it down.”

beyerdynamic Opus 87 and 88 mics fit handily on percussion, with a Shure SM98 for the glockenspiel. (click to enlarge)

Radial JDIs handled stereo keyboard mixes and, in tandem with an internal Fishman pickup, acoustic guitar. The only amp in evidence – for electric guitar – was offstage in an isolation box and miked with a Sennheiser e 609 over the dome for the house and a Royer R-121 on the right edge of the cone for monitors.

Between electric/upright bass and tuba, Murphy had a lot going on, Froula notes. “On electric we used a TabFunkenwerk DI, and on upright a JDI and an Audio-Technica ATM350. On tuba, there’s a certain punch to brass at that frequency some mics don’t handle well. We tried a bunch of things and the last – probably the 10th mic we checked out – was a Sennheiser e 904 gaff-taped down the bell on a piece of foam. It wasn’t pretty, but it worked and was a method previously proved by George.”

Enjoyable & Engaging
In each case, it’s about choosing the right microphone for the instrument; an approach Froula and Cowan both adhere to. “I think that comes from my recording background,” Cowan says. “Whatever sounds good on a particular instrument on a particular day, we move forward with that.”

Nowhere is that approach more evident than when it’s applied to the most important instrument on stage, Jackson’s voice. “We’ve tried a number of microphones,” Cowan explains, noting that the Shure KSM9 has provided the answer. Specifically, it treats the mid range tone of Jackson’s voice very well and allows both engineers to reproduce his signature tone while applying very little in terms of EQ.

The approach also fits in perfectly with Jackson’s approach to his music. Like Duke Ellington, Jackson is known for taking pleasure in showcasing the individual voices of the musicians he works with, but also for recognizing the talents and art of the technicians who strive to reproduce those voices accurately, both for the band on stage and for their audiences.

“Joe spends a good deal of time rehearsing to get it right,” Cowan concludes. “He has high standards and so we generally have time to work out the kinks before we all get thrust into the limelight. He gives us the time to experiment, and that’s awesome. We truly appreciate working with him because it’s always enjoyable and engaging, night after night.”

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

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Posted by Keith Clark on 01/11 at 02:48 PM
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RE/P Files: clubs, Clubs, CLUBS: A Sound Reinforcement Supplement

A thorough look at the world of club sound circa 1974

Editor’s Note: From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature is a serious look at club sound in RE/P’s Concert Sound Reinforcement supplement. The article dates back to the January/February 1974 issue.

“Multi-Various” is a phrase we have invented to describe the spectrum of challenges, problems and responsibilities the sound man is continually exposed to in working to achieve the best possible audio environment in club-type establishments.

The obvious grammatical redundancy of the phrase is intended to emphasize what may well be the most demanding job in audio. “Multi-Various” equally describes the nature of the equipment.

Many clubs have their own, built-in sound system, others rely on the musicians to bring their own sound gear, while still others have a favored sound service contractor who is familiar with the club’s requirements.

The wide variety of club type operations has a great deal to do with the measure of effectiveness of various types of audio systems, permanent or portable systems, which are intended to be used in them.

Some of these establishments might best be called theatre-clubs, while others are the traditional night-club types.

Some are simply bars, and still others of the same sort, removing the alcoholic connotation, are coffee houses.

The conveyance of entertainment in one form or another, the owners’ profit margin notwithstanding, is the sole purpose of any of these types of establishment. Everything should play in second place to this entertainment consideration.

Additionally, for the purposes of this article, all clubs have a few other things in common; small audience, with seating for everyone (10-400), intimate atmosphere, probably some kind of food and beverage service during the performance.

Dance floors are common, and are often a special problem. But, without a doubt, regardless of the type of club, one thing a soundman can be absolutely certain of is that there will not be enough room or tolerance allowed for a proper audio set-up.

Specifically, the audio environment and equipment in the finer clubs often rivals that found in small, high quality recording studios.

A few have a booth in the audience, or set aside, with multiple input and output mixing consoles that control as many as 30 microphones, and banks of bi- or tri-amplified speakers.

Some of the better systems installed in larger rooms employ multiple banks of speakers distributed around the club, driven through delay lines to equalize the transit time so that the direct sound from the stage arrives at nearly the same time as the sound from a nearby speaker.

At the other end of the spectrum, there are jukebox and muzak rejects, leftovers from someone’s discarded hi-fi junk.

There is also a broad range of equipment intended primarily for only moderate to high level reproduction of music, or for reinforcement of speech only. These systems rarely do justice to the cost or effort put into their installation.

