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Friday, October 29, 2010

Audio Basics: Stage Monitoring Simplified

Sage advice on mixing monitors and the house simultaneously.

Encompassed in today’s live show are several individual shows, for example, light shows, laser shows and more.

It’s not uncommon for each of these independent shows to have its own set of engineers.

The focal point is the band/artist, and they’re the reason all of the other shows are taking place.

And, it should be pointed out that most of these “shows within the show” are presented for the audience.

Good engineers realize that the monitor mix is a show in itself. It’s the only show that the musicians get to hear, and it certainly is the mix that most affects their performance.

Big shows usually have a separate engineer for the monitor mix, but for average shows one valuable individual functions as both the front of house (FOH) and monitor engineer.

To accomplish a good monitor mix, you must understand your particular mixing console; we can, however, examine some basic principles that apply unilaterally.

While working your magic behind the FOH console, most people don’t realize the work you’re doing for the musicians.

The musicians also have to hear the performance, except they usually want to hear something totally different than the house mix.

In fact, many bands have members that each want (or even require) a different mix than another band member. That means multiple mixes, all running at the same time, which can present some challenges to the engineer, because all of these mixes cannot be monitored simultaneously.

Most consoles allow toggling between each of the mixes, allowing you to make changes to each individually. This, of course, depends on what features the console has and how you, as an engineer, decide to accomplish your monitor mix.

Let’s first take a look at some obstacles you may encounter in your equipment.

Snakes, Sends, & Returns
The snake, of course, is the multiple input cable that all of the instruments plug into on the stage. The cable then is plugged into the console to carry the signal from the stage to the mixing console.

The snake also has “returns,” which, as the name implies, route signal from the console back to the stage and musicians.

The amount of returns available will directly affect the amount of signal that can be routed back to the stage. For example, a 24 x 4 snake offers up to 24 pathways to the console from the stage, and four pathways back to the stage from the console.

With four return paths, the possible combinations of mixes are four independent mono mixes, two stereo mixes or one stereo mix and two mono mixes.

The more return paths, the more possibilities you have to run monitor mixes.

Bear in mind that these return paths also must carry the signal to the main loudspeakers.

Another limitation that can be encountered is the amount of pathways that a mixing console has to use as returns to the stage.

A 24 x 4 x 2 console has 24 inputs for instruments, four bus (or group) outputs, and a pair of outputs for the mains.

Bus or group outputs (sometimes called sub-outs) can be used as monitor return outputs.

Four-bus outputs would yield the same combination of possibilities as a four-return snake.

Obviously, more bus outputs equals more possibilities for this type of monitor mix.

Probably the method with the most possibilities is routing monitor mixes with the auxiliary sends. Like buses, auxiliary or aux sends can be used to route monitor mixes.

Although aux sends are used for routing signal to effects processors, they are very useful in running monitor mixes. For aux sends to be useful as monitor mixes, they must be able to be used in what is known as pre-fader mode.

There is usually a button next to the aux send pot on each channel that will allow you to switch between pre and post-fader modes. Keep in mind that for each mono return path, a separate amp at the other end for a power source is needed.

Stereo requires a two-channel power amplifier, or a separate amp for the left and right sides.

Have you checked your equipment for features? To make it easier to describe some basic techniques, we need a typical scenario.

Let’s assume we have a 24-channel input console with four bus outputs and at least four auxiliary sends. Our console is also equipped with a stereo headphone output, so we can listen to each mix separately without listening to (or affecting) the house mix.

Most consoles possess this capability because it is necessary for the engineer to listen to alternate mixes during a performance.

Our snake has 24 channels with four return paths. Let’s also assume that we will be running a stereo house mix. To achieve a good monitor mix, there are several ways to get there. So, let’s take the trip.

Get On The Bus
Each channel has a feature that will allow assignment of its signal to a group or bus out. These assignment groups include the main L/R bus.

The main L/R group will be used to route signal from the console’s main output jacks on the back of the board through the snake’s first two returns. The faders on the console labeled “mains” or “L/R” (or something similar) will control the amount of signal.

The returns from the snake on the stage side will be routed to the stereo power amp (or amps) that powers the main front of house speakers. This way, the signal from the power amps takes the shortest path to the speakers.

In addition to the main L/R outputs, a four-bus console has four additional outputs that correspond to faders that control the signal routed to them.

The faders are typically labeled “bus 1,2,3,4” or “Sub out 1,2,3,4” etc.

Each channel has a switch, which allows you to assign its signal to a particular bus.

With this configuration, we could run a stereo monitor mix.

If we decided to do this, we would assign each channel to buses 1 and 2.

The output of bus 1 will be routed through the snake’s return 3, and the output of bus 2 output will be routed through return 4.

Remember we’re using returns 1 and 2 for the main house mix. Return 3 will be routed to the amp that powers the left side of the monitor mix and return 4 to the right side.

I only recommend doing a stereo monitor mix if in-ear monitors are to be used. Most applications require the use of on stage loudspeakers (monitors or wedges).

For this reason, it is a good idea to run a mono monitor mix. Monitors on stage would normally be too complex to run a stereo monitor mix. By assigning all of the instruments to bus 1, we can be sure that all of the stage speakers represent an overall mix for all of the musicians on stage.

Because we are only using bus 1 for this mix, we could then use bus 2 for some accent monitors on stage.

For example, if all of the musicians can hear the overall mix, we could assign only the vocals and other lead instruments to bus 2 and route that bus to spot monitors for singers (who like to hear themselves heavily in the monitor mix) as well as other lead instrumentalists who cue off of each other.

If we had more returns in our snake we could even provide accent monitors for the bassist and drummer/percussionist, for example, by using buses 3 and 4.

One limitation of using buses for your monitor mix is that if you assign a particular channel to a bus, you might not be able to route that signal to another bus or at least not the bus you need.

Another limitation is that on some consoles, you cannot adjust the amount of signal that each channel contributes to each bus. That means that whenever the bus is assigned, the entire signal from that channel is routed to that bus assignment.

The amount of individual signal from each channel to the bus is determined by the position of the channel fader.

Many manufacturers design features that allow for more possibilities; however, that usually means more circuitry, hardware and board space for additional faders or pots.

More features means more expensive. This is a quick and easy way to run monitors, but there are other avenues that lead out of the console - the auxiliary send outputs (aux sends).

Working The Aux Sends
In addition to bus output jacks, mixing consoles also have output jacks that correspond to their auxiliary send pots.

Remember that our sample console has at least 4 auxiliary sends.

Each one of these channels has a level adjustment pot for each of the sends.

The console will almost certainly have an aux master section that will allow for control of the overall signal level of each aux output.

The aux 1 output jack should be routed to the snake’s return 3. Aux 2 out should be routed to the snake’s return 4.

We still need to use returns 1 and 2 for the house mix. By utilizing the aux sends, we can accomplish two mono mixes. Each of these mixes can contain any amount of each channel.

For example, if we want more of channel 9’s audio in the aux 1 mix, we simply turn up the aux 1 pot on channel 9.

By using the level adjustments for aux sends 1 and 2 on each channel, we can create a mix that is suitable for the lead instruments on aux 1 and a completely separate mix that is suitable for the rhythm section on aux 2.

One feature that our mixing console must incorporate is that ability to route the sends in “pre-fader” mode. In pre-fader mode the position of the volume faders on each channel has no effect on the level being sent down each of the aux sends.

This means that if we turn down the lead guitar in the house mix by lowering the channel’s volume fader, the monitor mix will not be changed. Once again, if our snake had more return paths, we could utilize additional sends for more on stage monitoring possibilities.

By assigning the aux mixes one at a time to the headphone outputs, you can make adjustments to each mix without affecting what the audience hears.

In addition, by routing the signal outputs of our effects processors to channel inputs, we can send some effects to each of the monitor mixes.

For example, by routing the outputs of a reverb processor to the inputs of channels 23 and 24 of our mixing console, we can send an adjustable level of reverb to each of the two monitor mixes by simply turning up sends 1 and 2 on those channels.

Be cautious of routing signal from channels 23 and 24 back to the inputs of the reverb unit as this will result in an electronic feedback loop.

If you use send 3 and 4 for the inputs to your effects processors while using channels 23 and 24 for your reverb returns, turning up sends 3 and/or 4 on these channels will create this kind of loop.

