Thursday, November 07, 2013

In The Studio: A Handy (And Inexpensive) Wireless Remote Control Solution

It's no fun to stop, take off your headphones, and walk back over to the computer
Article provided by Home Studio Corner.

If you’re like me, then a lot of your recording sessions in your studio involve you wearing several different hats.

For me, I’m a musician, so I’m always recording myself.

The problem is studios tend to be noisy. I like to get as far away from the computer and hard drive as I can. That means moving across the room.

Then the problem, of course, is that now I’m very far away from the computer. I have to do what I call the “recording dance,” where I scurry back and forth between the microphone and the computer.

This gets old really quick.

When you’re in the zone to record, and you’re feeling very creative and musical, it’s no fun to stop, take off your headphones, and walk back over to the computer to stop recording and set up a new take.

This is especially frustrating if you make a mistake two bars into the first song, and you have to stop everything and start over. You’ll find pretty quickly that you’ll lose that “zone” that you were in, and playing the music then becomes a chore.

There are a few possible solutions to this. Over the last few years, there have been at least a handful of wireless transport control products on the market.

Frontier Designs made one called The Tranzport. I don’t believe it’s for sale anymore, but it was essentially a wireless transport control that allowed you to start and stop playback and do a few other functions wirelessly.

Another solution is a very cool product from PreSonus called FaderPort. It’s great because it allows you to have volume control with the fader and also all the transport controls you need for recording. The one problem is that it’s not wireless. That’s not a huge problem. All you have to do is get a very long USB cable and place the transport next to you at the recording position.

Now you have the transport controls right there, within arms’ reach to start recording, stop recording, or do whatever else you need to do without having to get up.

The only problem with that solution is that you have to run a cable all the way across the room to the recording position while recording. Then you have to run it back to your mix position when you want to use the fader port for mixing and other things.

My Solution
Here’s what I do. When I bought my iMac, it came with a wireless keyboard. It’s not a full-size keyboard. It doesn’t have the number pad to the right-hand side, so it’s fairly small.

I used this for a while, but, if you do a lot of work in Pro Tools or any DAW, you know that there are shortcuts that you can use with the number pad on the right-hand side of the keyboard.

Since this wireless keyboard didn’t have that, I eventually broke down and bought a full-size USB keyboard for the iMac. That left me with a very handy wireless tool. Now, whenever I record across the room, I simply turn on my wireless keyboard and carry it with me to the recording position.

Since I know all the shortcuts, I can quickly and easily start and stop recording. If I mess up an intro in Pro Tools, I simply hit Ctrl-period. That stops recording and I can press Cmd-Spacebar to restart recording again. It’s very handy and saves me a lot of time, since I’m not bouncing back and forth from the chair, back to the computer, and then back to the chair.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner. Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.


Posted by Keith Clark on 11/07 at 08:25 AM
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Wednesday, November 06, 2013

Problem Solvers: Mics And Techniques For Challenging Situations

Creative solutions to the daily challenges are readily available

Whether it’s a band using amplified and acoustic instruments, a bassist switching between electric and double bass, a drum kit right next to an acoustic piano, a particular sound that needs to be isolated in the face of loud ambient noise – or, well, you name it – selecting the right microphone (and deploying it in a certain way) can make a big difference in attaining the desired level and audio quality.

With that in mind, let’s take a survey of several live professionals about some of their “go to” mics and techniques when encountering acoustically difficult environments and/or in working with combinations of instruments.

Acoustic Piano
Preparing for a show with pianist/songwriter Mark Cohn, veteran FOH engineer Tom Dube announced that he could set up the piano mics in 60 seconds. With that, he reached inside the grand piano, positioned two DPA 4061 miniature omnidirectional condensers on the metal ribs of the piano’s harp with magnetic mounts, guided the cables, and plugged them into the stage snake.

Dube notes that he usually positions the mics in specific locations to maintain the proper phase relationship with the piano’s hammers. He then quickly adds a Barcus Berry XL4000 pickup to the soundboard under the lowest strings, running to the preamp.  (“Maybe the process takes 90 seconds,” he concedes.) 

At the console, he generally adds some gentle limiting and applies a high-pass filter to the 4061s at about 180 Hz to reduce stage rumble. Additional equalization depends on the particular piano, consisting of a bit of midrange attenuation centered somewhere between 300 and 700 Hz, and perhaps a slight boost between 3 and 4 kHz as well as 8 and 10 kHz. The DPA mics are “remarkably consistent and useable,” he notes.

“X” marks the spots – FOH engineer Tom Dube’s positioning for piano mics. (Credit: Tom Dube)

The XL4000 pickup serves to reinforce low frequencies that are sacrificed for feedback suppression with the high-pass filtering. “I’ll dump a whole lot of 1 kHz and 2.5 kHz, as well as bump up a bit of 12 5 Hz to add the missing bottom,” Dube adds. Finally, he has the player hold some lower tones during sound check to determine the best phase relationship between the mics and pickup.

When talking with FOH engineer Nick Malgieri at the main stage of the 2013 Monterey Jazz Festival, he referred to the Shoepps MK4 and its “magical gain-before-feedback ratio” even before any EQ when used as piano overheads.

“I’m not sure if any other mic overheads would work as well on an acoustic piano on a live stage with a band,” he says. Because of higher stage levels and instrumentation, the MK4 output is typically blended with some combination of internally mounted DPA 4060s, AKG C414s, or an Applied Microphone Technology (AMT) M40 boundary mic attached to the sound board. 

A DPA 4061 with convenient magnetic clip.

Acoustic Guitar
Getting an acoustic guitar to sound like it does unamplified, only louder, is a perennial challenge, especially when the guitar is doing more complex tasks than strumming chords in first position.

Solutions over the years have included a sound hole cover to seal the instrument’s acoustic chamber and lower the odds of feedback, as well as magnetic pickups under the strings, contact and under-saddle vibration-sensing pickups with outboard processing, and various small mics attached to the guitar either by themselves or in combination with under-saddle sensors.

At small venues where the audience is silent and the performer is still, simply placing a directional mic close to the instrument – avoiding close proximity to monitors and mains and adding judicious equalization – may do the trick. 

A recent solution is the L.R. Baggs Lyric system. Designed to be mounted in the guitar, the system includes a full-range directional mic in a specially designed noise-cancelling mounting that attaches inside the instrument on the guitar’s bridge plate, a small rotary volume control and presence setting on the underside of the sound hole, and an end-pin jack containing sophisticated tonal circuitry – including compression, limiting, and EQ to output a balanced acoustic sound. 

The system allows freedom of movement for the performer, since the mic always stays in the same position. I tested the Lyric by installing it into a custom handmade acoustic by luthier Mike Kelly (of Goodyears Bar, CA), with good results.

Keith Sewell, a touring guitarist with Lyle Lovett and the Dixie Chicks (among others), has been involved with the system since it was in prototype in 2012, seeking an internal mic-only solution for performing in amphitheatres and other large venues that would sound as close as possible to an unamplified acoustic. 

Because every venue and stage differs, Sewell says the Lyric needs a bit of tweaking each night, but once dialed in, he states, “it’s the best sound I’ve ever had as far as a guitar pickup,” adding that there is no way he could get even the best condenser mic as clear and loud in a live setting.

Lovett’s FOH engineer, John Richards, finds the mic very stable and musical in the mix. In a full band setting, he uses a high-pass filter at around 100 Hz on the guitar.

Guitarist Keith Sewell in concert with the L.R. Baggs Lyric system. (Credit: Keith Sewell)

Other effective solutions I’ve run across include an AMT S15G mic, which has a gooseneck external cardioid condenser element that positions between 3 to almost 6 inches above the surface of the guitar using a specialized clamp for the body and a beltpack preamp.

The DPA d:vote 4099G acoustic guitar mic provides a supercardioid element on a gooseneck with body clip, and is used externally pointing toward guitar. DPA offers a variety of exchangeable clips for this mic so that it can be used on violins and other strings, brass instruments, piano, drums, and so on.

And, the Fishman Ellipse Matrix Blend combines an under-saddle pickup with a miniature cardioid gooseneck condenser mic that goes inside the instrument. 

Guitar Amp
With decades of touring and studio experience, Mick Conley has miked his fair share of guitar amps. His choice is often a Shure SM7,  more typically used in radio broadcast as an announce mic. Though by the specs the SM7 is cardioid, Conley cites its “really tight pattern that isolates so well” as a key reason it works as desired in this application. It also includes bass roll-off and midrange presence boost controls for further tonal shaping.

When he has the time at a given show, he moves the mic position a bit to find the “sweet spot,”  listening through the house system. When using the same guitar amps at every stop on a tour, he may also mark the best spot with a piece of tape.

Dube has a few favored mics for guitar amps, including the ubiquitous Shure SM57. He also sometimes selects a ribbon mic such as a figure-8 Royer R-121 or a beyerdynamic M160 hypercardioid – or if available, a Sennheiser MD409 dynamic supercardioid (currently updated to the e609 Silver).

He emphasizes that mic positioning is critical to maintain a correct phase relationship, adding “use your ears.” Depending on the guitar, playing style, and amp, finding the best place to point the mic between the edge and center of the speaker cone is also a matter of listening. 

Kick Drum
Also while at Monterey, I ran into FOH engineer Dunning Butler as he was returning from the on-site equipment area with a mic to solve an audio problem on an upcoming performance at a venue that’s dubbed “Dizzy’s Den” at the festival.

It’s a long, rectangular room, designed for county fair displays rather than musical performances. As an acoustic space, it could be accurately described as “tubby” since its dimensions tend to accentuate low-frequency energy coming from the stage and the loudspeakers – leading to a lack of definition for bass instruments and kick drum. 

Dunning placed the beyerdynamic M88 cardioid that he’d retrieved close to the front head of the 18-inch kick drum (without a hole cut in the head), since in his experience the tight pattern and “bright sound” of the mic allow him to capture the sound without additional “boominess.”