Acoustic Considerations

The Room:
The acoustic of the room is at the same time the most important factor to be considered, and unfortunately, is the least likely to receive much forethought and attention.

Typically, to produce a feeling of warmth and intimacy, many clubs install quite low ceilings, especially, for some unknown reason, in the stage area. Rooms tend to be small enough and live enough to have serious resonance modes in the fundamental vowel range (250 100011/.).

Sound reflections off such objects as hard wall panels and large- glass areas wreak havoc with the reverberant response of the room.

Another acoustic problem area is the speaker-to-front-seat / speaker-to-rear-seat distance ratio of the room, which is often so imbalanccd that it is extremely difficult to distribute direct sound evenly. Exposed low ceiling beams can compound the problem.

For aesthetic design reasons there may not be very much the soundman can do to alter the resonance and reverberation conditions that exist in the room. However, where the management is negotiable on these points, the empirical and tactical use of drapes, carpets and curtains will certainly improve these acoustic characteristics.

Even though it is all too obvious, we should all be reminded that a full house (versus a room without an audience) also helps to reduce room resonances… in addition to its beneficial effect on the proprietor’s financial problems.

In dealing with the uneven sound distribution problem it has been found that most high frequency speakers suitable for club applications are short throw devices, quite different from those used in large concert halls or outdoors.

In order to optimize the pattern of these short-throw speakers and overcome the direct sound distribution problem, the high frequency speakers should always be located as high up as possible, and aimed or angled downward toward the audience.

The Stage:
The stage acoustic seldom, if ever, gets any attention. It has however, the governing effect on the performer’s sense of tonal balance, particularly in the middle and high frequency range with amplified rock bands.

It has often been the case that the high frequency energy from the percussion will bounce around the stage-area, causing a ringing effect. This multiple reflection, when uncontrolled, causes intense sound pressure levels in the higher end of the audio spectrum, which can and does cause hearing threshold shift.

A cycle of events then develops where the soundman and musicians not hearing enough highs boosts the high frequency equalization, this in turn causes further acoustic trauma, with the resultant at tempt to boost the highs even further.

Threshold shift is another way of saying that the ear and brain become desensitized to a sound. It is the result of fatigue, especially after exposure to high frequency, high energy sound, where the shift causes a loss of perceived highs.

Threshold shift is usually temporary, although with prolonged or extreme exposure it can be permanent. Tonal balance is judged from what you hear: as a sound mixer you may be in a club working a loud rock and roll act for some time and generally you detect that the bass is muddy and there aren’t enough highs so you boost the EQ and the level until it sounds right again.

Actually the problem may have been simply that your hearing over a period of time has lost the highs, due to threshold shift. When someone walks in from the street, he may perceive the sound as terribly shrill. His hearing hasn’t been affected yet.

But if he stays around for a couple of hours, his hearing may be just as affected as yours. After working one of these shows, as you drive home you sometimes can’t hear the engine in your car quite right. The motor may sound funny because you can’t hear the highs.

High frequency threshold shift can affect the audience as well as the musicians and the soundman.

The instrument amplifier speakers are often mounted at the musicians’ hip-height. This beams the highs straight out.

So while the musicians don’t get exposed to the highs, the sound is aimed right at the audience. However, the musicians do get the highs from the drums because the sound, unless confined somehow, is washing all around the stage area: Cymbals, for example, will reflect off the floor, walls, and ceiling.

These reflections must be controlled in order to provide the optimum performing atmosphere on the stage.

A method for selective absorption of low frequencies and an extension of the principle to broad band absorption is the slat absorber shown in fig’s 1a and 1b. These are practical renditions of the Helmholtz resonator.

The absorber consists of resonant chambers (space between slats) connected to an air mass spring (the volume behind the slats). The absorption spectrum may be broadened by the addition of damping material between the two as shown.

The use of acoustic rather than electronic “equalization” is necessary because the reinforcement system has no effect on the local acoustic response of the environment to the musicians and their instruments.

Figure 1.

What can be done in most cases is to use combinations of reflection, diffusion, and absorption, arranged to focus and control the drum sound. Very helpful is a reflector panel behind the drums which can be fashioned from 3’ x 6’ x 1/4” thick sections of plexiglass, joined with piano hinges, and fitted with edge trim.

Placing the drums on a riser also helps, but practically speaking, you can’t stuff the drummer in the rafters, and even if you can, most drummers won’t stand for it.

Large instrument and equipment cases, instead of being the usual pain in the neck to store until moving on to the next gig, can be used strategically to break up any plane surfaces in the rear of the stage.