Using aux sends for monitor mixing is probably the best and most popular approach as it affords the engineer the most versatility and functionality.

Hopefully, this article will arm you with some of the knowledge critical in implementing some basic rules of audio. Always make sure you know the equipment you own and the equipment you intend to buy. Good luck, and have fun!

 
Scott Foulkrod is the Audiovisual Coordinator for the Houston Rockets in Houston, TX.

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Royer Labs Adds New European Distribution Partner

The new strengthens Royer's presence within Germany, Switzerland, and Austria.

Royer Labs has announced that the company has reached a new distribution agreement with the German company S.E.A. Vertrieb & Consulting GmbH.

The new distribution agreement enables Royer Labs to strengthen the presence of its premium microphone products among musicians and audio professionals throughout the key European countries of Germany, Switzerland, and Austria.

Headquartered in Emsbüren, Germany, S.E.A. Vertrieb & Consulting carries a wide range of products for today’s audio professional, including microphones, signal processors, mixing consoles, digital audio networking equipment, and sound reinforcement systems.

Guided by Managing Director Uwe Kirchfeld, the team of fifteen employees provides rapid order processing and shipping, along with full consultation and support services.

“We are very pleased to be representing Royer Labs,” said Uwe Kirchfeld. “Royer is one of the world’s premium microphone companies and is a perfect fit with our existing line card.”

“The microphone is one of the most important components in terms of ensuring a quality recording or true to the source sound acquisition for live sound reinforcement purposes and Royer Labs is a first-tier choice among today’s discriminating audio pros.”

“Royer microphones stand at the forefront of ribbon microphone technology. Their ability to add a rich, warm sound makes Royer microphones the ideal complement to the digital recording process and their robust build quality and ability to handle high SPLs makes them equally well suited to miking applications for today’s touring professionals. We are positively delighted to represent the line.”

John Jennings, VP of Sales and Marketing for Royer Labs, is also excited about the new arrangement. “We believe Royer Labs fits in perfectly with the brands represented by S.E.A.,” said Jennings.

“The addition of Royer Labs microphones brings a much sought after, and previously elusive, brand to S.E.A.’s product offerings and I believe Royer makes a great fit with their other lines. S.E.A. has a solid reputation among the professional audio community and I believe the addition of the Royer line makes a great fit for both firms. I look forward to working with them and have full confidence in their ability to represent our company.”

Royer Labs Website

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Selecting Active Speakers For Your Worship Needs

Do they offer a lower entry cost for an entire system?

For years, when you thought of active speakers, often the image was of studio monitors.

However, now when you pick up a catalog it’s obvious that nearly every speaker manufacture has an active loudspeaker line.

I often hear people ask, “What’s so special about active speakers? Why would you want to put all of your eggs in one basket?”

“What are the advantages of active speakers? Aren’t active speakers overpriced?”

Read on, let’s find out!

Active Or Powered
What’s the difference? Active speakers are also known as powered speakers.

I refer to them as “active”, because there is much more happening inside the box besides amplification. An active speaker includes of a box, drivers, electronic crossover, compressor/limiters, delay, equalization and amplifiers.

The truth is powered speakers should sound better than conventional speaker designs.

All of the crossover points, equalization, time alignment, compression, limiting and amplification matching are fine tuned to meet the manufacturer’s intended sound.

The key here is the intended sound quality!

We all feel that we can do a better job tweaking the speaker than the manufacture, right? What most people don’t understand is where the break point is.

Today, speaker processors give speaker manufactures more control over crossover points and equalization but certainly not with power amplification to speaker component matching.

Proper gain or power (wattage) matching is one of the most important elements of making a speaker sound good and insuring the longevity of speaker components.

I come from the old audio school of thought that there is no such thing as too much “available” power.

There is, however, such a thing as wasting money on power. You don’t need 5KW power amps for the high frequencies.

Don’t forget though, that under powering any speaker can be as harmful as overpowering. The key to success is controlling or harnessing amp power.

Quality active loudspeakers match amplifier power (wattage) to component need (i.e. power handling). The manufacturer already did the math for you.

Bold, But True
Powered speakers, as a total sound system design, cost less than equivalent conventional component PA systems.

For this bold statement to be true, one must consider everything that makes up a PA such as: crossovers, equalizers, compressor/limiters, speaker processors, amplifiers, rack space and cabling.

Material cost of an active speaker remains the same when compared to a passive type box.

Further, active and passive designs both require crossover networks, equalization, compressor/limiting, time alignment and power. Don’t under estimate this reduction in required space.

With powered speakers there is much less gear to install and store. Most importantly, once your loudspeaker system goes “active”, the total system cost is reduced. Lets break this down into fundamentals, and see how this is achieved.

The audio phrase “front-end” refers to the first part of the signal path. In this case, it is the input. Depending on the speaker application, this input will be at line level, mic level, or both.

A good front-end design is essential to prevent those nasty RF signals from bleeding in the audio signal. It is pretty embarrassing to be sitting in church to hear a local trucker, on his CB radio, bled thru the sound system during the sermon saying “Some Smoky is on his a—-“. Not good…so why not use technology which prevents this issue?

For many small jobs, an obvious cost savings comes when no mixer is required. Still, every advantage comes at a price. As such, from a design and material cost viewpoint, quality front-end design is one of the few areas where of active speakers add direct cost.

Back-End Electronics
The back end of the electronics in an active speaker are where processing, such as active crossovers, EQ points, compressor/limiters and delay, takes place.

These multiple functions are why considerable savings can be via internal processing compared to external digital speaker processing.

With an active speaker, electrical parameters are pre-determined by the manufacturer. This is not all digital processing wizardry either. Precise filtering can be implemented using low cost analog components. In fact, it is still cheaper to use analog components compared to operating in the digital domain.

As soon as there is a need to have external processing adjustments, this may no longer be true. However, if tweaking is ever allowed, the main purpose of an out-of-the-box, acoustically aligned speaker would be effectively eliminated

Eventually, the digital domain will become more cost effective over analog, especially when audio signal remains all-digital from the mixer to the loudspeaker front-end. The high cost of digital is the conversion from analog to digital and from digital back to analog.

DSP and the microprocessor are the inexpensive part of a digital audio design. As of today, it is still more cost effective to use analog components, for active speakers, rather than digital.

Power Supplies
The essential difference between stand-alone amplifiers and active speakers is the required wattage the power supply has to deliver.

Traditional component amplifiers must be designed to handle various external loads. This requirement inevitably causes component amplifiers to be overbuilt.

With active speakers, speaker load is predetermined, as is the maximum current load. If current load is predetermined, you can reduce the requirements of the supply thus, reducing the design cost.

As soon as the external elements faced by stand-alone amplifiers are removed, a designer needs only to implement the exact amount of circuitry required.

Maximum current requirements are then fixed and cannot be altered from an outside source.

Once you know this, the need for short circuit protection, additional output transistors, larger pre-drivers and massive heatsinks are practically eliminated.

Another advantage active speakers have over stand-alone amplifiers is weight loss (i.e. less metal). The elimination of large heavy heatsinks and power supply transformer seriously reduces overall system weight.

The need for an expensive, heavy rack mount chassis is also eliminated, and should yield further cost reductions, right? One could only hope!

Looking Forward
I am certainly not trying to say that active speakers are the solution to every audio application as there will always remain a need for traditional PA systems.

My point is that anyone designing a new system should always consider active speakers as a total system option, especially if your needs often require occasional portability.

I suspect you will find discover some extra cash in your budget and fewer headaches down-the-road if you choose active speakers over passive.

Jeff Kuells is an audio engineer and audio manufacturing consultant and was previously director of engineering for a major amplifier manufacturer.

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L-ACOUSTICS Accepts Parnelli Award for K1

The line source array system takes top prize this year in category of “Indispensable Technology – Audio”

L-ACOUSTICS has announce that its K1 stadium line array system was recognized with the award for “Indispensable Technology – Audio” at the 10th annual Parnelli Awards ceremony held at the Rio All Suites Hotel & Casino in Las Vegas on October 22nd.

“We are very thankful and humbled by this recognition of our innovative K1 system,” said Stéphane Ecalle, L-ACOUSTICS’ marketing director.

“But much of the honor should also be extended to those K1/KUDO/LA8 Rental Network owners, sound designers, and touring engineers that have collaborated with us during the product’s Pilot Phase because their contributions truly helped hone K1 into the remarkable system that it is today.”