Conley too mentioned the M88 as well as the more recent TG-series equivalent as “one of his favorite kick drum mics” that he also finds useful on toms and guitar cabinets.

Further Solutions
Flutes don’t get all that much attention, but they’re actually quite common to musical ensembles of a variety of styles, and they present unique challenges.

Noted flutist Michael Mason has found a solution with a Countryman ISOMAX 2, which is available in omni, cardioid, and hypercardioid patterns. Specifically, he mounts the mic to the instrument with it’s specialized flute clip, and sends signal to FOH via a Shure wireless system.

“The ISOMAX 2 provides excellent response and the clip mount offers all the flexibility I require in order to position my embouchure, with the ability to adjust the clip position for many of the extended techniques I perform,” he notes. “I position the clip and mic onto various areas of the headjoint, but never too close to the lip plate. I use the mic without the windscreen because it enables me to capture a wide range of articulations and wind sounds.” He adds that he knows of several other prominent flutists who regularly use the ISOMAX 2.

On the recent Justin Timberlake world tour, FOH engineer Andy Meyer turned to a unique solution he’s developed in applying an Audio-Technica AE5400 cardioid condenser mic – normally for live vocals – on the top and bottom of the snare drum. “I’ve been doing that since I was like seven years old,” he jokes. “Seriously, you cannot beat the 5400 in that application, and I keep trying.”

For unobtrusive close-miking of acoustic instruments and other audio sources, engineer Nick Malgieri chooses the DPA 4060, also a miniature omnidirectional model. He finds that this lavalier-style mic retains the clarity and high-end response along with faithfully reproducing “the sound of the body of the instrument.”

At times, he uses the 4060 as a “contact mic” by taping it to the instrument, and since it’s an omni, it doesn’t exhibit proximity effect that can color the sound. In more esoteric situations, he’s even taped it to a target to reinforce the impact of an arrow hitting home – with the audio relayed to the console via a Sennheiser G3 or Shure ULX-D wireless system. 

In addition to piano overhead applications, Malgieri sometimes uses a Shoepps MK4 as a lectern mic and to capture acoustic guitars and other acoustic instruments. For picking up more distant audio sources, he finds that the output of the MK4 is much more transparent than typical shotgun mics while still achieving the necessary gain.

Flutist Michael Mason showing the Countryman ISOMAX 2 placement on his instrument.

Stage Setup & Isolation
Audio and piano technician Brian Alexander has spent many years behind the scenes, touring with Chick Corea and others. He focuses on the stage setup and the interaction between stage levels, monitors, and mics. 

When stage levels are higher, emphasizing isolation between instruments based on where they’re positioned relative to each other, or even using sonic barriers (“if the musicians will put up with it,” he adds) can lead to more control over the audio to the front of house and the audience. Careful positioning of the null zones of the mics is also important. 

Further, in working at Monterey with vocalist Bobby McFerrin and his multi-piece band, including instruments ranging from drums, electric guitar, and keyboards to acoustic guitar, dobro, and bass ukulele, Alexander was greatly aided by both the careful stage arrangement and the use of in-ear monitors to keep the stage level low. Mixed by Dan Vicari, the results for the audience were dynamic, with every instrument able to be heard distinctly whether the band was rocking or playing intimately.

With so many mics to choose from, each with a unique set of characteristics, creative solutions to the daily acoustic challenges are readily available. The best approach is to set aside time to try some of the ideas presented here, as well as to come up with novel solutions of your own.

To the extent that the musicians will work with you, experiment with the positioning of instruments, amps, and monitors so that better isolation can be maintained – giving you more control at front of house.  The audience will appreciate your efforts.

Gary Parks is special projects writer for PSW/LSI, and has worked in the industry for more than 25 years, including serving as marketing manager and wireless product manager for Clear-Com, handling RF planning software sales with EDX Wireless, and managing loudspeaker and wireless product management at Electro-Voice.

Posted by Keith Clark on 11/06 at 11:28 AM
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Church Sound: Alternatives To Using Y-Cables With Source Devices (iPod, Laptop, CD Player)

Avoid burning out the outputs of your components

Question: Will anything bad happen if I use a Y-cable on my CD player or laptop to hook its two outputs into one input on my mixer? I’m running out of channels.

Answer: The answer is yes, probably. If not immediately, then some time in the future. What happens is that most modern audio gear has a very low output impedance, typically under 100 ohms. This is great in that it can drive audio over very long cable runs while ignoring interference from light dimmer buzz and cell phones, but bad in that a short circuit will cause its output transistors to put out too much current and overheat, eventually killing them.

But here’s the crazy thing…if you’re running the exact same signal out both the left and right outputs of your CD player, laptop, or iPod, say from a mono sound track, then there will be no current flow between the left and right output stages and all will be well.

However, if you then play a music track with a lot of dissimilar info on the left and right channels, say from a split-track song with music on the left channel and guide vocals on the right channel, then there will be essentially a short circuit current between the left and right output stages. This is very hard on the CD player’s and iPod’s electronics, and they will begin to overheat internally.

So if you only play these backing tracks once in a while or for only a few minutes at a time, then the output stages may never overheat enough to burn out. However, play these same backing tracks for an extended period of time (perhaps 30 minutes) and you’ll probably find that one of the outputs of your CD player, laptop, or iPod has been burned out. Not a good day for your gear. 

The best way to combine two outputs into a common input is by using a box with special build-out resistors that limit this current. Whirlwind makes a box called the podDI that not only safely combines the two output signals from the sound source into a common input on the console/mixer, it also gives provides separate volume knobs so you can turn the left and right channels up and down in volume independently. (The podDI is pictured above with a Y-splitter.)

In addition,  it provides a balanced XLR output transformer that’s perfect for isolating the ground of your gear from the PA system and stopping that nasty power supply buzz that often occurs when using a laptop as a sound source that’s powered from its own 120-volt transformer.

You can buy one for about $75 from Full Compass Systems here.

Mike Sokol is the chief instructor of the HOW-TO Church Sound Workshops. He has 40 years of experience as a sound engineer, musician and author. Mike works with HOW-TO Sound Workshop Managing Partner Hector La Torre on the national, annual HOW-TO Church Sound Workshop tour. Find out more here.

Posted by Keith Clark on 11/06 at 10:12 AM
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LynTec Updates RPC Panel With New Circuit Breaker Technology & Controller

Version 2 accelerates the design and installation process as well as decreasing operating costs

LynTec has announced the availability of the RPC (Remote Power Controller) Series Panel version 2, designed to facilitate audio-video-lighting system build-outs by accelerating the design and installation process as well as decreasing operating costs.

The update combines the latest generation of motorized circuit breaker technology with a new and improved controller, offering more memory and a faster processor for real-time monitoring down to the circuit level and the capacity to use multiple control protocols simultaneously — now including sACN.

Engineered to protect and control installed entertainment AVL systems, LynTec’s RPC v2 provides increased memory capacity and faster internal processing. The controller is available across the entire RPC product line, enabling users to experience familiar point-and-click installation and setup with the company’s motorized circuit breaker panels, retrofit relay panels, and mobile power distribution panels.

The RPC v2 also includes two market firsts: the capacity to use multiple control protocols simultaneously and an emergency auto-on function to activate egress lighting via contact closure inputs from fire alarms.

“Before the update, using multiple protocols simultaneously within a single panel was very difficult,” explains Mark Bishop, president of LynTec. “With RPC version 2, we’ve revolutionized AVL power control by creating the ultimate, all-in-one panel for simpler operation and deployments, giving our customers faster processing times, more protocol options, and the ability to monitor current draws in real time on any browser-enabled device.”

The RPC controller has an onboard server that is accessible on a network or from any smart device as well as motorized circuit breakers that can be loaded per specific user requirements. The device also includes built-in auto-off and brownout features that allow controlled shutdown and restart of circuit breakers, optional sequential on/off at the circuit level, and the ability for third-party control directly from systems via TCP/IP, DMX, RS-232, sACN, or contact closures.

As of November 2013, customers ordering an RPC panel, RPCR relay panel, or RPCM mobile power distro panel will automatically receive the RPC v2 upgrade.


Posted by Keith Clark on 11/06 at 08:00 AM
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Tuesday, November 05, 2013

FOH Engineer Eddie Mapp Captures Paramore Live With sE Electronics VR1 Mics, IRF2 Filters

Voodoo VR1 ribbon mics and IRF2 filters for band’s guitar sounds

Eddie Mapp, front of house engineer with alt-rock/pop band Paramore, is using three sE Electronics Voodoo VR1 ribbon microphones and a pair of patented IRF2 instrument reflexion filters with the band on tour.

Paramore is currently playing arenas in Latin America before heading to Europe for a month in support of its self-titled fourth album, which made its debut at #1 on the U.S. Billboard 200 chart in April.

Mapp is using the VR1s on the electric guitar rigs in the live line-up, where three touring musicians supplement the three-piece band.

“Each of the three guitar players has a clean guitar rig and a distorted guitar rig; I’ve got the VR1s on the distorted cabinets as well as Taylor’s clean amp,” explains Mapp.

Although the three guitarists—band member Taylor York, his brother Justin York and multi-instrumentalist Jon Howard—prefer more traditional dynamic or condenser mics with a little faster reacting top end for their in-ear monitor mixes, says Mapp, “I like to tune my PA as a nice linear system, so it’s nice to have a flat microphone like the VR1 where I can place the high-end emphasis wherever I need it out front and the ribbon element really helps it sit in the mix with out over powering the vocal in the upper mid range.”

He adds, “I’ve always liked ribbon microphones and mainly used them more in the studio as room mics and for different applications. But I hadn’t used them a lot on guitar in the past just because of the lack of top end.”

The Voodoo VR1, however, corrects that typical ribbon microphone characteristic through the use of a patent pending mechanical device designed by Siwei Zou, CEO of sE Electronics that enables a flat response from 20 Hz through 20 kHz.