This increases the low and middle frequency absorption coefficient. Artfully covering a stack of them with drapes or curtains will help break up the highs as well. Just as is done in the recording studio it is sometimes advantageous to place a large beach umbrella, or some similar semi-absorbent surface, above the percussion instruments.

Normally much of the sound goes straight up into the ceiling and then returns within milliseconds, its spectrum altered by the complex attenuations of the transit and reflection.

Reflections of drum and percussion sounds contribute to poor sound quality as well as threshold shifts, not only among the musicians on stage, but also in the audience, since the mikes pick up the reflections. The umbrella or other semi-absorptive material above the percussion cures both problems.

Otherwise in the vicinity of a microphone element, the reflected ceiling sound will meet sound reflected from the floor. If the sound is approximately equal from both reflections, an addition or cancellation may occur at various frequencies, depending upon the relative phase relationships. The result will be unpredictable.

Speakers
Speaker selection and placement is, singularly, the most distinctive feature of any given sound system.

The two most popular styles are: combinations of horn-loaded direct radiators and compression drivers, and line arrays of direct radiator cone loudspeakers.

If one has decided to go the horn route, it is very important to remember the facts of life concerning horns: Any horn’s capability to deliver clean sound at rated power levels decreases at least 18 dB per octave below the manufacturer’s recommended crossover frequency.

Horn choice should be limited to those made from composite materials, such as; combinations of fiberglass and plastic, fiberglass and plywood, or non- homogeneous fiberglass.

Generally, horns should not be boxed, although this is sometimes done to facilitate carrying and stacking. However, if horns must be boxed, it is advisable to fill the box with foam or fiberglass to prevent any sound which leaks through the back of the horn from reflecting around the inside of the box causing extraneous noise and distortion.

Dense foam or fiberglass will also protect the horns from breaking loose if a box is accidentally dropped. If the horn is to stand free, it is good practice to coat the rear facing surfaces with auto undercoat, carpet, or even machinery rubber.

This will reduce rear sound radiation as well as adding mass to the horn material. Mass will reduce the tendency for the horn to ring. It will also help to reduce the tendency of the horn material to propagate a waveform faster than the air in the horn, which can cause a position smear in the output of the horn, and non-linear frequency response. Speed of sound in aluminum is 10 times the speed of sound in free air.

Figure 2.

Always attempt to keep all the driver elements covering a given sector in close proximity to one another. The higher the frequency range, the closer they should be kept together. This same point is valid for high frequency direct-radiator arrays. Both systems have equal difficulty in achieving the desired effect, and compromises in distortion are involved with either system.

The difference in direct-radiator and horn loaded sound lies in the spectral distributions of the distortion products. In other words, everything has some distortion, but the frequencies involved may vary.

Choice of speakers then becomes a matter of personal taste, at least partially. Some people prefer the “punch” they get with horns. Others feel that horns have a lot of distortion, and they prefer the “soft” sound a cone produces.

Practically speaking, the difference in cost, size, weight, and efficiency, as well as maintenance, must be weighed against the requirements of the specific application. A permanent installation has it all over the road system in this category.

Speakers are usually located on both sides of the stage, at or slightly above the stage level. Ideally, such units should be kept well off the floor for two reasons. One, people get in the way. And two, the previously stated problem of distributing the sound equally to the front and the rear seats.

Particularly where the high frequency units are too close to the audience in front, a vastly different tonal balance will exist in that area. Sometimes the speaker system can be divided into short throw and longer throw sectors, each radiating primarily into its own section of the audience.

Projecting the highs to the rear is fairly easy. But throwing high bass (400-800Hz) to the rear, especially in a club where the floor slopes up sharply, as in theatre clubs, requires a horn that is angled upward.

Accurate and carefully placed sound is very critical if there is a dance floor over which the performers must relate well to their audience: If a horn system is to be considered here, it should have an acoustic lens for the near-stage area so that the edges of the horn pattern fall off gradually II more than one horn is to cover an area, they should be positioned as shown in figure 2. A suggestion for some effective direct-radiator arrays is illustrated in figure 3 (Page 5).

The configurations just described aid in obtaining phase coherence and better pattern control (no hot spots) throughout the driver’s range. This results in better intelligibility and localization because there is smoother direct and reverberant response. Another way to enhance localization is to suspend a speaker over the performers, and to direct it at the audience.

The position of the mixing console within the sound field should provide the mixer with virtually the same blend of direct and reverberant sound as that part of the audience which he most wishes to please.

Power
To make an estimate of the current which a particular set-up requires, we add the AC fuse capacities of the amplifiers, and then we throw in another 25% to 33% for a margin of overload protection.