These early adopters of K1 include:

Adlib Audio (UK), Agora (Italy) Arpege Son Lumiere (France) Black Box Music (Germany) Britannia Row Productions (UK) Clearwing Productions (US) Dispatch (France) Euroshow (Russia) Firehouse Productions (US) Hibino (Japan) Jands Production Services (Australia) Loud and Clear (US) Loudness (Brazil) Mid-America Sound (US) MSI Japan, Music & Lights (Sweden) Potar Hurlant (France) Rat Sound (US) Rent-All (Netherlands) Satis&fy (Germany) Sirius Showequipment (Germany) Solotech (Canada) Sound Image (US) SSE Audio Group (UK) Tokyo Sanko (Japan)

In related news, Sound Image of Escondido, California was honored with the award for Sound Company of the Year, while six-time TEC Award winner Robert Scovill took the prize for FOH Mixer of the Year.

Sound Image was one of the first North American sound companies to purchase L-ACOUSTICS’ K1 system, putting it into the hands of Scovill early this year for Tom Petty and the Heartbreakers’ highly-successful Mojo tour.

Also, A1 Audio’s Al Siniscal – who previously installed one of the world’s largest V-DOSC systems into Las Vegas’ Planet Hollywood (formerly Aladdin) Theatre of the Performing Arts – was honored for his technical contributions with the Audio Innovator of the Year award.

And Miami Gardens, Florida-based Beach Sound – a longtime member of L-ACOUSTICS’ Rental Network – was also presented with an award for Hometown Hero Sound.

L-ACOUSTICS Website

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Video: An Interview With Kenton Forsythe

Part 1 of 4 in a series of interviews with the founder of EAW and industry legend Kenton Forsythe.

This first part in an ongoing series recorded by Gary Zandstra, Kenton Forsythe discusses EAW’s history and the American way.

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

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Recording Guitarists: How To Recreate The Setups Of Classic Guitar Gods

An excerpt from Jon Chappell's book which highlights classic guitar setups and how to dial in that sought after sound.

Guitarists are constantly seeking their own sound or unique voice.

However, producers often resort to giving instructions like, “Give me a raunchy blues sound à la Stevie Ray Vaughan,” rather than “Give me something wholly original that I’ve never imagined before.”

It’s not that producers are unimaginative or that they deliberately want to mimic another guitarist’s sound; it’s just that categorizing sounds saves a lot of time and gives you a point of departure.

Often you’ll hear producers requesting that a guitarist get an “early Van Halen sound,” or a “Hendrix rhythm sound à la ‘Little Wing,’” or a “Dimebag Darrell over-the-top-solid-state-distortion” sound.

These are perfectly legitimate requests, and will come as often as the ones involving instruments and amps—as in, “Give me that Les-Paulthrough- a-Marshall sound, will ya?”

With that in mind, here are the setups of 14 well-known guitarists, from slide master Sonny Landreth to neoclassical god Yngwie Malmsteen. Keep in mind that these are only guides to one guitarist’s particular sound.

This is not the only sound that a particular guitarist produces, but the one he has used for a significant portion of his recorded work, and the one we associate with his “classic” sound.

Dimebag Darrell
The late Dimebag Darrell was the poster boy for the heavy metal ethos, and never made any apologies for his piercing solid-state distortion sound.

Like Hendrix, Darrell went first into a wah pedal (either the Dunlop or the DigiTech were on, but not at the same time), and then into another pitch shifter (the PS-3) before hitting the gain-shaping distortion box, the Boss DS-2.

After going into an MXR pedal–based graphic EQ (set in a “V” shape, as all good metal guitarists do), Darrell passed his signal through a rack containing a parametric EQ (for any final tone shaping before the amp stage), an MXR Flanger/ Doubler (his only time-based effect), and then through the Rocktron Guitar Silencer as his noise gate.

The gain on Darrell’s amp was rarely set to anything but 10, and the presence and bass were goosed while the treble and mids were cut, which kept the sound from becoming too brittle.

Kirk Hammett
The unique setup of Kirk Hammett’s wah pedals is the result of Metallica’s performing logistics.

The band usually sets up different “performing stations” when they play, and at various times during the concert they simply rotate around to the next area. But this presents a problem when you have to have access to your effects at all the various locations. So

Hammett devised a rig where the wah pedals act as mere controllers (allowing for longer low-impedance lines to run between them), while the brains of the wah sit in the rack offstage.

Hammett uses two preamps, the Marshall JMP-1 and the Mesa/Boogie TriAxis. These feed Mesa power amps, which drive three Boogie 4✕12 cabinets.

Hammett also uses a Mesa Dual Rectifier, but instead of relying on one head to drive three cabs (or employing three amp heads), he will load down the speaker out with a 300- watt speaker (which is buried offstage somewhere), while a load box takes the line-level signal and delivers that to the three power amps.

This has the effect of normalizing the amp output with the preamps’ output and gives a relatively consistent signal to all three tone-shaping devices.

Jimi Hendrix (Then and Now)
Jimi Hendrix’s tone is possibly the most emulated of all time, so many manufacturers have devoted considerable resources to recreating the gear that helped shape Hendrix’s tone, but is no longer available (or is too rare and expensive to come by easily).

First, a look at Hendrix’s original setup. Hendrix went first into a Cry Baby or Vox wah and then into a fuzz.

He used primarily two: an Axis Fuzz and the Roger Mayer–designed Fuzz Face.

From there he went into a Dunlop Uni-Vibe and Mayer Octavia before going into a 6550-equipped Marshall.

For the modern Hendrix sound, the Dunlop Cry Baby is the wah of choice, and then distortion units by Fulltone (Fuzz) or Prescription Electronics (Experience Fuzz) are considered de rigueur.

Fulltone makes the DejáVibe that closely emulates the original for a fraction of the price. For that square-wave octave sound, the Boss OC-2 does a nice job.

Much has been made of the fact that Hendrix played a right-handed guitar flipped upside down and strung left-handed.

This means that as a left-hander himself, Hendrix played the guitar conventionally, but there were several key differences in the imposition of his instrument:

1. The string tensions were all different, because Strats normally have the first string as the longest.

By reversing the strings on the Strat’s inline tuner configuration, the sixth string became the longest, and the increased tension on the thickest string significantly changed the resonant properties of the guitar.

2. The pickups were angled the “wrong” way, with the bridge pickup slanted toward the neck instead of the bridge.

This placed the first-string pole piece well up from the bridge instead of right next to it, as on a normally strung guitar. (The pole-piece heights were all different as well.)

3. The bar and the controls were above the right hand. While this may not affect the guitar’s tone per se, it affects the way a performer approaches the instrument, and we know from listening to Hendrix’s music that this was perhaps the most influential element on his tone.

Eric Johnson
Eric Johnson is a tone purist and therefore runs a fairly straightforward setup.

He gets almost all of his tone from the amp, invoking a Chandler Tube Driver judiciously and mostly for increased sustain rather than distortion.

Johnson employs two A/B boxes, which gives him three separate audio paths to choose from.

The first path goes to a tapedriven Echoplex and then into the Chandler Tube Driver and into a 100-watt Marshall head and cab.

The second path goes to a Mayer Fuzz Face and then to an MXR Digital Delay before also going to a 100-watt Marshall head and cab.

The third path begins with an Echoplex (like path #1), but is then split by a TC Electronic Stereo Chorus (mono in/stereo out) and output to two Fender Vibrolux amps.

Sonny Landreth
Slide guitarist Sonny Landreth has one of the most unique sounds going, partly because he employs an actual miked speaker in his setup.

He isolates and encloses the speaker and mic, and often works an honest-to-goodness Leslie cabinet into his rig (no simulators here!). A healthy overdrive and a mature and evolved vibrato technique round out this singular slide artist’s setup.

Known for his monstrous slide chops and widely varied repertoire, Landreth is also a purist when it comes to sound. He goes through an old red MXR Dyna Comp to get a smooth, sustained sound, and then into either a Demeter TGA- 3 or a 1954 Fender Deluxe amp.

From there, things get interesting. The speaker out goes to a Demeter Silent Speaker Chamber, which is an iso-cab housing a Celestion Vintage 30 or Classic 80 speaker and a mic.