“The VR1 is super-sturdy, it takes EQ really well and I like the fact that it does go down really low—lower than I normally need for the show,” Mapp continues. “There are certain parts in the show where maybe the guitar does a little ‘dive-bomb’ and I take out the high-pass filter and push it up. It’s really fun, the way it just fills the room.”

Mapp also enjoys the low-mid push that the VR1 provides, especially when using them on outdoor shows. “I think perhaps because you don’t have the room pushing back against them, sometimes you don’t get the same feel of the guitars outdoors. But with the amount of usable low-mid in the VR1, outdoors I’ll sometimes leave a little bit more of it on the guitars, which just makes them sound gigantic.”

Mapp, who has also worked with bands such as Taking Back Sunday, Evanescence and Stone Temple Pilots, first heard about the Voodoo VR1 from two fellow touring FOH engineers: “It was Andy Meyer, who mixes Mötley Crüe and Justin Timberlake, and [Ken] Pooch [Van Druten], who mixes Linkin Park—they had both started using the VR1 and recommended that I check it out. I just fell in love with the thing.”

As for the positioning of the VR1 mics, he adds, “In the studio I tried miking the cabinet dead center and then on the edge of the speaker. At first I really liked the sound just on the edge of the dust cover, but I found that in context with everything it cuts a little better right in the middle.”

Mapp uses z-bars to position the VR1s on each cabinet, and has adapted hardware to allow him to also position IRF2 filters on two of the rigs. “The two upstage guitar players both have rear-facing cabinets, so I’m using IRF2 filters on those, especially when we’re playing smaller places with a close back wall, just to eliminate that first reflection. I try to keep everything on stage as isolated as possible because I’m already sending it out into this giant, ambient space.”

sE Electronics

Posted by Keith Clark on 11/05 at 03:57 PM
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Monday, November 04, 2013

Neutrik USA Announces 2013 premierePARTNER Award Recipients

Recognition of innovative audio solutions with Neutrik connectivity

Neutrik USA has announced the recipients of the company’s 2013 premierePARTNER Awards program.

The premierePARTNER Awards recognize 10 select companies that have developed an innovative product that incorporates Neutrik-brand connectors and meets the highest level of innovation for their segment of the industry.

Only the most innovative solutions are eligible for the award, which is presented to less than 1 percent of the company’s USA customer base.

Recipients of a Neutrik premierePARTNER Award have their products displayed on Neutrik USA’s Experience Neutrik Website (

Recipients a;sp receive a framed award for display on their company premises, and Neutrik USA displays a similar version of the award at the company’s Charlotte headquarters.

Additionally, a video of these awards runs at Neutrik’s booth during tradeshows to help promote the various innovative products incorporating Neutrik connectivity.

The recipients of the Neutrik 2013 premierePARTNER Awards include:

• A-Designs Audio, Pacifica Microphone Preamplifier
• Ashly Audio, nX Family of Amplifiers
• BAE Audio, 1073DMP Mic Pre Direct Box
• Benchmark Media, AHB2 Amplifier
• EAW, Anya Powered Array Module
• Leprecon, Watson Power Management
• Slate Media Technology, Raven multi-touch mixer
• Shure, GLX-D Wireless Systems
• TMB, ProPlex Opto-Splitter
• FiberPlex Technologies, WDM16 16 Channel Active Wave Division Multiplexer

“With its integration into the EAW Anya system, Neutrik is truly the connection to an entirely new way to deliver sound,” notes Jeff Rocha, EAW president. “It’s only logical that a product as adaptive as Anya allows customers access to audio and control via the Ethercon connector and the ability to safely connect and disconnect live power via the TRUE1 connector. Together, we free engineers to focus on managing the audience experience, not the system, opening entirely new paths for what’s possible in live sound delivery.”

Steven Slate, president of Slate Media Technology, adds, “It’s an honor to be chosen as a Premier Neutrik Partner. We use Neutrik connectors in our award winning hardware audio products and trust their quality and reliability over anything else in the industry.”

Peter Milbery, president of Neutrik USA,states, “We are fortunate to be involved with so many amazing manufacturers in the professional audio industry. This program is an excellent opportunity for industry professionals to learn about new and innovative products that incorporate Neutrik connectors.

“We constantly discuss new product designs and debate which ones should be included in the Top 10. It’s not an easy job. We believe that promoting cutting edge products that utilize our connectivity solutions benefits not just Neutrik and our partner companies, but most importantly, it benefits professional users by making them aware of these cutting-edge solutions.

“When you look closely at the award, you’ll see that it’s a badge much like a Neutrik employee badge. What this represents to us is that these companies are true partners and a vibrant part of our team,” he concludes. “Presenting these awards is one of the highlights of the year for all of us at Neutrik USA.”

Neutrik USA

Posted by Keith Clark on 11/04 at 03:52 PM
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An In-Depth Look At Microphone Cable Anatomy & Properties

Answers to key FAQs about mic cables and related issues

What is impedance?
Impedance is the AC (alternating current) version of the DC (direct current) term resistance, which is the opposition to electron current flow in a circuit and is expressed in ohms.

Impedance (often abbreviated as “Z”) includes capactive reactance and inductive reactance in addition to simple DC resistance.

Reactance depends upon the frequency of the signal flowing in the circuit.

Capactive reactance increases as frequency decreases: inductive reactance increases as frequency increases. Because of this frequency dependence, impedance is not directly measurable with a multimeter as DC resistance is.

What are the differences between high- and low-impedance microphones?
To answer this requires a little historical background. High-impedance microphones are capable of producing higher output voltages than low-impedance types. Until recently, “consumer” audio gear (small PA systems, home and semi-pro recording equipment, etc.) was always designed for high-Z mics because their relatively high output level required less amplification or gain.

The lower output of low-Z mics required the equipment manufacturer to use input transformers in front of the mic preamplifiers to step up the strength of the signal, which substantially increased the cost of the circuitry.

Hence, low-Z mics were rare outside of professional recording and broadcast studios. In these “big-budget” facilities, low impedance lines offered several big advantages. A high-Z mic’s high source impedance (approximately 10,000 ohms) combines with the capactive shunt reactance of the mic cable to form a low-pass filter which progressively cuts high frequencies. The severity of the loss is determined primarily by the length and construction of the cable.

The low source impedance (less than 200 ohms) of low-Z microphones proportionally reduces the high-frequency loss. Equally important, the high load impedances demanded by high-Z lines are much more susceptible to various forms of interference than low-Z lines, especially high-frequency noise and radio. Both of these high-Z liabilities made cable runs longer than 15-20 feet a problem.

Isn’t the use of balanced lines the biggest advantage of low-impedance microphones? What is a balanced line?

Balanced lines are wonderful, but they are sometimes given credit for benefits that they are not actually responsible for. Balanced, unbalanced, low-impedance and high-impedance are all individual properties.

Many people erroneously refer to anything with a 3-pin XLR-type connector as “low impedance” and assume it to be “balanced.” Others call any line connecting two pieces of equipment with 1/4-inch phone jacks “high-Z.” In reality, a lot of equipment has unbalanced inputs and outputs that are carried on XLR connectors, and there are even more low-Z lines on phone jacks.

Medical instrumentation uses a lot of high-impedance balanced lines for sensors, and most line-level unbalanced outputs are very low-impedance.

Electrical systems need a reference point for their voltages. Generally referred to as common or ground, although it may not be actually connected with the earth, this reference remains at “zero volts” while the “hot” signal voltage “swings” positive (above) and negative (below) it. This is referred to as an unbalanced configuration.

Physically, the common may be a wire, a trace on a printed-circuit board, a metal chassis - virtually anything that conducts electricity. Ideally it is a perfect conductor - that is, it must have no resistance or impedance. In a cable connecting two pieces of equipment, the shield is used as signal common.

As the complexity and size of the system is increased, the imperfect conductivity of the common (ground) conductor inevitably causes problems. Since it is made of a real material, it must have some resistance, which must (Ohm’s Law says) cause voltage drop when current flows through it, which means it cannot be at a perfect “zero volts” at both ends. The larger the system and the greater the distances between the source and load, the less effective this unbalanced configuration becomes.

The voltages of a balanced line are not referenced to the ground or common. Instead, the signal is carried on a pair of conductors with the signal applied to this pair differentially. The signals are electrical “mirror images” of each other - their levels are the same, but their polarities are opposite.

In other words, as the applied signal “swings,” one conductor will be negative with respect to the common, the other will be positive. These polarities alternate with the frequency of the signal, and the total signal level is the difference between the two individual voltages.

For example, if one conductor is at +5 volts, the other will be at -5 volts, and the signal level is +5 volts minus -5 volts or 10 volts. If, for same reason, the two conductors were both at +5 volts simultaneously, the level would be +5 volts minus +5 volts, which is zero volts. Very tricky!


Because of this differential signal transmission, two very valuable things happen when using balanced lines. First of all, each piece of equipment can have its circuitry referenced to its own common, because the interconnection of the equipment does not require that the commons are connected in order to move the signal around. This eliminates the major cause of a lot of noisy audio gremlins, ground loops.

Secondly, because the signal is differentially transmitted and received, any common-mode interference signal superimposed on the signal in the line will be carried by both sides at identical level and polarity. In other words, if the line has +5 volts of external noise induced, both conductors will have +5 volts of noise on them. This equals a total interference level of +5 volts minus +5 volts or zero volts. The interference cancels itself. This is called common-mode rejection.

There are several ways to balance lines.

Actually, the term “balanced” is very often used incorrectly to refer to lines that are actually floating. Properly speaking, a balanced line is one which has equal impedance from each side to ground.

An unbalanced signal may be derived from it by using one side of the pair as “hot” and ground as common. A floating line has no reference to ground, and must have on side of the line tied to common to “unfloat” it.)