Relatively few amplifiers have good power line regulation. Unfortunately the same goes for mixers, as well as for other ancillary equipment.

This sometimes causes problems which can be baffling if you don’t realize what is happening: There are times when a high gain mixer will interact with the amplifiers through the power mains, causing a low frequency motorboat type oscillation.

This problem can be solved by connecting the mixer to a separate circuit, preferably on the opposite side of the AC neutral when the power is split-neutral 220V AC, single phase. This method of delivering power to light and plug panels is common, although sometimes power will come only via a single 110V circuit.

The soundman then might have to experiment to find an outlet in the house that sufficiently isolates the mixer from fluctuations in voltage, due to power amplifier current draw.

By far, the recommended way of getting power to the mixer is from a well-regulated split-neutral distribution system, (described in “The Concert Soundman” article which appeared in the November/December 1973 issue of this magazine).

Any such power system should obviously be constructed to comply with the local industrial electrical codes, and should include a built in tester. We have seen, on occasion, some clubs with three- wire “delta-connected” mains, (220-250V AC), to run refrigeration and air conditioning units.

Figure 3.

This voltage can cause instant total destruction of solid state equipment in the interval it takes for the line fuse to blow.

These mains are easily identified because all three wires are fused or switched simultaneously: Split- neutral single phase 220V AC normally fuses the two “hot” legs with the neutral going directly to the service meter, in which case the neutral is often bonded to ground.

This ground, while it can be used for an audio circuit earth ground, is not preferred to a separate, high capacity current path to earth, such as cold water or drain pipes. Beware of PVC plastic pipe, which is becoming more and more- popular in buildings.

PVC will not provide a ground. Watch for situations where metal pipes in the building to which you may have attached a ground may either be interrupted, or may terminate outside the building, above ground, in PVC.

Microphone Considerations
Each performer or group has its own setup. Most tend to think that what they do in the recording studio will work in a reinforcement situation. This is only somewhat true.

It must be remembered that live performances are the most difficult to record, and for the same reasons reinforcement is always difficult and frustrating to engineers.

Some guidelines can be established with regard to staging and microphone technique in general. Most specific comments that can be made are along the lines of improving the working conditions and increasing the amount of confidence, as well as relieving a good deal of confusion.

Experiments should be conducted to determine the exact microphone and the technique to be used in each case. Using as few mikes as possible is desirable because each time the number of “open” mikes is doubled, the acoustic feedback threshold is lowered by about 6dB.

A very real consideration here is that some performers really feel naked without a lot of mikes. If such is the case, as sneaky as it sounds, it may even be helpful to set up mikes that are not connected to the mixer.

The pattern and leakage for each microphone should be considered in terms of what is being done with the microphone.

Wherever possible, omnidirectional microphones are advisable because of their generally superior amplitude and phase response. Amplitude response is the parameter commonly referred to as the frequency response.

Amplitude is one kind of response relative to frequency. Phase is another kind of frequency response. Non linear phase response can be analogous to time delay or impulse response (risetime).

For vocal microphones, or other applications involving unusually high gain, a differential microphone pair is appropriate. When contemplating the purchase of omni microphones for this purpose, be wary.

Peruse the manufacturer’s data sheet to determine that the microphone is a pressure transducer with flat response for both near and far sound sources. This should not be confused with a pressure gradient microphone, which responds to the difference in pressure between two diaphragms, and which is more sensitive to proximity effects.

However, it is important to remember, as has been discussed by Jim Coc in his article “Differential Microphones” in the May/June 1973 issue of Re/p, that not all performances are able to accommodate themselves to the properties of differential microphone operation.

Some popular omnis have acoustic or electrical equalization in them to obtain a response that the manufacturer judges is popular with engineers.

Unidirectional microphones exhibit ragged off-axis response, often with harmonically related peaks and dips. Your ear may perceive this as being flat, but it doesn’t help the feedback threshold. Since much more of inter-microphone isolation is due to proximity than to the directionality of the microphone, little is to be lost and much is to be gained with omnis.

As a rule of thumb, if you get one mike 4 times closer to a sound source than another mike, the other mike essentially isn’t there. An additional advantage is that omni microphones are less expensive and more rugged than comparable directional microphones. This is an important consideration in the real world of sound reinforcement.

Percussion instruments sometimes require a distant, directional microphone technique. Only the highest quality dynamic microphone should be considered here. Condensers tend to clip, and ribbons are too delicate.

So dynamics do the best job, providing long life and good sound quality. In reinforcement work drum overhead microphones seem to be more trouble than they are worth. It can be assumed that there is excessive cymbal sound already reflecting off the heads of the drums, sound that overhead mikes will pick up disproportionately.