The mic is either an SM57 or a custom-specified mic supplied by Demeter. From there the signal gets treated to an API mic pre with EQ and delivered straight to analog tape.

Yngwie Malmsteen
Much of Yngwie Malmsteen’s expressive phrasing technique can be attributed to his scalloped-fretboard Strat.

The increased distance between the string and the fretboard that the scallop creates allows him to control his vibrato to a great degree. By pushing down on the string and pulling it from side to side, Malmsteen creates some of the most expressive notes around—especially when he employs nearinfinite sustain.

That sustain is created by two maxed-out 50-watt Marshall heads, one outputting a dry sound, the other outputting the effected sound.

Note that one amp feeds a cabinet wired at 16 ohms, while the other feeds two 8-ohm cabs in series, so that the speaker output is equivalent.

Malmsteen is one of the only guitarists to use a sonic enhancer (a BBE Sonic Maximizer) in creating his sound, which he credits with adding a little more sizzle and definition to his top end.

Steve Morse
With his three-pickup Ernie Ball Strat hybrid and an arsenal of effects and amps, Steve Morse is ready for any kind of sound—fusion, new age, blues rock, hot country, and classic metal (for when he plays with Deep Purple).

Morse first goes into a pedalboard, which has two boost switches: one feeds his Peavey VTM-100 amp, and the other goes into his Mesa/Boogie TriAxis preamp.

He uses a series of volume pedals to bring various elements in and out, such as the synth guitar volume, the delayed sounds generated by the various Lexicon devices in his rack, and an arpeggiator or clock from his Lexicon PCM42.

Morse favors Peavey 4✕12 cabs for his straight guitar sound, but also has two full-range three-way speakers for his acoustic and synth outputs.

Joe Satriani
Joe Satriani sets up his signal chain in a fairly orthodox manner: wah first, distortion second, and time-based effects after that, before finally going into the front end of a Marshall 6100 head, with 6550 power tubes substituting for the more common EL34s.

Satch’s setup is suspiciously Hendrix-like, having the wah in front and then a Fulltone Ultimate Octave later on in the chain.

But Satriani also makes expert use of delay (something Hendrix didn’t experiment with much), using three different delays.

First is the Boss DD-3, which feeds into two Chandler Digital Delays in series.

Steve Vai
Everything starts out normally enough in Steve Vai’s rig, with a distortion pedal, wah, and whammy pedal, but a switching controller steps in to turn this setup into something ingenious and unconventional.

The switching system selects between the various time-based effects in the rack while sending the pedaldriven signal to the amps’ inputs.

The amps’ effects loops bring in the effects via the send and return jacks, and the amps’ slave outputs go into two VHT power amps.

Eddie Van Halen
He’s come a long way from the days when he would just plug a “Frankenstein Strat” into an MXR Distortion+ and a Phase 90. Actually, the Phase 90 is still in his rig, but Eddie’s setup has become a little more sophisticated.

He gets all of his distortion from the amps, whether Marshalls or Peavey 5150s.

The amp switcher can select the path between the Marshall and Peavey, and the timebased effects (which include Eventides, Rolands, and Lexicons) all come via the amps’ effects sends.

After going from the amp preamps to the effects sends to the effects rack, the signals are not returned, so the power amp section of the Marshalls and Peaveys never gets used.

Instead the signals are sent to three pedals (a Cry Baby wah, a Boss OC-2 octaver, and the abovementioned Phase 90) before going into a rack containing speaker simulators and power amps.

The speaker simulators are necessary to take the high-end edge off of the line-level signals from the Marshalls’ and Peaveys’ preamp sections.

After being simulated, the line-level signal is delivered to the power amps and sent to three 4✕12 cabinets.

Stevie Ray Vaughan
He was a little bit blues and a little bit rock and roll. Stevie Ray Vaughan’s setup gave a nod to Hendrix, with the Vox wah in front and the Diaz Square Fuzz and Tycobrahe Octavia in the chain.

Vaughan also employed a Boss chorus pedal and a real Leslie before driving an armada of Fender-made amps.

Vaughan also played through a Marshall configured with 6550 power tubes instead of EL34s.

Carl Verheyen
Studio ace Carl Verheyen uses a combination of devices from pedals to rack-mount gear, but he essentially runs two paths: clean and distorted.

For his distorted sound, he goes through his pedalboard (which contains such front-end devices as a Cry Baby wah and an Ibanez Tube Screamer) and then into either a Marshall head or a THD modded Plexi.

The speaker outs of both heads are run through a load box (the THD Hot Plate) to convert them to line level.

At this point the signals are carrying all that wonderful amp distortion, both from the preamp and power amp stages, but are at line level, where they can be further processed by a Lexicon PCM41 before being amped up by a solid-state power amp to drive the speaker cab.

Verheyen approaches his clean sound a little differently. After the pedalboard, Verheyen first goes through a tube-driven

Fender reverb unit, which provides a gain boost. He then puts the signal in mono through a Chandler Digital Delay and takes the output through a stereo delay (mono in/stereo out).

The right and left outputs each go to matched Vox AC30s. Verheyen considers his clean sound, as outlined here, to be one of his trademarks.

To acquire “The Recording Guitarist” from Backbeat Books, click over to www.musicdispatch.com.

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StagePro Provided JBL VERTEC Line Arrays & Crown VRACK For LDI Show Awards Ceremony

The response to the Awards Ceremony was very positive and the audio system performed admirably..

Rental sound, lighting and staging production company StagePro provided the live sound reinforcement system for the LDI Awards Ceremony, held at the recent LDI trade show in Las Vegas.

The Awards recognized excellence in a variety of categories related to the live design, events and staging industry, including lighting, audio and video.

The audio system featured a comprehensive range of products from Harman Professional, highlighted by JBL’s subcompact VERTEC line arrays.

For the event, StagePro supplied the audio reinforcement system as part of its Apex Mobile Stage.

The main PA system included left/right hangs of four VERTEC VT4886 subcompact line array loudspeakers and two VT4883 subcompact arrayable subwoofers per side, supplemented by an additional four VT4886 loudspeakers placed along the center edge of the stage for front fill.

“The response to the Awards Ceremony was very positive and the audio system performed wonderfully,” said Jay Waller, Owner of StagePro.

“The subcompact VERTEC line arrays were a terrific solution for this event, which provided an opportunity to expose this relatively new product line to a variety of potential clients.” 

A new Crown VRACK loaded with I-Tech HD amplifiers powered the system, which also included a Soundcraft Vi4 digital mixing console along with AKG wireless microphones.

“The Vi4 console performed flawlessly,” Waller said. “We received a lot of comments that the effects package incorporating other Harman technologies like Lexicon is very impressive, and it truly is.”

“Of all the digital consoles on the market, it has one of the nicest sounding effects packages available.”

“The LDI Awards are a great way for the show to recognize excellence in the industry, from the Wally Award in memory of Wally Russell honoring a veteran in the lighting industry, to the Redden Awards in memory of Craig Redden and presented by Epic Production Technologies to honor excellence in lighting in theater, concerts, and corporate events,” said Ellen Lampert-Greaux, Consulting Editor and Conference Director, Live Design/LDI.

“The awards also include the ESTA Members Choice Product Awards and LDI’s Best Debuting Product Awards, which honor technical innovation. The LDI booth awards accented the excitement on the show floor.”

“The final award, sponsored by Showman Fabricators, was for the best Green Product Award in recognition of the greening of our industry. We appreciate the support of StagePro and Harman/JBL in making the awards ceremony a successful event.”

JBL Website

Crown Website

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Tech Tip Of The Day: Recording The Low Frequencies Of Acoustic Guitar

Would a change in microphone help me to better achieve the sonic image I'm looking for?

 
Q: I’ve been recording some acoustic guitar lately and have been having some difficulty.

The problem is that I just don’t seem to be getting the low-end reproduction I was hoping for.

I know I can probably boost this later with EQ, but is there something I can do?

Maybe a different mic technique?

A: There’s no doubt about it, recording an acoustic guitar can be tricky.

There are several ways to bring out more of the bottom end fullness you’re looking for.

One solution — assuming your guitar has a pickup — is to record the pickup output via a DI at the same time as you record the microphones.

When it is time for mixdown, filter the pickup signal with a lowpass filter, so that all that is left is the low-end.

Shape this further with EQ, perhaps add a dash of compression, and blend to taste with the miked-up guitar tracks.