The input transformers once required by low-Z mic preamps also provided a floating input as long as neither side of the transformer’s primary winding was tied to common. This is where the “low-impedance-is-balanced” misconception began.

The use of balanced lines was actually just a by-product of the requirement for a transformer to step up the low signal level. Using modern low-noise integrated-circuit design, a low-Z mic preamp can be clean, quiet, balanced and a lot cheaper to build - without a transformer.

What are the basic parts of a high-Z microphone cable and what does each one do?
A high impedance mic has many of the traits of an electric guitar, so the cable used for it is generally a coaxial instrument cable. The “hot” center conductor is insulated with a high-quality dielectric; shielded electrostatically to reduce handling noise and triboelectric effects; shielded with a braid, serve, or foil which is also used as the current return path for the signal; and jacketed for protection.


What are the basic parts of a low-impedance microphone cable and what does each one do?
The basic cable construction for low-Z mic or balanced line applications is the shielded twisted pair. It consists of two copper conductors which are insulated, twisted together (often with fillers), shielded with copper, and jacketed.


What gauge and stranding should the two conductors be?
The amount of copper in any electrical cable is usually dictated by the amount of current it has to carry, or by the tensile strength it requires to perform without breaking. If we take the worst-case situation, where the cable is used for a line-level (+24 dBm) 600-ohm circuit, the current is a negligible 13 milliamperes (that’s 13 thousandths of an ampere). The power in such a circuit is 100 milliwatts, or one-tenth of a watt. The current produced by a typical 150-ohm microphone connected to a 1,000-ohm preamp input is less than 10 microamperes (that’s 10 millionths of an ampere), with power of less than a microwatt.

By these figures it is apparent that not much copper is required to actually move signals around, except in applications demanding extremely long cable runs. Many low-impedance mic cables use 24 AWG conductors with excellent performance, and most multipair “snake” cables have 24 AWG (7 strands of 32 AWG) conductors.

Other things being equal, more individual strands in each conductor mean better longevity and flex life. Since singers using hand-held microphones can put a cable through several hours of tugging, twisting, straining and other abuse, these situations call for finer stranding and often larger conductors, sometimes as large as 18 or 20 AWG. However, the sonic properties of the cable may be compromised by using large conductors.


Why are the two conductors twisted together?
As previously explained, the interference-canceling common-mode rejection of the balanced line is based on the premise that the unwanted external noise is induced into both signal conductors equally.

Minimizing the distance between the two conductors by twisting them together helps to equalize their reception of external interference and improve the common-mode rejection ratio (CMRR) of the line.

The two conductors also form a sort of “loop antenna” for stray magnetic fields. The farther apart the two conductors are the larger the “antenna” becomes, and the more interference it picks up from sources like transformers, fluorescent lighting ballasts, SCR-chopped AC lines to stage lighting, etc.

Minimizing the loop area of the cable helps to reduce the unwanted hum and buzz from this type of interference, which the cable’s shield is almost totally ineffective against.

The distance between the twists is called the lay of the pair. Shortening the lay (increasing the number of twists) improves its common-mode rejection, and also improves its flexibility. The typical pair lay in microphone cables is about 3/4-inch to 1-1/2 inches. Shortening the pair lay uses more wire and more machine time to produce the same overall finished length, so of course it increases the cost of the cable.

What is “star-quad” cable?
This four-conductor-shielded configuration can best be thought of as two twisted pairs twisted together. Using four small conductors in place of two large ones allows the loop area of the cable to be further reduced and its rejection of electromagnetic interference (EMI) is improved by a factor of ten (20 dB). This makes star-quad cable very popular for microphones and balanced lines used in applications such as television production, where huge amounts of power cable for lighting and camera equipment surround the performers.


Does star-quad actually sound better?
When used for low-impedance microphones, star-quad construction substantially reduces the inductive reactance of the cable. Inductance was previously mentioned in discussing impedance. An inductor can be thought of as a resistor whose resistance increases as frequency increases.

Thus, series inductance has a low-pass filter characteristic, progressively attenuating high frequencies. While parallel capacitance, the enemy of high-frequency response in high-impedance instrument cable, is largely insignificant in low-impedance applications, series inductance (expressed in microHenries, or uH) is not. The inductance of a round conductor is largely independent of its diameter or gauge, and is not directly proportional to its length, either. Parallel inductors behave like parallel resistors: paralleling two inductors of equal value doesn’t double the inductance, it halves it.

In cable construction, using two 25 AWG conductors connected in parallel to replace each of the conductors of a 22 AWG twisted pair will result in the same DC resistance, but approximately half the series inductance. This will result in improved high-frequency performance: better clarity without the need for equalization to boost the high end.

Also of significance is skin effect, a phenomenon that causes current flow in a round conductor to be concentrated more to the surface of the conductor at higher frequencies, almost as if it were a hollow tube. This increases the apparent resistance of the conductor at high frequencies, and also brings significant phase shift.


What is phase shift?
Phase shift is a term describing the displacement of two signals in time.

When we described the two sides of a balanced line as being of opposite polarity, we could have said that they are 180 degrees out of phase with each other.

Each time an AC waveform completes a cycle from zero to positive peak to zero to negative peak and back to zero, it travels though 360 degrees (just like a circle).

A simple 1 kHz (1,000 cycles per second) sine wave travels through this 360-degree rotation in one millisecond.

If we consider its starting point to be zero, it will reach its positive peak one-quarter of a millisecond later, cross zero in another one-quarter of a millisecond, reach its negative peak a quarter-millisecond after that, and return to zero after a fourth quarter of a millisecond has elapsed.

Thus, each quarter of a millisecond equals 90 degrees of phase difference. When two identical signals are in phase with one another, their zero crossings and peaks are the same, and summing (combining) the two will double the amplitude of the signal. When they are 180 degrees out of phase, summing them will result in cancellation of both signals.

This property is very straightforward when considering simple sine waves. Sine waves consist only of a single fundamental frequency and have no harmonics. Harmonics are multiples of the fundamental, and are the elements of which complex waveforms are composed.

An excellent example of complex waveforms is called music. The reason a middle C note on a piano sounds different from the same note played on a flute is because the two instruments generate different waveforms - the harmonics of the piano are present in different amounts and have different attack and decay characteristics than the harmonics of the flute.

When complex waveforms are traveling in a cable, it would be ideal if the amplitude and phase relationships they enter the cable with are the same as those they exit the cable with. When the
effects of phase shift alter those relationships - when the upper harmonics that define the initial “pluck” of a string, for instance, are delayed with respect to the fundamental that forms the “body” of the note - a sort of subtle “smearing” begins to occur, and the sense of immediacy and realism of the music is diminished.

How can phase shift be minimized?
The phase lag caused by skin effect is one radian (about 57.3 degrees) per skin depth, and the effective skin depth of a conductor at a particular frequency is the same whether the conductor is very large or very small in diameter. For instance, the skin depth of a copper wire at 20 kHz is about .020 inches, while an 18 AWG conductor has a diameter of about .040 inches. This means that at frequencies from DC to 20 kHz, the full cross-sectional area of the conductor is utilized.

Because the skin depth (.020 inch) is never less than half the diameter of the conductor (.040 inch), there is never more than one radian of phase shift present. In short, star-quad cables seem to offer lower inductance and lower phase shift, both of which are parameters that directly affect the clarity and coherence of high-frequency complex waveforms.

Their inherently superior noise-rejection also reduces intermodulation distortion, a type which is particularly offensive because it produces “side-tones” not harmonically related to the fundamental. While the improvement may not be as dramatic as changing the microphone, an increasing number of audio professionals seem to be embracing the sonic benefits of star-quad construction.


What about the insulation? Does it affect the sound?
Even though the effects of cable capacitance are much less than that encountered in high-impedance applications, the use of low-loss, high-quality (low dielectric constant) insulation materials such as polyethylene and polypropylene are still preferred, especially when long cable runs are necessary.

Because of the desire to keep cable diameter to approximately 1/4 inch, the insulation thickness of a typical two-conductor microphone cable is generally about .020 inches, half that of a coaxial-type instrument cable. This relatively thin wall means that soldering requires good heat control to prevent melting.

For very thin (.010 inch) applications, cross-linked polyethylene insulation is sometimes used. The cross-linking process (similar to that used in manufacturing heat-shrinkable tubing) greatly reduces the problems of insulation meltdown and shrinkage during soldering.

Why does some cable have string-like fillers twisted with the conductors?
The primary use for fillers is to make the core of the cable round to eliminate convolution in the finished cable.

A twisted-pair is not round, and without fillers the finished cable will have an undulating, “wavy” appearance unless a very thick jacket is applied, which will greatly affect its flexibility and make it very difficult to strip.

A good example of convolution is found in the various thinly-jacketed twisted-pair cables used for pulling in conduit in permanent installations. Such cable is designed for economy and easy termination and so is not required to be round, only flexible and cheap.

Fillers also help to stabilize the cables shape and strengthen it, allowing some of the tugging, twisting and other stresses encountered to be absorbed by the filers rather than the conductors or shield.

Some special miniature cables used for the “tie-clip” lavalier microphones use conductors that are literally copper strands wound around cores of synthetic kevlar fiber. This cable is less than 1/8-inch in diameter, yet is enormously strong. (Unfortunately, it is also very difficult to terminate because of the necessity of sorting out the unsolderable kevlar from the solderable copper strands.)

Why don’t low-impedance cables require electrostatic shielding like high-impedance cables?
The “noise-reducing” semiconductive tape wrap or conductive PVC layers used on coaxial cable are used to “drain off” static electricity generated by the shield rubbing against the inner conductor insulation. When the source impedance is very high, these static charges will be heard as “crackling” noises as the cable is flexed and handled. A low source impedance has a damping effect on this type
of static generation which minimizes its effect.

There are cables available which use conductive textile or plastic shields for 100 percent coverage, with copper drain wires or very low-coverage copper braid added for ease of termination and low DC resistance. While this type of construction is very flexible, its shielding effectiveness suffers greatly as frequency increases, offering very little effect above 10 kHz because of its low conductivity.

