Tuesday, February 10, 2015

Wired In: Developing A System For A Unique Chicagoland Venue

Meeting the needs for live music performance and more in a space carved out of an old Teamsters hall

The poet Henry Wadsworth Longfellow famously said, “Music is the universal language of mankind.” It’s certainly a dialect shared by four lifelong Chicago-area musicians who have come together to create Wire, a venue for live music performance, education and production carved out of an old Teamsters hall in near-west suburban Berwyn, IL.

The four musicians include Chris Neville of Tributosaurus, which faithfully reproduces the recordings of classic rock groups (a different one each month); Paul Bolger of popular jam band Mr. Blotto; Tracy Dear of alt country group Waco Brothers, and Jon Smith, a noted recording engineer.

A collective statement from the partners presents their vision: “Wire was an idea that came about several years ago, an idea that intends to reclaim music’s heritage as a method of communicating – not something done in isolation, but something shared with other musicians, the surrounding community, and the world at large.”

Wire’s stage hosts a steady stream of live performances both diverse and eclectic: rock, blues, reggae, acoustic, jazz – and those that escape easy definitions. Space behind the stage has been carved out as the classrooms for Rock University, where students learn both music and production, and above that is the makings of a soon-to-be implemented recording studio.

A perspective of Wire, with Bose Pro RoomMatch arrays flying left and right.

The venue is an important part of a thriving transformation of the arts community along Roosevelt Road in Berwyn, just a 20-minute drive from downtown Chicago and already the home of venerable FitzGerald’s Nightclub, a staple of the music circuit celebrating its 35th year.

It’s not surprising given the collective decades of performance experience that the partners are well-informed on the subject of sound reinforcement. Bolger, for example, notes that Mr. Blotto has owned its own PA for two decades, made up of premium components. Equally unsurprising is that they turned to fellow musicians for design and installation of Wire’s house and monitor systems.

Neville’s initial call went to TC Furlong, head of the sound company based in Lake Forest, IL since 1973 that bears his name and also a long-time player on the region’s music circuit. Furlong in turn brought Brian O’Connell of his staff into the loop to serve as project manager, and he too is a veteran player on the Chicago scene, noting, “I’ve known Paul (Bolger) for a long time. Our bands have opened for each other over the years.”

Brian O’Connell at Yamaha CL console in the TC Furlong shop.

Developing The Space
The building that houses Wire actually began life as the Oak (and later the Oakwyn) Theater in 1934, largely presenting motion pictures. Later it was acquired by the local chapter of the International Brotherhood of Teamsters, which transformed it from a theater into a two-story office building. It remained that way, even after the Teamsters vacated it, until being purchased by the Wire partners.

The interior was essentially “gutted,” revealing a beautiful, open space framed by exposed beige brick. After decades of inattention, the brick was restored to its original beauty, topped by a natural wood ceiling that adds to a warm, clean aesthetic.

A large stage (27 feet wide and 15 feet deep) dominates the front of the room, and looking out from there, one sees a large audience area capable of accommodating about 400, with a bar behind that topped by a VIP area comprised of a couple of the old offices that were left in place.

Right away, it was obvious that the room’s numerous hard, parallel surfaces (the floor is concrete) presented a primary demand of the house loudspeakers: pattern control. At the same time, the partners wanted a system with a highly musical signature.

A closer look at one of the Bose Pro RoomMatch arrays.

During the evaluation process, O’Connell tipped off Bolger and Neville to a demonstration being presented by Bose Professional of its RoomMatch loudspeaker arrays at another local theatre. They liked what they heard, and a direction was set.

“RoomMatch has a beautiful musical signature,” Bolger says. “You can hear all of the subtleties, the true tonality of acoustic instruments, the ‘whisper’ of guitar strings. It’s beautiful, clean, and intelligible.”

O’Connell notes that a key aspect of the RoomMatch approach, with models available in numerous dispersion patterns, also met the absolute need for output that could be focused on the audience while keeping stray energy off of the room’s many reflective surfaces. Further, with the TC Furlong team new to RoomMatch, Bose Professional stepped up to provide design support, particularly in terms of modeling via the company’s Modeler software.

Main Configuration
What resulted are flown left and right main full-range arrays flanking the Wire stage, each comprised (bottom to top) of one RM12020 module, two RM9020 modules, and a RM7010 module.

The model numbers reveal the coverage patterns of the modules, with horizontal coverage wider at the bottom (120 by 20 degrees) and then tapering to more narrow (90 by 20 and then 70 by 20 degrees) as you move up the array. The result, according to Bolger, is uniform coverage of the room both front to back and side to side.

“Dispersion is very even, and there’s distinct clarity in fully revealing the subtle detail of the music,” he states. “You can walk anywhere in the room and the hi-hat doesn’t go away, the vocals don’t go away, the real character of the music doesn’t go away.”