The high-hat cymbals can be picked up by a non- directional or a bi-directional microphone. The snare drum can be assigned to the same mike by placing the mike somewhere between them in a position which gives the best and desired balance.

Staging
Set changes are always traumatic experiences if they are not carefully planned. The results of a poor set change can foul up the sound of the entire following set. Disorganized microphone setups and cabling can trip up the performers, causing both distraction and anger.

A valuable suggestion to begin with is to use one master multi-pair snake for routing all microphone lines from some central location on the stage to the mixer. Smaller, multi-pair subsnakes for drums and for vocals will also reduce the effort and confusion.

Immediately after the set, all the microphones used for that set should be unplugged and gotten out of the way of stagehands and equipment men. The mikes can be removed to the side or front of the stage, and are then replaced when the next setup is complete: This job should be assigned to one man.

Most club stages are small, and any increase in the freedom of movement will be appreciated. Also, it is remarkable how many microphones are saved from the need to be shipped back to the factory for complete repair as a result of having been knocked over, or trampled on during set changes. This little extra seems well worth the effort.

Monitors
Until recently there was one, and only one, way to provide monitor foldback on stage.

Floor mounted, slant-back speaker enclosures were placed on the floor in front of the performer, and were angled so as to aim in the vicinity of the head. However, since this locality was also covered by the rear lobes of the performer’s unidirectional microphone, this meant that feedback could and often does become a problem.

Newer, more effective monitoring involves far-field canceling microphones (differential microphones) and arrays of speakers spaced about the stage near ear level.

Far-field canceling mikes are not always desirable, however, because some performers can’t get accustomed to the associated close mike technique that is necessary, as commented upon previously.

In small enclosed spaces, such as a small club environment, the reverberant field of the audience speaker system con tributes greatly to the sound in the stage area.

This contribution, mainly in the vocal vowel range (250-1000Hz), combines with the monitor sound field and the cavity resonances of the stage.

The interaction of the two systems can cause operator confusion concerning feedback. Monitor feedback must receive the “killer of the ear award.” Monitors tend to feed back at high frequencies and high intensities. The resultant accelerations and loads placed on the ear structure are painfully dangerous.

Normally, the system will start to ring and your ears detect a hollow sound before full feedback actually occurs. But operator threshold shift occurs, and the feedback then becomes fully developed before it is detected.

To help avoid this severe, damaging feedback, a limiter on the monitor channels is recommended. It must be adjusted so that the gain reduction portion of its range falls slightly above the normal operating levels: This will not cause wholesale gain changes when program ceases, which can cause feedback.

One-third octave equalization can be quite helpful in “flattening” the response of a given setup, and should also be used to eliminate some of the serious “ring modes” or feedback modes after the room/system has been made usefully “flat”,

The equalization applies to the house system as well as to the monitors, although separate curves will be used. One thing to keep in mind is that over equalization will cause a hollow sound due to rapid frequency dependent phase shifts concomitant with extreme narrow band equalization.

Also, the soundman shouldn’t be overly eager to comply with requests to put almost everything in the monitors. This is often requested by less experienced performers. One slightly sneaky way to resist this is to have only a few inputs on the monitor mixer.

Equalization & Analysis
The purpose for narrow band equalization and analysis is to “flatten” the response in a given area for the room/ speaker combination. After this first step, the equalization equipment can and should be used to eliminate the most severe ring modes that are present when the microphone gain approaches the feedback point.

The stage should be configured as it will be for the performance during these adjustments. Depending on the amount of acoustic gain desired and on the amount of “naturalness” desired in the channel, this second step can be minimized or left out altogether.

Feedback is due to the combined effects of sound, from the speakers’ direct and diffuse fields, impinging on the microphones (on or off axis) either directly or by single or multiple reflections.

When the microphone to speaker distances are small, the off axis sound from the speaker may enter the microphone directly, causing feedback at unusually low frequencies. Although the low frequency feedback is sometimes difficult to hear, it does cause added distortion, use power unnecessarily, and it can obviously disturb people.

There are two criteria which, together, determine whether or not a system will feedback, and at what frequency the feedback will occur.

The frequency is directly related to both the transit time of the sound wave through the air, and to any electrical delays due to irregular frequency and phase response.

While the air delay is usually much larger than the electrical delay, excessive equalization can have an effect. The total phase shift at the frequency of feedback will be related to the feedback frequency in this manner: The total loop phase shift is equal to one complete period (360°) or some multiple of a period of the feedback frequency.