What about handling noise?
The triboelectric effect that causes impact-related “slapping” noise as the cable hits the stage or is stepped upon during use is related to capacitance, specifically the change in capacitance that takes place as the insulation or dielectric is deformed. This causes it to behave as a crude piezoelectric transducer, a relative of an electret condenser microphone.

Because such transducers are extremely high-impedance sources, the drastic impedance mismatch presented by a low-impedance microphone and its preamp or input transformer makes the extraneous noise generated by triboelectric effects negligible except in cases involving very low-level signals.

In low-impedance applications, handling noise is best addressed by using soft, impact-absorbing insulation and jacket materials in a very solid construction with ample fillers to insure that the cable retains its shape. Note that it is totally invalid to evaluate the handling noise of a low-impedance mic cable without using a resistive termination to simulate the microphone element. A cable with no termination essentially presents an infinitely high source impedance, a situation that is beyond worst-case!

What special considerations should be given to shielding low-impedance cables?
Low-impedance microphone cables are shielded using the same basic methods as coaxial-type instrument cables.

Woven copper braid generally offers the best high-frequency shielding performance and protection from radio-frequency interference (RFI).

This is due to the very high electrical conductivity of the braid, and to its low-inductance, self-shorting configuration. Its disadvantages are primarily economic; it is the most expensive to manufacture and the hardest to terminate.

Spiral-wrapped copper serve shields are very inductive in nature, as they resemble a long coil of wire when extended. This can compromise high-frequency shielding and is not recommended when effective shielding above 100 kHz is required. Serve shields are relatively inexpensive and easy to terminate, making them a popular choice for medium-quality cables.

Foil-shielded cable is very heavily used for permanent installation work and for portable multipair “snake” cables. The extremely low cost, light weight and slim profile makes foil very advantageous in applications involving pulling cable into conduit.

In these cases the conduit (if metallic and properly grounded) can greatly enhance the RFI and EMI shielding properties of the thin mylar/aluminum foil generally used. The 100 percent coverage of the foil shield, which should be of great benefit at radio frequencies, is somewhat compromised by the inductive nature of the copper drain wire typically used for terminating it.

At low frequencies, performance is hampered by the relatively low conductivity of the foil/drain configuration. In applications involving repeated flexing and coiling, the metallized mylar tape will begin to lose its aluminum particles, opening up gaps in the shielding. This can be a particular problem with multipair cable used for touring systems, where the shield breakdown may lead to increased crosstalk between channels and to annoying radio pickup problems.

Does the use of 48-volt phantom power affect the performance of the shield?

The current typically drawn by a phantom-powered condensor microphone is generally limited by 6.81 kohm resistors, resulting in a current of less than 15 mA total.

This is not a significant factor unless the shield begins to break down mechanically due to use: tearing or fraying are possible, which could create intermittant changes in shield resistance. It has lead a few professionals to prefer the use of three-conductor microphone cables, with the common carried by a drain wire in addition to the shield.

• Ballou, Greg, ed., Handbook for Sound Engineers: The New Audio Cyclopedia, Howard W. Sams and Co., Indianapolis, 1987.
• Cable Shield Performance and Selection Guide, Belden Electronic Wire and Cable, 1983.
• Colloms, Martin, “Crystals: Linear and Large,” Hi-Fi News and Record Review, November 1984.
• Cooke, Nelson M. and Herbert F. R. Adams, Basic Mathematics for Electronics, McGraw-Hill, Inc., New York, 1970.
• Davis, Gary and Ralph Jones, Sound Reinforcement Handbook, Hal Leonard Publishing Corp., Milwaukee, 1970.
• Electronic Wire and Cable Catalog E-100, American Insulated Wire Corp., 1984.
• Fause, Ken, “Shielding, Grounding and Safety,” Recording Engineer/Producer, circa 1980.
• Ford, Hugh, “Audio Cables,” Studio Sound, Novemer 1980.
• Guide to Wire and Cable Construction, American Insulated Wire Corp., 1981.
• Grundy, Albert, “Grounding and Shielding Revisited,” dB, October 1980.
• Jung, Walt and Dick Marsh, “Pooge-2: A Mod Symphony for Your Hafler DH200 or Other Power Amplifiers,” The Audio Amateur, 4/1981.
• Maynard, Harry, “Speaker Cables,” Radio-Electronics, December 1978,
• Miller, Paul, “Audio Cable: The Neglected Component,” dB, December 1978.
• Morgen, Bruce, “Shield The Cable!,” Electronic Procucts, August 15, 1983.
• Morrison, Ralph, Grounding and Shielding Techniques in Instrumentation, John Wiley and Sons, New York, 1977.
• Ott, Henry W., Noise Reduciton in Electronic Systems, John Wiley and Sons, New York, 1976.
• Ruck, Bill, “Current Thoughts on Wire,” The Audio Amateur, 4/82.

This article contributed by Pro Co Sound.

Posted by Keith Clark on 11/04 at 03:42 PM
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Roland Releases New Windows 8.1 & Mac OS X Mavericks USB Drivers

Includes audio interfaces, MIDI interfaces, MIDI keyboard controllers, recorders, and more

Roland Corporation U.S. has announced that new Windows 8.1 and Mac OS X Mavericks (10.9) USB drivers are available for a wide range of Roland products including audio interfaces, MIDI interfaces, MIDI keyboard controllers, recorders, and more.

Roland’s engineering teams forge strong links with Microsoft and Apple to build and test new drivers alongside beta OS releases. Starting with the original MPU-40 in 1984, the company has nearly 30 years of experience creating interfaces for computer music applications.

Drivers for Roland’s extensive range of gear are released in concert with operating system updates.

Windows 8.1 (64-bit and 32-bit) and Mac OS X Mavericks (10.9) USB drivers are now available for the Roland products listed below.

Additional drivers for Roland products will also be added as they are completed.

Go here to download all available USB drivers

Audio Interface

MIDI Interface
UM-ONE mk2


MIDI Keyboard Controller


Other DTMP Product
Mobile Studio Canvas (SD-50)

Roland Corporation U.S.

Posted by Keith Clark on 11/04 at 02:44 PM
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Friday, November 01, 2013

Tech Talk: Building Directional Subwoofer Arrays

Working toward consistency throughout the listening area

Directional subwoofers are one more tool that can be used by sound system designers in their quest to achieve consistent sound throughout the intended listening area.

When using traditional, more or less omni-directional bass reflex (a.k.a., “vented,” “ported,” or “front-loaded”) subs arranged left and right of a stage, there is a build-up or “power alley” created in the center, where the energy from each source location shows up at the same time, with no phase difference, and sums quite nicely.

Moving left and right off of the center line, this area of addition is followed by alleys of cancellation.

Wavelengths of 40 to 100 Hz are roughly 11 to 23 feet long. At any frequency in this range, as you move away from the center line and change the path length difference between the two sources by half a wavelength (about 5.5 to 11.5 feet) there will be a cancellation, with higher frequency “nulls” encountered first.

To alleviate this there are three methods that have been employed: line arrays of subs, end-fired sub arrays, and cardioid subs, which are sometimes combined.

Line Arrays
Lines of subwoofers are one application of what Harry Olson discussed in the 1957 text Harry F. Olson, Acoustical Engineering, when he described a straight line source; using omni-directional elements, in a line, all reproducing the same signal, with relative close spacing compared to the wavelength, pattern control can be achieved.

Imagine a row of subs is assembled across the front of a stage. If it’s longer than the wavelength of the lowest frequency for which pattern control is desired (25 Hz is about 45 feet) and if the elements are close enough to one another, within two-thirds of a wavelength of the highest frequency produced (100 Hz is about 11 feet, so 2/3 is about 7 feet), cancellation at the ends of the line and addition in front of the array (and behind the array!) will be achieved.

Observed from the audience area, from one end of the line to the other, enough of the energy from each of the elements of the array arrives within +/- 120 degrees, at about the same level and sums.

Observed from the end of the array, enough energy from each of the elements arrives enough out of time but at similar enough level, causing destructive interference and level loss.

The use of a line array (yep, that’s what it is) of subwoofers can avoid horizontal differences in frequency response and deliver more energy to the audience area, while avoiding those nasty side wall reflections at lower frequencies.

Further, maximizing spacing can reduce the level differences from the front to back. In the interest of making sound where the audience is and not making noise where they are not, this is one option.

Remember, though, that the energy is the same in front and behind the array.

These arrays can also be assembled vertically, though space between the elements is not easily achieved with most rigging systems, so they are generally closely spaced arrays.

In amphitheater and arena situations where coverage to the sides is desirable, incrementally delay-tapering the horizontal array - so that moving away from center, each sub is slightly later than the one before it - can spread the coverage out towards the sides.

End-Fired Arrays
The end-fired array can be made up of two or more subs, spaced closely together, one facing the rear of the next, in a row along the “z-axis,” facing the audience and the direction of coverage.

Yes folks; it looks like it won’t sound “right.” Each cabinet needs its own drive line because we are going to incrementally delay all but the rear-most.

The rear, upstage, sub is delay time zero.

Moving towards the audience, each sub needs delay added corresponding to its distance from “sub zero.”

Let’s say the spacing is 3.5 feet: the delay time would be 3.1 ms (speed of sound = 0.9 ms/ft) for the next element.

The end-fired array produces gain in front of the array because the energy from each of the elements arrives in time at all frequencies being reproduced.

Cancellation behind the array is the summation of the energy produced by each source that is out of time and arrives at almost the same level.

There are a number of dips in frequency response based on the number of signals that have 180 degrees of phase difference. The level difference between front and rear is about 18 dB with a four-element array.

Cardioid Arrays
A few manufactures make multi-driver, single-cabinet cardioid enclosures, but they can be created with simple arrays of two or more cabinets.