The open nature of the space and a truss grid above the stage translated to easy selection of optimum locations to fly the arrays. Meanwhile, there wasn’t enough height to fly the subwoofers with the arrays, so to keep the floor clear of obstructions, the system’s two RMS218 (dual-18-inch) subs are positioned together beneath the center/front of the stage, and they’re in a cardioid configuration topped by a judicious use of digital processing to help keep their output under control.

All main system loudspeakers and the subs are driven by four Bose Professional PowerMatch PM8500N networked amplifiers and managed by a ControlSpace ESP-00 digital processor using four ESP I/O cards. These components are located backstage.

The dual Bose Pro RMS218 subwoofers are under the stage in a cardioid configuration.

Mix & I/O
The other primary component in the system is a Yamaha Commercial Audio CL5 digital console that’s capable of handling both house and monitors. Fronting the custom sound booth constructed in one of the rear corners of the room, it provides plenty of capabilities for guest engineers.

The console is linked via (Audinate) Dante networking to two Rio stage boxes, one to each side of the stage, that accommodate up to 64 inputs, with another eight inputs available on the console.

“The CL5 is a great choice for this application,” O’Connell notes. “The technology is proven, engineers like to mix on it, there’s plenty of capability, plus we’ll be able to link it via Dante to the recording studio system when it’s ready.”

“This is an excellent console both sonically and operation-wise, with plenty of onboard facilities in terms of effects,” Bolger adds. “Plus it’s great for teaching our students. We like to get them behind the board and show them the cause and effect of what they’re doing on stage as it relates to what’s happening with sound in the house.”

Bolger in the booth at the venue’s Yamaha CL5 console that handles the house, monitors,tand also serves as a teaching tool.

Currently a Dante run from the console feeds a multitrack recorder for capturing live performances, and Cat-5 is also in place to facilitate a Dante link to the main system rack in the near future. (That feed is analog for the time being.)

There really weren’t many options with respect to the booth location – placing it centrally would have impeded crowd traffic flow while also occupying too much prime listening real estate. However, this difficulty is largely alleviated with the use of the console’s StageMix app with an iPad that’s kept at front of house, allowing engineers to dial in the mix from anywhere in the room.

At The Stage
This applies to monitors as well, with the CL5 accommodating up to 16 mixes on stage. Several QSC KW122 (12-inch) active 2-way loudspeakers are available for artists, placed horizontally on their cabinet’s monitor position, with a 15-inch KW152 provided for drum fill. They’re also outfitted with EQ modes for switching between optimized settings with the press of a button on the back panel.

QSC KW122 active loudspeakers serving as monitors on the Wire stage.

“I really like these boxes as monitors. Both the 12s and 15s get it done,” Bolger says. “I also like that you can adjust them if you want. It’s incredible what compact loudspeakers can do these days.”

“Actually, my band played at Wire a few months ago, and as a musician, I found the stage monitoring situation to be great,” O’Connell adds.

The microphone package offers a variety of models from Shure (SM57 and SM58, several BETA mics, and three KSM137 condensers). Direct needs are met with Radial Engineering ProDi boxes as well as a ProD8 providing eight channels in a single rack-mount box.

All of the equipment hadn’t arrived as the venue’s grand opening date approached, so TC Furlong supplied loaner gear from its extensive rental stock to handle the situation until everything was delivered and installed. The company also provided training on the console and a technician for opening night.

Desired Result
Wire is convenient to expressways and public transportation, seeing notable success in attracting patrons from all over the Chicagoland area for live music several nights a week. It’s already garnered a reputation among musicians as a place to play and be heard, and among patrons who appreciate the care that’s gone into creating the venue.

“It sounds more like a concert hall than a club,” says O’Connell. “This system sounds fantastic, to us and to the club’s owners and to everyone who’s been there.”

“We set out to create a place where we’d want to play as artists, and that’s led to a great result and what we see as a valuable addition to the area music scene,” Bolger concludes. “We’ve worked carefully in how we’ve allocated our resources, and those choices have paid off for everyone involved.”

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

Posted by Keith Clark on 02/10 at 02:42 PM
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Sunday, February 08, 2015

In The Studio: Introducing The New MIDI HD

This article is provided by Bobby Owsinski.

The MIDI standard is celebrating its 30th anniversary this year, so let’s take a minute to appreciate just how cool an invention this really was.

For many of you reading this, you don’t remember a time before MIDI, but the rest of us remember it as a time when hardware synthesizers, sequencers and drum machines couldn’t talk to one another if not from the same manufacturer. You chose a manufacturer if you wanted all three, and usually at least some of these were not to your liking.

MIDI, originally proposed by Sequential Circuits founder Dave Smith (now with Dave Smith Instruments and pictured above/left), solved this by suddenly making not only these devices interchangeable, but controllable via external switches, pedals and, a little down the road, computers.

The standard hasn’t changed much over the years, but definitely needs some updating, and that’s exactly what the newly proposed MIDI HD does.