Go lightly; you’ll find that it doesn’t take much of this acoustic guitar “subwoofer” track to fill out the bottom end.

This trick is especially useful with detuned, alternate-tuned, and drop-tuned acoustics.

However, if you use this method you should double-check that you aren’t creating phase issues by mixing the direct pickup signal in with the miked signals.

In many cases the DI will arrive slightly before the mic signals; use your ears to check whether this is affecting the mix or not. If so, slide the DI track slightly in time until the phasiness clears up.

As you mention, changing the microphone is another possibility. You’re best bed in that instance is to compare the frequency responses between the prospective microphones.

Those which offer a more of a low-end boost are likely what you’re looking for in this instance.

As always, we welcome input from the PSW community and would love to know how you would handle the situation at hand. Feel free to let us know in the comments below.

 
For more tech tips go to Sweetwater.com

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GefenPRO Announces 10x4 Dual Link DVI Matrix

The matrix supports output from ultra high resolution computer systems

Gefen has announced the availability of its 10x4 DVI DL Matrix, a new addition to the recently introduced GefenPRO product line.

GefenPRO offers integrators and end users professional products with advanced 24/7 technical support for broadcast, rental/staging and post-production environments.

With the 10x4 DVI DL Matrix, any ten computers can be switched to any four professional monitors with full cross-point routing.

Users have the flexibility to create one or two extended desktops or send sources to individual displays.

High resolutions are supported up to 3840x2400, accommodating both single and dual link DVI formats.

Built-in EDID management ensures the stability of pre-settings throughout switching. RS-232 serial control and front panel buttons offer easy access and control.

An internal power supply further strengthens performance, eliminating potential disconnects. The black metallic enclosure is rack mountable and rugged enough to withstand high performance environments.

Gefen Website

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Auralex To Present Class A Fire-Rated StudioFoam Pro At AES 2010

StudioFoam is the first melamine-free Class A fire-rated acoustical foam in the indusrty.

Auralex Acoustics is showcasing its new Studiofoam Pro, the industry’s first melamine-free Class A fire-rated acoustical foam, at the upcoming 2010 AES Convention in San Francisco (Booth 729).

Studiofoam Pro presents a low-cost option for sound absorption in studios, churches, restaurants, clubs and any other venues that requires a Class A fire-rated acoustical treatment.

The absorption level has not been tainted by creating acoustical foam that is Class A fire-rated. In fact, the foam maintains the same cell structure, but the cells are smaller, condensing the thickness of the panel and provides superior performance to a standard 1-inch thick fiberglass panel.

Studiofoam Pro is available in two sizes: 2-foot x 2-foot and 2-foot x 4-foot, both 1.5 inches thick. It is available in charcoal gray and features beveled edges and provides a Noise Coefficient Rating (NRC) of 0.90.

“Auralex is proud to expand on our Studiofoam product line and to introduce another industry landmark by bringing melamine–free Class A fire-retardant acoustical foam to market,” says Eric Smith, founder and president of Auralex Acoustics.  “Studiofoam Pro is the perfect combination of appearance, pricing, flame retardancy and the renowned physical characteristics of Auralex’s best-selling Studiofoam.”

Many public buildings and local fire codes require that acoustical materials be Class A fire-retardant. Until now, Class A rated acoustical treatments included basic cloth-covered fiberglass and a special flame-retardant type of foam called melamine. Melamine can serve as an absorber, but it is not only expensive, its physical attributes make it very sensitive to physical damage.

Due to Auralex’s proprietary chemical formulation, Studiofoam Pro has all the traditional benefits of Auralex’s industry-leading Studiofoam, including outstanding durability due to reduced oxidation. Studiofoam Pro will not rot, crumble or suffer surface harm from normal use like other reported Class A alternatives.

Studiofoam Pro is Class A rated according to ASTM-E84, which evaluates flame spread and smoke density. It passed the UL 94 HF1,2, which is a fire-rating test specifically for nonstructural foam materials like acoustical foam.

It also passed MVSS 302, a fire test requirement for materials installed in motor vehicles. Passing the MVSS 302 is significant for product installation in RVs and media vehicles. The last test that Studiofoam Pro passed is the California Technical Bulletin 117 (Cal 117), which contains an open flame test and a smoldering cigarette test.

Auralex Acoustics Website

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Sound Devices USBPre 2 Is Now Shipping Just In Time For AES 2010

Balanced XLR outputs are switch-selectable between mic or line level; consumer RCA-type output is available for connection to unbalanced input.

Sound Devices USBPre 2 is now shipping just in time for AES 2010.

The two-channel USBPre 2 offers a powerful, easy-to-use portable interface to interconnect audio sources to Mac OS and Windows computers over USB.

Featuring an entirely new electronic design, it uses the same extended-bandwidth, low-noise microphone preamplifiers and digital converters as Sound Devices 7-Series digital recorders.

The USBPre 2 fits a broad range of applications, including voiceover recording, reference playback and monitoring, and test and measurement.

The class-compliant, plug-and-play device accepts mic level, line level, consumer line level and SPDIF digital (coaxial or TOSLINK) inputs. Its microphone preamplifiers have selectable analog limiters, high-pass filters, 48 V phantom power, and high-resolution LED meters. Because the USBPre 2 draws its power solely from the computer’s USB port, no additional power source is required.

Sound Devices designed the USBPre 2 for both reference quality input and output. Its balanced XLR outputs offer superior rejection to interference and are switch-selectable between mic or line level. Additionally, a consumer RCA-type output is available for connection to unbalanced inputs. Its headphone amplifier easily drives full-sized headphones with extensive, clean gain.

With its unique stand-alone mode, the USBPre 2 functions as a two-channel microphone preamplifier with analog, digital and headphone outputs. Stand-alone mode is perfect for applications that require an easy-to-use, quality microphone preamplifier. A built-in high-resolution LED level meter helps further facilitate these types of applications.

“The original Sound Devices USBPre was the very first bus-powered USB interface with phantom-powered mic inputs. The convenience and flexibility it offered proved quite popular with sound engineers,” says Jon Tatooles, managing director for Sound Devices.

“The USBPre 2 is an all-new update of that original, with powerful new features and superior audio performance. It is the perfect portable audio interface for Avid, Apple Final Cut Pro, and Adobe Premiere Pro editing systems. System engineers will also appreciate its ability to output full line level signals on XLR connectors.”

Sound Devices Website

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Thursday, October 28, 2010

Tannoy Loudspeakers & Lab.gruppen Amplifiers Installed At Cannery Row

The Monterey California eatery chose Tannoy speakers and Lab.grupper amplifiers to set the energetic tone during a recent renovation.

Widely known as the setting of John Steinbeck’s 1945 novel, Cannery Row, and his 1954 sequel, Sweet Thursday, this historic area of Monterey occupies a unique niche in the nation’s cultural consciousness.

In the opening sentence of the former Steinbeck describes Cannery Row in dramatically contrasting terms, calling it ‘a poem, a stink, a grating noise, a quality of light, a tone, a habit, a nostalgia, a dream.’

Today the area is a major tourist draw and marine sanctuary, home to a large population of California sea lions, an increasing number of hotels, restaurants and some few fishing companies that add a layer of modern authenticity to what is an increasingly popular entertainment hub.

Although the Cannery Row of Steinbeck’s day collapsed along with the Monterey Bay fishing industry in the 1950’s, the area is now populated by a different sort of angler; people fishing for cold beers, good eats and good times.

Situated in an historic former Cannery and a one-time brothel the Cannery Row Brewing Company aims to deliver just that. Offering up seventy-three brews on tap, an extensive list of bottled beers and ciders, thirty brands of small batch bourbons and the talents of Executive Chef, Mark Ayers, formerly of the Highlands Inn and Pacific’s Edge restaurant and now culinary director of the Annual Pebble Beach Food and Wine.

“It’s a gorgeous area,” says Nathaniel DiMaggio, project manager for Coastal Luxury Management, the developer of the project. Although it does lack some of the hipper variety of establishments more commonplace in larger cities the Cannery Row Brewing Company will help fill that hole, DiMaggio believes.

As well as appealing to a wide demographic with a combination of vintage industrial style, great food and beer fueled fun in one of the buildings allegedly actually mentioned in Steinbeck’s book.