The physical arrangement can be one of two options, both speakers facing the audience, one upstage of the other, lined up on the ‘z-axis,’ or one sub oriented facing backwards next to one or more facing forward. Again, people will question the appearance.

When both subs are facing the audience, one upstage of the other, delay and a polarity flip are applied to the signal going to the rear speaker.

In the rear, the energy from both loudspeakers arrives in time, at almost the same level, but with reversed polarity, resulting in broadband destructive interference and reduced level. In front of the array, the two signals arrive with polarity different and out of time.

This is a little tricky, but the first dip in the comb filter in this example is going to be at 160 Hz, out of band of the sub. If the spacing between the subs is 3.5 feet and the delay time is 3.1 ms, the two signals arrive 7 feet apart in front.

The wavelength of 160 Hz is 7 feet. With the polarity flip, the first dip of the comb filter will be at 160 Hz, not 80 Hz. The two signals in front are also at about the same level, so the dip will be significant.

The cardioid arrangement using forward and rearward facing subs can be assembled vertically or horizontally, subs stacked one on top of another or laid side by side, in a line, some facing the audience and one or more facing backwards.

Talk about looking like it won’t sound good. Behind the array, the output of the front and rear facing elements of the array need to match in time and be very close in level, but polarity backward to create cancellation behind the array.

A polarity flip and delay of the rear facing loudspeakers achieves this.

Determining the number of forward and rearward facing elements depends on model and how much energy needs to be created behind the array to cancel the energy from the forward facing subs.

The delay time will vary too, depending on model, and dimensions of the array, both vertically and horizontally.

Measurement is needed to determine level and time relationships between the front and rear subs.

An FFT transfer function can quantify this accurately. In front of the array, the summation of the rear facing loudspeakers is out of time and polarity different from the energy being produced by the forward facing subs.

The problem in frequency response, that first dip in the comb filter, must be kept out-of-band, higher in frequency than the operational range of the subs.

Alternative Methods
A hybrid approach, combining cardioid pairs, arranged in a line across the front of the stage, results in cancellation left, right, and to the rear. Alternatively, combining end-fired arrays and line-arrays also achieves additional directional control.

Using a directional array left and right affords the opportunity to join -6 dB down points in the middle of the audience and minimize the interaction between the arrays by minimizing the area where they are level similar, moving quickly into isolation of one or the other arrays. This would lend itself to very wide audience areas, such as amphitheaters and festival sites.

Directional arrays are often misconceived, mis-assembled, or are faulty in their operation. They require a knowledgeable operator, good equipment, and proper implementation.

The benefits can be substantial and are sometimes worth the risks. Avoiding some reflections in rooms, decreasing the amount of low-frequency energy on the stage (turn the floor monitors down, folks), and making the coverage smoother in amplitude and frequency response in the audience area are the substantial benefits when considering the use of directional low-frequency arrays.

There are several critical factors of performance that must be considered when assembling these types of arrays. Control of low frequency directivity is only possible when using exceptionally linear systems, precision-manufactured to perform identically.

The relationship between individual components must be consistent. What is sent electrically to the array elements needs to be turned into acoustic energy, without distortion or changes in frequency response as signal level changes.

New Tools
Historically, directional low-frequency loudspeakers have been in existence for some time.

Meyer Sound developed the first commercially available design, the PSW-6, a dozen years ago.

The PSW-6 uses a four-channel amplifier and signal processing built-into an enclosure that houses dual 18- and 15-inch drivers facing the audience, plus two more 15-inch drivers mounted in its rear.

This self-powered subwoofer provided cardioid vertical and horizontal polar response, serving as a new tool in the challenge of designing sound systems.

It eliminated 15 to 20 dB of the energy from the rear that would have bounced around and arrived in the audience area late.

Another advantage was the ability to place these loudspeakers in front of large walls without having to consider boundary reflections.

These and others continue to be advantages over omni-directional designs.

The PSW-6 design was a result of field experiments using the SIM (Source Independent Measurement) FFT measurement platform, along with prediction results from Meyer Sound’s then new MAPP Online (Multipurpose Acoustical Prediction Program).

MAPP, among its many uses, has become a tool that many practitioners use to design low-frequency directional arrays. Users are able to apply signal processing, arrange elements, and observe the results graphically as a narrow-band pressure plots or as broad-band Virtual SIM transfer functions, all predicted from the interaction of measured data sets of real loudspeakers.

“Measure twice and pile it up once.” Let’s face it, moving subs around in a parking lot is a lot of work and requires a substantial investment of time and effort, plus there’s tinkering with signal processing and measurement, as well as additional DSP and multiple drive lines.

On the other hand, moving subwoofers around on a computer screen is a two-finger event, and without the need for real subs, signal processing, and measurement platforms, a real time and money saver.

Not having to build and measure subwoofer arrays in the physical world as a first step has allowed users to design arrays that they might not have spent the time to experiment with in real life.

Steve Bush is a technical support representative for Meyer Sound.

Posted by Keith Clark on 11/01 at 05:54 PM
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Thursday, October 31, 2013

Roland Systems Group Expands Audio Recorder Line-Up With The New AR-3000SD

Offers programmable timers, network efficiency, support for multiple playback and control formats, plus support for audio formats up to 24-bit and 96 kHz.

Roland Systems Group has added the new Roland AR-3000SD to its line of announcement recorders/players, providing advanced sound quality and features with expanded compatibility that includes programmable timers, network efficiency, support for multiple playback and control formats, plus support for high quality audio formats up to 24-bit and 96 kHz.

Roland AR series of recorder/players are widely installed in train stations, airports, shopping malls, amusement parks, broadcast stations and museums, serving a variety of capacities such as broadcast announcements in airports, TV, radio, navigation guidance in shopping malls, sound design for amusement rides, and automated productions in event venues and showrooms.

Key features:

• 24-bit recording and playback at 96 kHz for higher sound quality
• Built-in yearly programmable timer
• LAN control with web server functionality
• External control of playback using the connection terminals
• AR Series Card Data Editor, ARE-3000
• Optional highly durable 4GB SD/SDHC Memory Card using SLC (Single Level Cell)

The AR-3000SD offers a new programmable timer and LAN control with web server functionality enabling settings and control from anywhere. It allows the user to select the optimal playback schemes for the application or system setup, providing control from the input connector or RS-232 port on the back.

There are five different playback types that include direct, program, binary, computer and MIDI as well as loop and repeat options.

The AR-3000SD can be an integral component of a larger installed system. Improved external controls of the AR can be achieved using switch-based audio guidance, sensor-activated guide narration, computer, phrase playback and timecode using MIDI or control over a network.

The new recorder/player can also control external devices such as playback of music on MIDI instruments, turning on amps or other devices, group control of lighting, linking audio to video playback or switching audio over a network.

A variety of audio formats are supported for recording and playback. AES/EBU allows for output of high-quality digital audio. For storage, SD or Compact Flash memory cards are supported with two virtual cards that provide recording of 2,000 phrases.

Playback and recorded files are uncompressed WAV and MP3 formats and up to 24-bit 96 kHz.

The AR-3000SD will be available in Q2, 2014


Roland Systems Group



Posted by Keith Clark on 10/31 at 03:12 PM
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Wednesday, October 30, 2013

A Whole Lot Going On: Computers And Interfaces At Front Of House

How various engineers approach the tasks to get the job done

Computers, of course, are omnipresent in today’s live audio production environment. They’re used for recording, backing tracks, system tuning and monitoring—and don’t forget walk-in/out music.

As I delved into the subject, I was struck by how various front of house engineers approach the tasks to get the job done and was surprised to find that computers were not always their first choice for every chore.

On The Record
There are often two processes that happen simultaneously when recording a live show: a “stereo’ reference and a multitrack recording. Recording the event for reference is sometimes done using a laptop, while recording multiple channels for possible live release is usually done via a transformer isolated split or direct from the console using a dedicated recorder.

Jon Burton (The Prodigy, Radiohead): “I’ve a portable ‘go with me everywhere set-up’ using a Zoom H4N 4-track compact recorder that allows me to record a stereo ambient track with the built-in mics along with a stereo desk-mix via the XLR inputs. I also have a (JoeCo) BlackBox 24-track recorder that usually sits at the stage splitter box to record individual channels, or on occasion it will be at front of house, where I send it sub groups and a few key channels on the direct outs.”

Jon Burton

Eddie Mapp (Evanescence, Stone Temple Pilots) notes that he employs a very similar setup, while Jon Garber (The Band Perry, Rascal Flatts) uses a slightly different approach: “When we record the show, we use a Pro Tools HD rig through an Avid HD MADI interface right out of the console. We also record a 2-track mix on CD every night.”

When asking about backing tracks, I discovered that multi-channel playback systems are rarely employed at FOH. Instead, many major touring acts employ a dedicated playback engineer that is back stage, cuing tracks, adding a ‘bigger sounding’ production to the stage.

This, I think, makes tremendous sense when you consider that fans go to shows expecting to hear production levels that approach what they hear when they listen to their favorite CDs at home. Most playback systems incorporate a couple of dedicated hard disc recorders along with a backing track switcher to ensure the show will proceed without a glitch, should the digital source fail.

Testing Un, Deux, Trois
A few weeks ago, I had the opportunity to hang with Brad Madix (Rush), and during the show, I noticed that he was monitoring the sound using a 360-degree mic array connected to his laptop. The laptop was being used as a multiband audio analyzer, displaying sound pressure levels at the various frequencies.

Eddie Mapp

As a result, I figured it was worth asking around to see what other folks are using and once again, the approach differs greatly. Dave Natale (Rolling Stones, Lionel Richie) says he likes to keep things simple: “I use an Alesis Masterlink to play back my trusted Sheffield Labs, James-Neton Howard and Friends disc.”

James Towler (Steve Windwood) simplifies the process even more with an iPod as his sound source, and Doug Short (Megadeth, Van Halen) told me, “I use (Rational Acoustics) Smaart with my Mac and sample the room from 3 to 5 locations.”