While the time delay just described will determine the feedback frequency, it alone docs not determine whether feedback will actually occur at that frequency or any other. The gain of the total system “loop” must be greater than unity at that same frequency before feedback commences.

A short discussion of the relationship between phase and time seems appropriate. Phase shift and time delay arc two ways to say the same thing in different systems of measurement. Phase relates to percentage of one complete cycle for a wave of a given frequency.

The units of phase measurements arc degrees; 360 degrees = 1 cycle or period. Phase may also be thought of in terms of the period of the wave. For example; a phase shift of 270° at 1000 Hz, corresponds to a time delay of 3/4 of the period of that frequency, or .75 millisecond.

Figure 4.

Given the speed of sound as approximately 1100 feet per second this converts to a transit distance of about .6 feet. Figure 4 shows the potential feedback frequencies for an idealized feedback system based on microphone-to-speaker distance of 10 feet.

Changing the position of the speakers and/or the microphones will shift the frequencies of the potential feedbacks. If this shift moves the potential feedback frequency into a range where there is insufficient “loop” gain to allow feedback, then the shift in position will cure the feedback problem.

Related ways to eliminate feedback include changing the pattern or direction of a speaker or microphone and changing the reflection of sound in the stage area. These changes can alter the gain spectrum sufficiently to move the potential feedback into a more manageable portion of the audio spectrum

Ideally, 1/3-octave or other narrow band equalization for ring mode suppression should not be applied to the total program channel, it is best applied only to the group or microphones involved in the feedback threshold problem.

Narrow band equalization can be applied in a manner that is less than grossly apparent to the ear and yet retains some beneficial effect on the feedback.

It should be noted that reversing the phase of the system anywhere in the loop reduces any potential feedback frequency by one octave. This is because the phase inversion has added one half period to all the frequencies. Phase reversal can be used, therefore, to change the nature of the feedback situation entirely.

It may also be noticed that in small rooms a reversal of phase has a profound effect on the low bass response of live music only. This is because the live music from the stage combines in the room with the sound from the reinforcement speakers, for better or for worse.

It is, for this reason, desirable to control the phase of the high and low frequency portions of the reinforced sound from the mix position. Some severe effects may be noticed in the frequency response in the vicinity of the crossover frequency when the phase is reversed, but the advantage in overall response may outweigh these effects.

A likely place to change the phase is in the system’s electronic crossover (assuming the system has or needs such a unit). Ideally the crossover would have independent controls, located at the mixer, for the phase and level of each frequency band.

There should also be some method for monitoring the power output levels of the amplifiers with respect to the optimum peak level. More detailed specifications would depend on the application and the specific speakers used, which is a subject worthy of a separate article.

Mixing
I feel that I should restrict most comments on mixing to areas that are peculiar to reinforcement. This is difficult because there is a great similarity to mixing for recording.

Microphone mixing and equalization go hand in hand with microphone selection and placement. The person responsible for engineering/production should have direct communication with and control over the stage microphones and environment.

However, often the person making production decisions will not be too familiar with the requirements and peculiarities of the audio system. For that matter, he may not know reinforcement technology in general.

An understanding should therefore be reached, before the show, about who will do what, and about when and where it will be done.

The system operator will usually be responsible for program continuity, including any recorded music or announcements during the intermission and set changes (“The group will be showing up any minute now, folks!”).

The usual signal processing techniques are used, but exercising extra care in the area of compression, limiting and equalization. These all have severe effects on the amount and the nature of feedback.

On the other hand, “hard” limiting is often used to protect the speakers. The best place for this is after the crossover, as this avoids modulation effects (specifically, the higher energy, low frequencies will not modulate the gain of the highs).

When there arc problems obtaining the desired degree of intelligibility, which frequently occur on the vocal channels, it is recommended that the mixer resist the urge to raise the gain of the respective middle or high frequency equalization. Instead, try turning everything else down, and start to work from there.

To do otherwise is an invitation to get caught up in the circle of increasing high frequency threshold shift, requiring further increase in HF boost. This results in listening fatigue, poor quality sound, and ultimately severe feedback. The combined effects of the hearing loss, due to threshold shift, and of the EQ allow the feedback to get out of control before it is noticed.

For every performance setup, some responsible person should walk throughout the audience areas to determine that the distribution of sound is even in most sections of the room. Any gross inadequacies in dispersion, reverberation, reflection, or phasing can be dealt with before the sound check. The sound system can then be adjusted to taste from the position of the mixing console.

Preparations Prior To The Gig
Some of the recommendations set forth in this article are idealistic, to say the least. Compromises will be made.