Beer fueled fun and a reasonably comfortable dining experience don’t always go hand in hand, however. That was the motivating factor behind the choice of a suite of Tannoy V Series and CMS in-ceiling loudspeakers powered by Lab.gruppen C Series amplifiers for the gastro-pub’s audio system.

DiMaggio’s primary concern was clear, intelligible playback of program music in both the Cannery’s restaurant and bar areas. A system that would provide maximum intelligibility and evenly distributed sound, zoned to allow a degree of control over in each area, with enough power to allow staff to provide high energy music when necessary.

“We wanted the whole place to rock if need be,” DiMaggio says.

“It’s a big restaurant,” Gianetta says. “They needed something more substantial than your typical background music system loudspeaker. So I pushed him in the direction of buying a pro sound reinforcement type of loudspeaker and a large subwoofer.”

“Tannoy’s proprietary Dual Concentric driver has benefits in every application and the trend is more and more towards deploying these kind of devices,” he continues, be it in a concert setting or club application.

“Right out of the box the Tannoy performs well. You don’t have to do a lot of equalization, and they have symmetry in both the horizontal and vertical axis, where drivers that aren’t coincident don’t.”

Though DiMaggio had some experience with audio technology he wasn’t sure what amplifiers were best suited to the build so he went with Lab.gruppen based on Gianetta’s recommendation.

“The thing that’s cool about Lab’s C Series is that they’re versatile and can do 70 volt and low impedance on the same amplifier on different channels, which was great. And they’re super efficient both for heat and electrical usage.”

“We think the Lab.gruppen C Series are vastly superior to the other amps that are out there,” Gianetta said. “They allow you to set what kind of load you’re going to put on the amplifier per channel so you can run low impedance load or high impedance loads, because it has selectable VPL (Voltage Peak Limiter) that sets the peak voltage on the amplifier.”

The system consists of one Tannoy VS 218DR sub mounted in a corner over the bar on a custom steel bracket, covering the entire 4000 square foot bar/restaurant. In addition, three Tannoy V8’s are mounted in ‘U’ brackets on ceiling beams in the main bar area, angled so bartenders aren’t obliterated when the system is cranked, and driven by a Lab.gruppen C28:4X.

The remaining V8’s are distributed evenly throughout, covering the main dining area and entryway and driven by a Lab.gruppen C20:8X. Sound reinforcement for customer washrooms and the hallway adjacent to them is provided by five Tannoy CVS 4 in ceiling loudspeakers.

A number of third party loudspeakers are situated in an outdoor seating area built around three large fire pits. The system also includes Rane and SurgeX components, Crestron System Automation and Autopatch Video distribution.

While DiMaggio describes the Cannery as ‘sports oriented’, it is not, strictly speaking, a sports bar. Sporting events will be in heavy rotation on eleven Panasonic flat screens tied into the audio system, but only special events like the FIFA World Cup or Super Bowl will be broadcast on the main audio system.

Time on the build, DiMaggio stresses, was a commodity that was in very short supply. “We started in February and wanted to hit summer. So it was a year-long project crushed into three or four months.”

All system components were supplied by Oakland based Leo’s Professional Audio, who service, among others, A-list clients like George Lucas, as well as providing design/install services of A/V, lighting and acoustical solutions for a variety of churches, schools and sporting venues.

Although Graham Cooper, VP of Leo’s Professional Audio install division, regularly specifies Tannoy and Lab.gruppen for his own projects, this time out his mandate was very basic. “Pretty much, ‘“here’s what we need and we need it NOW,” he says with a laugh, adding that Leo’s Professional Audio technician, Gordon Fava, also worked on the Cannery install, loading and terminating the rack.

For DiMaggio’s part, he had never heard of Lab.gruppen or Tannoy prior to this build, he explains, but is so pleased, he intends to use Tannoy and Lab.gruppen again in an upcoming project.

Tannoy Website

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Amped Up: MOSFET Or Bipolar?

Is one amplifier technology better than the other? Why?

Do you really care if the amplifier you are using is a MOSFET (Metal Oxide Semiconductor Field Effect Transistors) or Bipolar design?

Probably not, at least as long as the amplifier continues to perform.

With modern amplifiers, few people notice the difference between the designs until the amp fails and they receive the repair bill.

That’s when most people take a special notice. Keeping that in mind, let’s walk through some of the advantages, disadvantages and myths of these two designs.

Back On The Loud Frontier
For many years now, designers and users of professional amplifiers have had the same discussions (arguments) regarding which output devices sound or perform better.

Unfortunately, these discussions have created more myths than factual statistics.

When talking about Bipolar or MOSFET designs, we are usually talking about the output stage of an amplifier.

The output stage can be compared to the engine of a car. The output stage provides the horsepower to the speaker.

The differences between the two design approaches for this topic have very little to do with front end designs or power supplies.

Engineers of the 1960s and 1970s were frontiersmen of high power silicon for audio. Most of their success in amplifier design was based on the guinea pig method. Stand back and watch for smoke!

They didn’t have many textbooks for amp designs or the Internet to print off the latest findings.

Their designs were not always successful due to unproven engineering and inadvertently created modern misunderstandings about MOSFET and Bipolar transistors.

Bipolar designs have been around since the 1960s, when silicon evolution led us astray from the common tube designs. The majority of professional amplifiers in the market, past and present, are of the bipolar topology.

The sound quality of a bipolar design is what we have become acquainted with. Unfortunately, this does not mean that all bipolar amplifiers sound good.

Conventional Amplifier Wisdom
MOSFET amplifier designs are known to be more musical than Bipolar designs, especially in the mid to hi frequencies.

Bipolar designs are known for their ability to deliver high current into various loads.

This is good for low frequencies (LF).

As such, it was once believed that the ultimate sound system would be to use MOSFET amplifiers for the MF/ HF and Bipolar amplifiers for the LF.

Why is this? Is it still true today?

To understand the differences between the two topologies one must have an understanding of the history of theses devices and their designers.

Have these devices improved through the years? Of course they have! Have the engineers improved their designs through amplifier failure? Certainly!

The greatest amplifier design advancements have come from designers and not necessarily with the components.

Sure, many more manufacturers are supplying us with a better selection of these devices. However, the real change for today’s engineer is proven circuitry available for their reference.

MOSFETs are a relativity modern addition to the family of power transistors. They were introduced by Hitachi for audio applications in 1977. A MOSFET transistor consists of three elements, Gate, Source and Drain.

A MOSFET transistor in its simplest terms works like a water valve. When voltage is applied to the Gate, the valve opens and lets current flow from the Source to the Drain.

The early problem of these new devices was the Drain to Source internal impedances were high resulting in low Damping factors.

In those early days of the Hitachi device, the internal resistance of MOSFETs was greatly reduced to yield LF performance that rivaled Bipolar LF performance.

This gave us the impression that MOSFETs cannot produce good LF. As the old saying goes, “it only takes a minute to create a good impression but a lifetime to overcome a bad one”.

Big Drivers Required
Bipolar designs have the ability to deliver enormous amounts of current to a load. A Bipolar transistor consists of three elements as well, Base, Collector and Emitter. The current path is from the Collector to the Emitter.

There is also a significant amount of current flow from the Base to the Emitter. Bipolar amp designs require a hefty drive stage.

These devices require additional heat-sinking because the Base of each output transistor is current driven not voltage driven, causing them to heat up.

MOSFETs also require a drive stage, but have much less of a load requirement compared to a Bipolar driver stage because they are voltage driven, not current driven.

Bipolar transistors are positive temperature coefficient. When a transistor passes power, it heats up.

When a Bipolar transistor gets hotter its internal resistance decreases and tries to pass more power making it even hotter.

Because no two individual devices are identical, a positive coefficient means that one transistor tries to “hog” the load and tries to do all of the work.

To prevent this, large Emitter resistors are placed in the current path to help equal the sharing of the load.

If the heat is not controlled by temperature sensing devices thus reducing the drive of the output stage, thermal runaway will occur, thus blowing up the device.

The real “shocker” comes with failure. When a Bipolar fails, it tends to short, connecting the supply rail directly to the output.

When this happens you will experience the catastrophic effect causing all devices in parallel with the bad device also to short.

Have you ever experienced the “your amp went DC” diagnosis? Lets hope not! If you haven’t, your lucky, because when this happens whatever is connected to the output (i.e. loudspeakers) is also history.