Garber typically samples from two locations, while Mapp goes all out: “For testing I use a Roland Octa-Capture (USB) as my interface with Smaart 7 and then do all of my system tuning within a Meyer Galileo 616 AES. This gives me eight measurement inputs with one reference, two wireless, console PFL, FOH mic, plus three more hard-wired mics if needed.”

Walk On By
Playing familiar music tracks through the PA is, of course, often done to fine tune the system.  Computers and other playback devices are also regularly used to provide background music during walk-in, between sets and of course for walk out.

I particularly enjoyed Burton’s approach: “I tend to listen to music quietly to check the system and like to use a bit of Dusty Springfield or Dionne Warwick as it puts me—and everyone around me—in a good mood. A bit of (Burt) Bacharach mid afternoon never did anyone any harm! I use ‘The Horses’ by Rickie Lee Jones to EQ the system and then check the low end with a bit of Deadmau5.”

When it comes to selecting a laptop, most of the guys I spoke with tend to go with Mac. Burton: “I use a MacBook Pro for playback and always have a pretty large collection of tracks of all genres with me. I use a program called Djay rather than iTunes for playback because it’s easy to set on Auto-Mix, and it can fade between tracks automatically.

“When I do dance gigs, it can even be set to randomly do echo fades, backspins, brake fades and even tempo/beat match! On some shows I can be required to pretend to be a DJ for up to an hour which is really dull…so this makes it fun.”

The interface between the laptop and the mixing desk varies. Mapp: “I travel with a Mac Mini. It’s rack-mounted and runs several programs for system tuning as well as iTunes for walk in/out music. At the beginning of each new tour I collect whatever music the artists wish to have playing and then import it into iTunes, set to shuffle forever, and let it roll. The audio leaves the Mac via the line out, which feeds a Radial JPC (active DI) into the console.”

Jon Garber

Opinions Vary
While I was working on this article, Mapp sent me a note asking if he could test out the new Radial USB-Pro, which as a stereo USB-to-XLR interface designed to make it easy to patch the audio out of a laptop and run it into the PA. As it happens, I’d also asked some of these guys what they thought about USB interfaces and as you can imagine, the response varied greatly.

Short is definitively not a member of the USB cable fan club: “I use a 1/8-inch TRS to dual XLR adaptor cable to run the audio and leave the USB connectors for printers and charging my phone.” Burton, however, has the opposite view: “I always use the USB port and I have a bus-powered interface. I also used to have a FireWire interface but the connector is too easy to knock out. The USB is a very solid connector.”

In developing the USB-Pro, we went back and forth between the USB Type A (long flat slot) and the USB Type B (D shaped) and after doing a heap of research, ended up going with USB Type B. It seems that most pros find that the Type B is more robust. But there’s no denying that when using consumer grade connectors like a mini TRS or a USB, greater care is needed… these are not XLRs.

So there’s a lot going on with computers—as well as interfaces and dedicated—inside the gated little cubicle we call FOH. Let’s go to Burton for the final word: “I take great joy in doing walk-in and walk-out music. On The Prodigy, it’s pretty much solid dance beats. but I do like to get a bit of old dub reggae in. For outro we sometimes use ‘Love Is In The Air’ by John Paul Young, and quite often see people dancing all the way out!”

Peter Janis is president of Radial Engineering, which has been producing snake systems, direct boxes and interfaces for more than 20 years.

Posted by Keith Clark on 10/30 at 02:54 PM
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Friday, October 25, 2013

Power Lines: AC Regulation And Conditioning For Systems

Power factors that can seriously compromise your system

Do I really need an AC (alternating current) power conditioner or regulator? Isn’t my surge protector enough?

Two of the most commonly asked questions regarding power, for sure.

Answers, in part, come from an understanding of modern-day realities. AC power is far noisier and contaminated than at any time since its inception.

This is due not only to an increased population taxing utility lines, but also because of the proliferation of computers and microprocessors.

These devices typically run on switching power supplies, which can have a devastating effect on noise levels as well as on the bandwidth and character of the noise.

The past 20 years or so have seen a 100-fold increase in AC noise (Figure 1), while at the same time, performance expectations from audio (and video) devices/systems has risen dramatically.

AC power is the foundation of every critical component used in a system, so as AC power quality deteriorates, so too does audio and video performance (and reliability).

A bit more specifically, line noise can smear, mask, and distort high-resolution signal, seriously compromising performance. These AC anomalies arrive from your breaker panel or service, and ultimately flow into every sensitive circuit.

Figure 1: Note how average AC line noise levels have steadily and dramatically increased in noise levels. (120 Hz to 1 GHz)

Better Yields Less
Experiencing the benefit of system components like superior audio processors, microphone pre-amps and even line array loudspeakers requires resolution.

To achieve this, low-level audio information must be rendered without distortion, unmasked, pristine.

Thus the paradox: as the AC noise level continues to rise, advanced audio technologies yield less and less of their potential, because low-level information is either masked or distorted. Additionally, trouble-free performance with unwavering microprocessor presets calls for highly stable AC power.

Years ago, we all drank water from the household tap. How many of us continue to do so?

Under current conditions, a rack of components that’s been without problems in 10 prior projects may hum, buzz, and worse come project number 11.

Why? Because the circumstances that produce ground loops can be quite complicated and are affected by the internal grounding scheme of the components.

In spite of efforts to create dedicated AC lines and “star-configured” AC cable grounds, in the end, few systems render an ideal “zero ohms” between ground connections. Lighting equipment, large appliances or machinery that share the same power grid can easily cause ground noise.

Quite often discrete power from generators is not possible, or may not be desired, given current limitations, ambient noise, and voltage stability concerns. Unfortunately, AC ground contamination, loops, and distortion may result.

Pursuit Of Specifics
Given the “Murphy’s Law” situation of our profession, many times these problems happen to occur just before the concert or prior to your fixed system’s first “big show.”

Thus we arrive at a useful tool in helping to keep Mr. Murphy at bay, the AC isolation transformer.

As the name indicates, these devices are intended to isolate AC power (particularly incoming AC ground), thus eliminating the source of many AC problems.

In the pursuit of specifics that can lead to further understanding, let’s have a look at the primary types of AC noise as well as solutions to help with the problem.

Common mode noise attaches itself to AC lines in even proportion (both line and neutral wiring referenced to ground).

It comes primarily from AC fields and all 60-cycle harmonics, as well as a sizable portion of RF (radio frequencies) inducing noise into the AC line. (Figure 2)

Though some products are designed with some form of common mode rejection from input to output (such as most audio power amplifiers), the majority of products are still susceptible to performance corruption from induction of this noise into their circuitry.

The most frequently seen common mode noise rejection built into audio components offer “choke and capacitor filter” designs, and most certainly, many of these do aid in reducing the problem, particularly in terms of RF noise.

Figure 2: Common mode noise attaches itself to AC lines in even proportion (both line and neutral wiring.)

However, the addition of a symmetrically balanced transformer goes many steps further in reducing and often completely eliminating common mode noise across the audio and video bandwidth.

Symmetrically balanced power is achieved by running incoming AC power into a 1:1 ratio isolation transformer, with a precisely placed center tap on the secondary.

This takes the incoming voltage (120 volts on the line terminal, and 0 volts on neutral and ground) and splits it in perfect halves on the output secondary of the transformer.

Thus the output line terminal now has 60 volts AC while the neutral terminal also has 60 volts when referenced to its center tap ground, which remains at 0 volts AC. (Figure 3)

Figure 3: To help eliminate common mode noise, balanced power is achieved by running incoming power into a 1:1 ratio isolation transformer.

What’s significant about this is that the two 60-volt terminals are now in opposite polarity. So, like opposing magnets, the fields cancel.

This canceling of common mode noise is extraordinarily efficient and linear across a huge bandwidth. (Recording and broadcast microphones have utilized this same noise reduction principal for over 60 years.)

And in the modern era, live sound professionals never consider using a microphone (or other components, for that matter) that is not balanced. Further, a host of ground loop problems can be eliminated in this manner, due to the transformer’s isolated signal ground.

Will Balanced Suffice?
The second type of noise found with AC power is called transverse mode, and it’s essentially the opposite of common mode in that the noise does not attach itself to the lines in even proportion. (Figure 4)

Also, transverse mode is even more devastating than common mode, and any common mode filter design (including symmetrically balanced) is generally useless in defeating it.

This is important to understand, because there are those in our industry who would have us believe that “balanced power” alone will suffice in the quest to reduce AC noise.

Transverse mode noise is typically produced by motors, appliances, switching power supplies and digital processing circuits. Significantly reducing this noise requires a low-pass filter of considerable range.

Here’s why. Recall the previously mentioned dramatic increase in AC line noise seen in the last 20 years. What’s just as critical is that the character and bandwidth of the noise has changed. As late as 1980, most electronic circuits used large transformers and linear power supplies.

But with the massive rise in personal computers in both business and residential applications, the character of AC noise changed forever.

Figure 4: Transverse (differential) mode noise does not attach itself to the lines in even proportion.

Two decades ago, AC noise was mostly limited to 60 Hz harmonics up to almost 400 Hz, followed by virtually unmeasurable noise levels up to approximately.100 kHz. Most of this noise appeared in the AM radio band (and beyond), in the 100 kHz to 1 GHz range.

This is significant, because it still serves as the basis of the design (or architecture) of nearly every AC conditioning filter produced today.

Most transverse mode filter designs are merely refinements of filters that were designed by Bell Laboratories nearly 100 years ago. These were conceived by tremendous engineers, but the designs were created to meet standards of the 1920’s.

This was a time when the AM radio was a pinnacle of technology, all electronic circuits were vacuum tube and switching power supply and microprocessors were barely, if at all, conceived.