Various types of equipment and differing complexities of staging, audience, and artist requirements will demand compromises not to mention money, or the lack of it. This author is of the opinion that sufficient advance notice of some unusual requirement will always help the situation.

If need be, some special equipment can be rented for a single show. Artists’ managers, their agents and soundmen should know what their performance requires in the realm of equipment and of service. They should make some effort to communicate these requirements to the sound system operator at least a week in advance so that the necessary provisions can be made.

 
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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Posted by Keith Clark on 01/11 at 01:48 PM
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WorxAudio Deployed At First Christian Church

Worxaudio X3-P compact line array provides superior coverage and audio quality to Venice, Florida-based First Christian Church.

With three services each Sunday—two of which are contemporary in nature— First Christian Church, Venice, Florida, has a vibrant worship program that caters to a diverse congregation.

After enduring lackluster performance from their sound reinforcement system for far too long, the decision was made to upgrade their sanctuary’s facilities with a system that would deliver a high level of speech intelligibility and first-rate music reproduction.

They ultimately resolved their sound system shortcomings by installing a WorxAudio X3-P compact line array.

Cape Coral, FL-based Creative Sound Solutions, LLC, was contracted to design and implement the church’s new sound reinforcement system.

Rob Robinson, general manager and co-owner of Creative Sound Solutions, discussed the challenges that led to the installation of the WorxAudio X3-P compact line array.

“First Christian Church’s sanctuary seats roughly 500 people and is 80 feet wide by 50 feet deep,” Robinson explained, “with the stage / pulpit area facing into the width of the room.

“This required a loudspeaker system with broad horizontal dispersion in order to provide consistent coverage from side to side as well as front to rear. The lack of consistent sound coverage with the previous sound system created a number of issues—most notably, the inability of the congregation to understand what was being said.

“Because of this, speech intelligibility was a crucial factor. Services are nowhere near as meaningful if people can’t understand what’s being said.”

“Music also plays a vital role in the church’s worship services,” he continued. “With a 5-piece praise band and a vocal team of roughly the same size, the new sound system had to have equally capable music reproduction characteristics.

“The WorxAudio X3-P line array is unique in that it is a 3-module, all-in-one compact system with extremely broad—160-degrees—horizontal dispersion and excellent music reproduction characteristics. After consulting with Hugh Sarvis at WorxAudio Technologies, we determined that a central cluster—positioned 20 feet over the pulpit area—would serve the room nicely.”

The X3-P’s throw capabilities mark a dramatic departure from conventional line array systems of this size and class. The upper two modules of the X3-P provide 10-degree vertical dispersion while the lower module delivers a 25-degree vertical pattern.

Combined, the three modules create a 36-degree vertical system with an unusually broad horizontal dispersion of 160 degrees. The system is powered by WorxAudio Technologies’ highly-regarded PMD-1.5 digital power amp with built in DSP processing.

First Christian Church’s new sound reinforcement system was deployed recently and was placed into service immediately thereafter. Since that time, Robinson reports the new system has had an extremely positive impact.

“Because of the X3-P’s dispersion characteristics, we were able to cover every inch of the space. This system effectively eliminated the dead spots inherent in the previous sound system and, now, there is great sound quality and high speech intelligibility everywhere you sit.

“Ken Wagner, the church’s media director, was really impressed with the fact that no matter where he walks throughout the room, sound quality is remarkably consistent, clear, and free or dropouts. The new WorxAudio system has been generating rave reviews—and for us, that’s the best compliment of all.”

WorxAudio

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Posted by Keith Clark on 01/11 at 11:24 AM
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Scott Leslie Named New President Of Renkus-Heinz

Will spearhead company's ongoing growth in sales and staff

Renkus-Heinz has announced the appointment of Scott Leslie to the position of president.

Leslie comes to Renkus-Heinz from JBL Professional, where he served as vice president of engineering. His industry background includes stints at a wide range of technology leaders including Sun Microsystems, Tektronix, and Altec Lansing.

Leslie is also founder of Evidant Corporation, providers of business intelligence for IT solutions, where he has held executive level positions since 2002.

He joins Renkus-Heinz as the company is experiencing unprecedented growth, with a substantial surge in sales and significant growth in its engineering staff.

Leslie’s new position will see him spearheading that continued expansion, working closely with company founder and chairman Harro Heinz.

“Scott brings with him a wealth of experience in business and marketing, combined with a deep understanding of today’s complex technologies,” states Heinz. “He is the ideal person to assume this position at a very exciting time for us.”

“Renkus-Heinz is one of our industry’s legendary success stories, built on innovation and great leadership,” Leslie adds. “My new position represents the opportunity to bring together everything I’ve learned throughout my career, to bring Renkus-Heinz to an even greater level of success.”