In most designs there are DC sensing circuits incorporated to shut the channel or amp off. In most cases, however, these circuits aren’t fast enough to prevent serious speaker damage caused by thermal runaway.

How Can Negative Be Positive?
MOSFET Transistors are negative temperature coefficient. When a MOSFET heats up the internal resistance increases causing it to pass less power.

MOSFET amplifier designs are inherently thermally stable without additional circuitry and by nature has no chance of thermal runaway.

Each output device in effect self regulates to carry the load equally with all of the other output devices. When a MOSFET fails it usually opens, so there is rarely a chance of damaging a speaker from output stage failure.

This means there is no real need for additional thermal tracking and DC sensing circuitry.

Many designers that feel MOSFETs, if not matched properly, may be unnecessarily paralleled without the use of large (physical size) source resistors to equalize the current draw between devices.

As such, MOSFETs have no advantage over Bipolars in relation to space and cost reduction of expensive Emitter resistors for Bipolar circuits and Source resistors of MOSFET circuits.

MOSFET amplifiers have a much higher slew rate than Bipolar designs. MOSFET devices are very fast and can switch several amps in nanoseconds. This speed makes them thirty to one hundred times faster than equivalent Bipolar devices.

If you are accustomed to listening to traditional Bipolar designs and compare them to the speed of MOSFET design, you will notice a significant, almost unnatural audible difference in the HF. Such clarity is a pleasant added experience.

Engineer Wayne Colburn of PASS Labs, a world leader in MOSFET amplifier design, stated,” there really are no major disadvantages of MOSFET devices”. When compared to Bipolar devices, there are only two disadvantages for concern, the first being cost.

MOSFETs can cost up to two-three times more than an equivalent Bipolar device. This is a huge concern, especially in high-powered amps where it takes several devices to achieve the desired power.

To achieve high output levels, a tiered power supply may be required to feed the front end several more volts than the output stage. This translates to additional cost for everyone.

Why Aren’t MOSFET’s Used More
If MOSFETs have more advantages than disadvantages over Bipolars, why are there so many more Bipolar amplifier designs in use? Cost is the biggest factor, but pricing is rapidly improving.

There is also a greater variety of Bipolar transistor manufactures to choose from, as well as packaging types, voltage and current selections. These areas are finally improving for MOSFETs as well.

Expect many more MOSFET designs to reach the professional market in the future.

Jeff Kuells is an audio engineer and audio manufacturing consultant and was previously director of engineering for a major amplifier manufacturer.

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Allen & Heath iLive DualRack Chosen For Kent Open Air Classical Festival

The iLive was chosen to mix performances of opera, musical theatre, a military band, and much more.

Kent-based PA company, SRD Group, was chosen to provide sound reinforcement for a large scale 16,000 capacity outdoor classical music event, selecting a range of units from its stock of Allen & Heath iLive digital hardware and using the latest ‘DualRack’ capability.

The varied program included opera, musical theatre, military band, traditional choral, and a headline performance by the Royal Philharmonic Orchestra conducted by John Rigby and accompanied by soloists Wynne Evans, Elizabeth Watts and Richard Morrison.

The evening was also narrated by actor Robert Powell, featured the marching band of the Brigade of Gurkhas, and included a fireworks and cannons finale.

“With crowd coverage extending 365m deep x 100m wide, and due to the predominantly orchestral material, audio requirements reached 180 inputs. To compound this issue, the FOH footprint needed to be very compact due to crowd sight line issues,” said SRD’s MD, Stuart Roberts.

“iLive was the obvious choice, as its flexible, distributed audio architecture makes it easy to build high capacity systems within a small footprint.”

The Royal Philharmonic Orchestra was mixed on a separate system by specialist classical sound engineer, Ian Barfoot, who provided 4 stereo subgroups to both the FOH and Monitor consoles, which took pressure off the engineers and provided a better audio balance within the Orchestra.

The main FOH system consisted of an Allen & Heath iDR10 modular MixRack loaded with 64 inputs - 56 analogue and 8 digital – connected to an iLive-80 Control Surface with a mixture of analogue and digital local outputs.

There was an EtherSound digital split to the Monitor system, which comprised an iDR0 miniRack connected to an iLive-112 Control Surface with 24 analogue local outputs.

The orchestra system included of two iDR10 MixRacks loaded with 128 inputs - 120 analogue and 8 digital - running in ‘DualRack’ mode, feeding 16 digital outputs - 8 digital feeds to the FOH system and 8 to locally required outputs - with 16 analogue outputs as back up.

All the MixRack and Control Surface connections, except the monitor system, were via Gigabit Network Smart Switches with fibre optic interfaces over deployable tactical fibre cable.

WiFi networks were installed on all of three iLive systems to allow remote access to control functions via iLive’s Editor control software. All the iLive systems and network interfaces were powered via UPS units for protection from variations and interruptions in the temporary AC supply.

“Everything worked according to plan,” said Barfoot. “iLive is still one of the best sounding, most intuitive digital desks around, which is why it was chosen. With the added flexibility of ‘DualRack’ mode, the channel count is doubled per system without compromise, making the choice of iLive even easier.”

“After the event we received a lot of compliments from the artists taking part who remarked on the quality of the monitor sound, and to the best of our knowledge there were no complaints from the crowd regarding any issues with sound or coverage.”

“We would definitely do it the same way again - the combination of good input sources with an excellent mixing console makes every ones job much more simple.”

Allen & Heath Website

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Worship Wisdom: Finding The People While Learning & Growing

Choosing and training the Church sound team.

I am often ask variants on this single question. What characteristics do you look for in a potential member for your sound team?

Should you look for a frustrated musician? A rocket scientist? A computer geek? A telephone lineman?

Maybe, or maybe not. Attitude is usually more important that pre-existing aptitude.

In this article, we’ll first examine how to identify the proper individuals to serve in your technical support ministry. We’ll also show you how to train them to achieve technical excellence.

A Servant’s Heart
After serving on the production staff at churches for nearly ten years, I’m here to testify you that you never want to choose a person based solely on their technical knowledge.

Instead, look for someone with a willing heart first. Then ask about their technical knowledge.

Why not look for technical knowledge first? While both are important, technical stuff is easy to teach over time.

Finding someone with a servant’s heart can be more difficult. It’s part of their core personality.

It also illustrates their relationship with God and predicts their ultimate utility within your tech support staff.

Serving in a church ministry requires a boatload of grace and more patience than many people have left at the end of a busy week. In some churches, it means working with difficult people every weekend.

The worship team and tech support team depend on each other’s gifts to be at full muster at the downbeat of the service. The process is much like preparing a weekly meal for all your worshipers.

The worship team and tech support team need to be in unity before, during and after the service. Being on the same page, spiritually, is the key ingredient to this recipe.

Staying F.A.T.
One principle I’ve relied on over the years is that anyone involved in tech support ministry needs to be F.A.T. ­ faithful, available, and teachable, in that order. Once they’ve joined the tech support team, these people must also be faithful to be there when they’ve promised to be there.

Most of our lives are too busy. Many people over-schedule our arrivals and departures to the nanosecond. But as a wise friend of mine once suggested, the only way we can be somewhere on time is to arrive there early.

The volunteer should also make themselves as available as is practical. To say they’re committed to the success of the ministry, but then to only make themselves available for one monthly service doesn’t work well in most situations.

Only operating a console one time a month isn’t often enough to become proficient at it. Would you climb on that airplane next weekend if you knew that the pilot only flies once a month? Granted, I’ve never heard of anyone dying from a bad mix, but you get my point.

There is another side of this issue, however. I’ve seen some volunteers make themselves too available, to the point that their relationship with their family starts to suffer. If you get your priorities out of line, your work in that ministry will.

Profiles & Personalities
These days, it’s common to find people who work in, on, or around computers, volunteering to serve in the tech support ministry.

Musicians who love all things electronic are another fertile source of tech support volunteers.

My friend, Blair McNair, worked on missiles while he was in the Navy.

At some point he started volunteering in the sound team at his local church.

Years later he became the Technical Director for Benny Hinn at Orlando Christian Center, and today designs sound systems for a living.

Most volunteers do something else for living. Your church may be blessed with a seasoned audio pro as the volunteer head of the sound team, but that’s not the norm.

This is why any successful volunteers must be clearly, consistently teachable.