Now we’re faced with a constant stream of new technologies and the challenges they present ­ for example, today it’s possible to measure more noise at 2 kHz, or 10 kHz, than octaves above in the RF band.

How We Hear It
Though it’s as important as ever to filter the RF bandwidth, it’s just as important to reduce noise in the audio band, particularly at 2 kHz to 20 kHz, where all the low-level harmonics occur.

For high quality audio, any noise in the upper octaves is devastating because of the way we hear music, along with the way that musical instruments work. (Figure 5)

For example, if we’re listening to music at our mixing console at an average level of 100 dB, most of the frequency content (at that amplitude) will be lower midrange and bass frequencies, and this will be primarily sustained energy.

However, the harmonics, upper partials of music, reverb-ambient information, percussive transient attacks and high-pitched instruments will be riding along at a FAR LOWER decibel level.

In fact, a great deal of information in these upper frequencies will be 20 dB, 40 dB, even 60 dB below the upper decibel level. This is how we hear music.

The reason a musician will pay a quarter of a million dollars for a Stradivarius violin, or the reason we can hear a significant difference between a Hamburg Steinway and a lesser grand piano is essentially the quantity and quality of harmonics that are typically very low in level compared with the fundamental tone.

We are capable of deciphering layers of information simultaneously, and often it’s these low-level sounds or signals that are most prized.

Figure 5: Typical audio frequency versus amplitude. Note how low in level the upper harmonics are when compared to the lower frequencies and sustained signals.

This is one of the reasons it’s important to have the headroom to raise volume levels well above the din of an audience. Comprehensive AC noise filtering can increase both dynamic range and audio quality.

Another valuable tool in the management of AC power is regulation, which essentially takes incoming voltages that are either too low or dangerously high and converts them into a constant, stable 120 volts. This is particularly important in systems where the incoming voltage is either continuously or intermittently above 125 volts or below 115 volts.

Low voltage can be particularly troublesome for power amplifiers because their rated power specifications are based on a constant 120 volts.

If incoming voltage ranges as low as 100 volts, a power amplifier will only produce a fraction of its rated power. urther, high voltage or poor power regulation can play havoc with video projectors.

If this equipment is fed from a generator, regulation can make the difference between a successful show and a rack full of broken equipment.

Figure 6: With AC regulation, widely variable voltage is converted into stable 120 volts.

Transient In Nature
AC power regulators are designed to insure delivery of a constant 120 volts output (typically within +/- 4 volts). Because the load demands of live audio equipment are transient in nature, AC voltage feeds may be taxed.

Further, portable AC generators are designed to produce a constant voltage only when connected to a constant load. Thus your amplifier rack could be receiving 116 volts during an opening set only to surge up to receipt of a dangerous 135 volts during an acoustic set because the amps are not in heavy use.

Particularly under these variable live conditions, it’s clear that AC regulation is essential. Surge protection and under/over voltage shutdown circuits are certainly as important as they’ve ever been.

But further steps are required if we wish to reap the full potential of today’s advanced technologies. Without comprehensive AC power management, both sound quality and reliability can be seriously compromised.

Garth Powell works in engineering and technical support for Furman Sound.

Posted by Keith Clark on 10/25 at 04:23 PM
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Monday, October 21, 2013

Radial Engineering Debuts Relay Xo Microphone Signal Switcher

Enables an artist to convert a vocal mic into a talk-back or communication mic in order to converse with system techs, monitor engineers or other band members

Radial Engineering has announced the Relay Xo, a microphone signal switcher that enables an artist to convert a vocal mic into a talk-back or communication mic in order to converse with system techs, monitor engineers or other band members.

Radial president Peter Janis explains: “Over the past few years, live touring has shifted dramatically with the universal use of in-ear monitors. Artists no longer use wedge monitors and side fills, and every instrument is piped directly into the ears. The benefit of course is greater control over the mix, consistency from venue to venue, no more feedback to contend with and sound bleeding from monitors into mics is virtually eliminated.

“Although the benefits cannot be understated, this has also introduced a new problem: band members have no way to communicate with each other. Radial produced a solution in the form of a simple momentary footswitch called the DM1 that is equipped with an XLR input and two outputs. When the footswitch is held down, the mic output toggles from the main PA system to the intercom system, enabling the artist to converse with band members or speak with the techs to adjust their in-ear mix. 

“The DM1 is designed for dynamic mics but falls short when used with condensers or wireless systems. The Relay Xo is basically a glorified version of the DM1 that is designed to sit at the wireless receiver bay and be remotely controlled using a standard momentary footswitch. The Relay 48Xo, now in the planning stages, will also enable condenser microphones to be employed without introducing a huge pop when phantom power is moved around. ”

The design begins with a simple XLR-F mic input and two XLR-M outputs. The user can togglesthe outputs using the side access A/B switch or connects a remote control such as the Radial JR1 or a momentary switch (like a sustain pedal) to the Relay Xo using either the XLR or 1/4-inch remote input jacks.

Although either a latching or momentary footswitch may be used, the latter is recommended as it prevents accidentally saying things to the audience if left on by mistake. With a momentary footswitch, the user needs to have a foot on the switch for it to work. A second 1/4-inch output provides a link to enable a second Relay Xo to be connected for stereo use. 

Because there are no buffers in the circuit path, the Relay Xo does not produce any form of distortion. There are only a few carefully selected blocking capacitors in line to prevent DC offset problems that could cause clicking in the audio system.

The Relay Xo follows Radial’s tradition of using 14-gauge steel throughout along with an internal I-Beam frame to ensure the sensitive PC board will not torque even when subjected to extreme rigors. A book-end design creates a protective zone around the connectors and switches to further prevent accidental damage. 

As with all Radial products, the Relay Xo is made in Canada and covered by a 3-year transferable warranty. It will ship in December, 2013, with an estimated retail price of $220 (USD).

Radial Engineering

Posted by Keith Clark on 10/21 at 09:47 AM
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Wednesday, October 16, 2013

Prism Sound Introduces Advanced Features For dScope Series III

“Check-box” automation sequencer gets sequence editor and easy form builder; soundcard support now extends to ASIO devices

Prism Sound has announced new features for the dScope Series III audio analyzer system used in R&D labs and production lines. The new release will be available for download from the company website in November 2013.

The new capabilities fall mainly into two categories:

a) Improvements to the easy-to-use “check-box” test automation system known as “Auto-Sequence”

b) Enhancements to sound card test capabilities

The Auto-Sequence tool provides a simple way to setup and execute a series of tests and to produce high-quality reports in html and Microsoft Excel formats without having to write any code. This update provides an “Edit Mode” that makes sequence construction and adjustment even faster so that automated production line tests can quickly and easily be set up using the design engineer’s test configurations as a starting point.

For example, it is now possible to browse to an existing test configuration file and import it directly as a new step into the test sequence and to rename, copy, delete and re-order test steps.

Also included is a “tweak” feature that enables a single parameter adjustment to be applied to several test steps automatically, avoiding the need for editing individual test configurations manually.

Finally for the Auto-Sequence update, a new Dialog (or Form) editor allows easy construction of pop-up forms for user interaction such as entering serial numbers and other information, confirming actions such as external test fixture or device adjustments or confirming observations of the equipment under test

In the dScope Series III Audio Analyzer System, Prism Sound pioneered the testing of sound cards using a direct software interface between the test application (dScope) and the sound card’s Windows (WDM) driver, combined with the complimentary function of the dScope hardware connected to the other side of the sound card. 

Using this method it is possible to drive the soundcard from the dScope software audio generator and to measure the analog soundcard output with the dScope’s analog measurement input. Similarly, it is possible to drive the soundcard with the dScope’s precision analogue generator and then measure the audio performance by measuring the audio stream returned from the soundcard using the dScope’s software analyzer functions.

The new release of software now adds the capability to support ASIO drivers as well as WDM, enabling a wide range of professional as well as consumer audio soundcards and interfaces to be tested in this way. In addition, ASIO is a useful tool that guarantees bit-perfect transfer between the host computer and the audio device, especially useful when testing the accuracy of the interface device.

In addition, dScope’s Continuous Time Analyzer (CTA) is now available for use with Soundcard inputs in addition to the FFT based analysis previously provided for soundcards. dScope’s unique FFT detector meters provide all the usual audio measurements such as amplitude, noise, THD+n and many others, but the CTA is a time-domain analysis process that mirrors the conventional notch filter plus voltmeter method commonly used in analogue instruments.

The CTA has several advantages; it is faster than the FFT analyzer, especially when performing sweeps; it can make peak and quasi peak measurements and perhaps most importantly it can be used to detect occasional glitches such as sample errors. The CTA can measure inter-channel phase, important with computer interfaces where it is essential that all channels experience the same latency. The CTA can also generate an output trace so we can measure and plot the residual noise & distortion after the measurement filters to aid understanding of the composition of the noise and distortion components

Finally, the dScope scripting environment now provides support for serial port control on 64-bit Windows systems, enabling the control of external devices using RS232/422.  This could be used to control external switching, DC power supplies or other equipment.

Graham Boswell, sales director at Prism Sound states “We are continuing in our mission to provide the most powerful and flexible audio analysis tools at the best price available in the market. These additions to the capability of dScope Series III ensure that the dScope remains the best available audio analyzer tool for your money. No other analyzer product provides as much capability for the price – indeed few, if any, other analyzers provide more capability – period!”

Attendees at the 135th AES convention in New York this week can get more information about dScope Series III at the Prism Sound booth, number 3032.

Prism Sound

Posted by Keith Clark on 10/16 at 05:42 PM
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Consoles & Mixers For Church Sound, In Context

Options are plentiful and growing in this dynamic marketplace

Over the past decade, we’ve seen the proliferation of digital consoles in both live and recorded sound. This is not to say that analog consoles are going away at all, particularly with respect to house of worship systems.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Digital console platforms offer several advantages for church sound applications.

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

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

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

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

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

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

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

Posted by Keith Clark on 10/16 at 05:11 PM
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