Renkus-Heinz

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Posted by Keith Clark on 01/11 at 10:41 AM
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Tuesday, January 08, 2013

The Basics Of Audio System Gain Structure

General guidelines to setting the gain structure for an audio system and making final adjustments

Although this is not a rigorous treatment of the subject, the following are general guidelines that may be helpful in setting the gain structure for an audio system and making final adjustments to it for best results.

A Basic Procedure
As a minimum, you need to use at least a voltmeter along with the maximum output voltage specification for each piece of electronics. The preferable tool is an oscilloscope that can be used to observe the signal directly. If neither of these are available, you can use the level or clipping indicators on each piece of equipment in the signal chain.

Without some method of determining the clipping point for each piece of equipment, you cannot expect to optimize the gain structure.

Before setting gain structure disconnect the loudspeaker(s) from the amplifiers. If controllers are used that sense the amplifier outputs do not disconnect the sense lines.

The basic procedure is to use a test signal (a sine wave signal is ideal) and set the first piece of equipment in the signal chain (usually the mixer) so it is just below maximum voltage output as read on the voltmeter or on equipment’s output meter or just below clipping as observed on the oscilloscope.

Without changing this signal level adjust the level controls on each piece of equipment following, including the power amplifiers, so it is just below its maximum output.

You will find that the input level controls on the power amplifier will end up being set anywhere from 10 dB to over 20 dB of attenuation.

Note:
Due to differences in the capabilities of devices in the signal chain, it may not be possible to achieve the results exactly as stated. Gain structure should be set after any equalization is set for the system so that any boosts (which reduce headroom in the equalizer) are taken into account. For each device, make sure it is the output and not the input that is clipping.

Always make sure that the any limiters in the system go into limiting prior to anything else in the system going into clipping. In this way the limiter, rather than a clipped signal somewhere else in the signal chain, will control the system’s maximum output.

Remove the test signal, turn off all equipment and reconnect the loudspeaker(s) and the system is ready for level balancing, assuming you have more than one loudspeaker. If you have only one loudspeaker, the system should be ready for use.

Residual Noise
Horn loaded loudspeakers and drivers have high sensitivities which means that they produce a relatively high output for a given electrical input. This includes the residual electronic noise of an audio system.

By setting gain structure properly and using high quality professional electronics, this noise should be at or near inaudibility.

Among the noisier electronic devices are 16 bit digital devices such as some signal delays. They have signal to noise ratios that are only 90 dB.

However, if the gain structure is set correctly this means that, for example, if the system can produce 120 dB SPL at maximum output, the residual noise should be about 30 dB SPL.

This would be acceptable for a quiet recording studio. If residual noise is a problem, gain structure is usually the culprit – it is never the loudspeakers.

Level Balancing
Once the system gain structure is set, the level balances can be adjusted. This may mean the levels between HF and LF sections of a biamplified loudspeaker, a full-range loudspeaker to subwoofer level, levels between multiple loudspeakers, or between main and delayed loudspeaker array.

The idea is to make the system sound the best it can without using any equalization. This may be done using acoustic test equipment such as an RTA (real-time analyzer), TEF analyzer or similar. You must always determine the final level balance by listening to a variety of known program material.

Level balancing can also be done entirely by ear if acoustic test equipment is not available. In any case, the preferred method of adjusting levels for balancing is to use the amplifier input level controls.

Important:
In order to maintain the system’s dynamic range that was maximized by setting the proper gain structure, do not turn up the input level controls of any of the amplifiers. For example, if you decide that a subwoofer is not loud enough, do not turn up the input level control of its amplifier. Instead, turn down the input level of the full-range loudspeaker’s amplifier.

Equalization
Once level balances are set, you can then equalize the loudspeaker(s), if necessary. Community loudspeakers are optimized for highly accurate and well balanced reproduction “out-of-the-box.” Generally, equalization should only be needed to eliminate difficult feedback frequencies or to adapt the system to a difficult acoustic environment.

You should not need more than a few dB of boost or cut equalization for any particular range of frequencies. The best equalization techniques involve cutting rather than boosting frequencies.

Final Results
Once the above procedures are followed, your loudspeaker should reproduce audio cleanly, clearly and with all frequencies in good balance.

Noise should not be audible and you should be able to drive the amplifier(s) in the system to maximum output on normal program material with no significant distortion or other undesirable sound. If limiters are used, the onset of limiting should occur just before any amplifier clipping.

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Posted by Keith Clark on 01/08 at 03:02 PM
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