This means to say that in the likely event that a particular volunteer doesn’t make his/her living in pro audio, they need to make a committee effort to learn the craft so they can reliably deliver technical excellence in every worship service.

I’m unconvinced that there is any one type of personality to look for. That’s because I don’t think we need assume that every sound team volunteer must be able to drive the FOH mixing desk.

The individual who typically seeks involvement in a tech support ministry has a detail-oriented personality. These folks make lists for everything.

I have a detail-oriented personality. Knowing that the guitarist is going to take a solo on the third chorus isn’t enough. I want to know what kind of sound he’s going to use, and how loud he will play. I want to know if he’s going to start out soft and build to a loud ending.

I must know if he’s going to use his own effects, or if I should plan on adding some echo effects on my own. Notice that I’m the audio guy, so I really don’t care what he’s wearing that day ­ that’s for the lighting guy or the programming director to think about.

Musical Background
This is one debate that has gone on for years and years. Should the person who will be driving the FOH mixing desk be a trained musician?

It’s easy for me to say yes, because I made my living as a player for twelve years, and I have a Bachelor of Music degree.

Clearly, someone who has experience as a player or a singer can be respected and accepted more readily by the players in the worship band simply because of the common bond and similar background.

But I do know of very capable mixers who have no formal music background, just a love of the music. I think this decision has to be a very individual one.

But I think we can agree that not everyone should be behind a console. Some can put together a great mix without even breaking a sweat. For others, it’s just not their gifting.

If the interest is there, however, the art of mixing can be learned. It’s not something they’ll grasp overnight, but time and practice and listening analytically are great teachers.

I’ve trained literally thousands of church music pastors, sound team volunteers and technical staff in my workshops. Of all of those people, I can only think of two individuals who just never seemed to get it.

Being a part of the church tech support team isn’t for everyone, but the majority of those who seem naturally drawn to the ministry seem capable of learning and managing the task.

Gifting & Getting The Job Done
Mixing sound is just one of the tasks that the sound ministry is charged with.

You could also find people who are thrilled to do a good job of running the tape duplicators after each service.

Others might enjoy fixing broken mic cables.

Still others might be happy setting up the stage every Saturday night.

Perhaps there’s a self-employed someone who could carve out some time to set the stage or run essential weekday errands.

Someone with a theatrical background might enjoy serving as a stage manager, a runner, or in some other role.

If your pastor has a daily or weekly radio program, someone must learn to use your nonlinear editing software to edit those programs.

If you identify all the tasks that need to be accomplished during a week, and then spread them out over a handful of people, you should find that the job can get done with excellence and without anyone getting overly stressed.

In a large church, you’ll find a trained individual at every post. The FOH desk, monitor desk, lighting desk, in the TV control room, at the video projection desk, all require trained technicians. Still, in the majority of churches, one person may serve all of those roles simultaneously.

The best idea is to cross-train everyone who becomes part of the tech support ministry. The lighting guy should at least be able to get sound out of the system, and the audio guy should at least be able to get the stage lights up and running if needed. (Editor’s note: Which one do you suspect will do a better job?)

People need a weekend off. People get sick. Cars break down in transit. Your staff needs to be prepared to help out as needed, in season and out of season.

Why Train The Team?
We must recognize that there’s a great disparity between the tech support team and the worship team in most churches.

Think about it. Every worship team member, who sings or plays, has inevitably studied music at sometime in his or her life.

Even if they are self-taught, they’ve invested their time and managed to learn how to play.

North American culture has given us easy access to musical training.

Most public schools have some form of music program.

I began to play music when I was in elementary school, played in various music groups all the way through college, and made my living playing in bands until I was thirty years old.

It was only after I got my music degree that I quit playing music for a living.

Even if we didn’t pursue music as our lifelong ambition, our studies helped us in numerous ways.

In contrast, the tools or programs to learn how to run sound, or the stage lights, or work with video hasn’t had the same kind of easy access, at least not until very recently.

After all, in school, I played a saxophone. I didn’t need a sound system. Maybe you played in the brass section, and they really didn’t need a sound system either.

So, is it fair to compare the talents of a stage full of trained musicians and singers with that of a beginning audio student? No, this is an unfair comparison or expectation.

In real life however, that is what many churches do every week. Predictably and unfortunately, some get frustrated and lose their cool in the process.

Training your crew also helps to strengthen their bond as friends and teammates. It can even enhance their self-esteem as individuals, giving them more confidence.

Where To Find Training
Churches all across the world are crying out for trained sound technicians. Strangely, only a very small percentage of these churches are willing to pay for that training. That’s one very clear reason you rarely see such training opportunities.

If you’re a eager student of audio, reasonably certain that you have your facts straight, and you believe you are ready to start training others, then do what all the rest of us who have trained others in audio have done.

Put together an outline to clearly and logically organize the materials and dig into the resource materials to gather your supporting information. Then gather up your courage and go for it.

I choose to organize the material according to signal flow. That’s an intentional approach. Understanding signal flow logic is key.

When I’m teaching someone to connect an amplifier, for example, and I see them connect the speaker cable first to the speaker, and then to the amp, I have them disconnect both ends and do it over again.

Obviously, this makes no difference to the signal itself and, because it’s an AC signal, it constantly reverses directions. In general, as you already know, audio signal flows directionally from the amplifier to the speaker.

One day, years after they’ve stopped calling me nasty names, they’re going to run into an exciting moment when five minutes before the downbeat of their Christmas Cantata, with 2,000 people out in the audience, their sound system stops working.

Suddenly, the success of the event falls squarely on their shoulders and rests in any audio team’s to troubleshoot and resolve the problem in a timely manner.

If the concept of signal flow logic is firmly ingrained into their thinking, they’ll be able to rest in their knowledge and resolve the problem quickly and efficiently.

Once, I had the great pleasure of visiting with Bill Johnson, Chief Audio Engineer for Kenneth Copeland Ministries.

As we were touring the facilities at Eagle Mountain Church, he shared with me that they require their tech support volunteers to attend a training session once a month.

Through a simple test, the audio team is divided into beginning, intermediate, and advanced groups.

The classes are taught by technical support staff. That is so cool.

Ultimately, it helps bring the entire crew onto the same page, and because it keeps everyone growing in their knowledge, so they can do an ever better job of supporting the technical needs of the worship services.

Source Knowledge
The Internet is overflowing with information about audio. Some of it is even correct. If you’ve been in audio for some time and you’re reasonably confident in your knowledge, then go ahead and explore.

Just be alert for the occasional piece of audio mythology. If you’re a beginning student, I encourage you to stick to the main information highway.

We strive to make our own ChurchSoundcheck.com a mythology free zone. Obviously, ProSoundWeb.com focuses on performance audio technology and works hard to ensure accuracy.

Believe it or not, you can trust comments that you may read posted on web sites by the major manufacturers. For example, you’ll find accurate, reliable information on sites by Rane, Crown, EAW, QSC, Allen & Heath, dbx, and others.

Online courses are available from the Sound Institute, and Syn-Aud-Con will begin offering online seminars later this year.

Wake Up & Smell The Silicone
Finally, I’d like to leave you with a wakeup call. Have you stopped growing in your technical knowledge? Have you stayed on top of the DSP revolution in regard to digital consoles, or are you letting digital know-how pass you by?

Even worse, are you a know-it-all? Are you the type of individual who figures that they know all there is to know about audio, or lighting, or video?

Let me suggest to you that one day, in the not too distant future, you’re going to find yourself left in the digital dust of some young kid who just figured out how cool audio is, who has never even touched an analog audio console and been raised on digital.

There’s so much new stuff in play these days. It is impossible to stay on top of every technological change, in every equipment category, but that’s no reason to roll over and ignore the digital revolution.

It’s cool to learn from the past, to apply micing techniques learned from the masters, for example. It’s not cool to have been mixing at your church for the past thirty years and to walk up to a new console one day only to discover that you can’t even locate the ON switch.

If you’re not achieving the level of technical excellence that you aspire to each week, maybe it’s not the gear. A simple lack of knowledge could be standing between your audio education goals and the reality you live with.

Fortunately,, technical stuff can be taught and technical savvy learned, but you must work at it. Likewise, your volunteers and tech support staff must work at it.

Stay on task. Read. Study, study, study. Attend trade shows, workshops and seminars. Subscribe to trade magazines. Buy technical books. Read and study some more. After that, go teach someone else.

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

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

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