Some of the new protocol’s features include:

· Plug and play network connectivity over USB and Ethernet

· Thousands of channels for handling complex systems

· More precise pitch control and articulation for expanded expressivity

· Tighter timing thanks to time stamped messaging

· More controllers and parameters

· Room for future expansion

· Backwards compatibility with MIDI 1.0

That sounds pretty cool, but don’t get the wrong idea—MIDI HD is not a replacement for the standard MIDI that we’re all used to. MIDI 1.0 is really cheap to implement and MIDI HD isn’t, at least at the moment, and it’s not an industry standard yet either. The added cost of MIDI HD means that many low cost devices just won’t have it for a while.

That said, it’s exciting that MIDI is stepping into the future. I can’t wait to check it out in person to see just what it can do.

For more information, go to the MIDI Manufactures Association’s office site at midi.org.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website, and go here to acquire copies of his top-selling books.

Posted by Keith Clark on 02/08 at 12:33 PM
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Friday, February 06, 2015

Factors In Capturing & Presenting Snare Drum

I created a frog. It wasn’t intentional. Naturally, I’m not talking about a real frog, but look at the photo that opens this article. You’ll never read a mixing book that says, “Make the snare’s EQ curve look like a frog in water.” (If you do, stop immediately and back away.)

Seriously, when it comes to snare mixing, the last place (literally) you want to be is behind the mixer. With that in mind, here are three primary factors I’ve identified in getting a good snare drum sound.

1: The Drum
Snare drums don’t all sound the same, just like all acoustic guitars don’t sound the same. Even with a house drum kit, a drummer might bring his own snare because of it’s sound. Know that each snare has a unique sound. This is the baseline sound for the mix. Use the same mic and the same EQ settings with two different snares, and you’ll get two different results.

Consider the three different snares shown in Figure 1. Just by looking at them, you can almost hear the tonal differences. Material composite, drum size, drum head skin, all of these are factors. Even tuning makes a difference. Snares can be tuned to match whatever the drum tuner decides is to his liking. To generalize, there can be a low or high tuning (true for any drum).

Figure 1: Left to right, PDP Blackout Maple, PDP LTD Classic Wood Hoop, and Pearl Chad Smith Signature snare drums.

Stand near the drum kit while the drummer plays the snare. This is the sound you’ll be mixing with – not against. Don’t try making it sound like something it’s not.

2: The Microphone(s)
A mic should be paired with an instrument and so it is with miking the snare. The Shure SM57 pairs great with a snare drum because of it’s polar pattern and frequency response.

I polled some techs and their pairings include the Telefunken M80, Heil PR22, Heil PR28, DPA 4099, and the Granelli Audio Labs G5790, a modified Shure SM57 designed for tight spots. And don’t think mic designs are the same (Figure 2).

Figure 2: Clockwise from top left, Granelli G5790, Telefunken M80, Heil PR31 BW, and DPA 4099.

Photos are nice but let’s get real – we need to look at specifics. They have different polar patterns, different sensitivity, and they don’t have to all be dynamic mics. For example, the Heil PR31 BW is a dynamic while the DPA 4099 is a condenser.

While several characteristics can make a big difference in how a mic treats sound, frequency response is a factor never to be overlooked. It alters the tonal characteristics of the snare drum. Take just one snare drum from above, like the Pearl Chad Smith Signature, and apply three different mics – the result is three different sounds. And we haven’t even touched the EQ.

For comparison, Figure 3 offers the frequency response charts for the SM57, PR31 BW, and 4099. (Note the charts with multiple lines are showing the differing frequency responses when not on-axis with the sound source).

Figure 3: Top to bottom, published frequency response charts for the Shure SM57, Heil PR31 BW, and DPA 4099.

In addition, snares can be (and are often) miked both over and under the drum.  Here are a few combinations my tech friends recently sent me: Shure Beta57 over, Shure SM81 under; Audix i5s both over and under; Sennheiser MD421 over, Heil PR31 BW under; Audix i5 over, SM57 under; Heil PR22 over, Sennheiser e904 under.

3: The EQ
Equalization should only happen after we listen to the natural tone of the snare and consider the mic(s) we’re pairing with it.

Here’s an example: take a snare tuned high and pair it with a mic that has a large high-end boost. Want to cut the highs in the mix? It’s not going to be easy as you’re mixing against what is being sent, not mixing with it.

At this point, you can be as simple or as creative as you want. How do you want to mix a dual-miked snare? How do you want the snare to sound for the song?

It’s not a matter of “how do I use the equipment?” but rather a matter of “what would sound right and how do I get there?” By having the right snare and mic combination, you’ve got the hard part out of the way. (I know this isn’t always within your control.)

I like a single-miked snare. That’s not to say I won’t fall in love with a dual mic setup next week. But a single mic setup is a good place to start. By establishing a good single-mic sound, when moving into two mics, you already know how to get a good sound from one. Make sense?

Sum Of The Parts
The ideas that follow are based on my experiences and can serve as a starting point in capturing and mixing a snare. All of the sounds of the drum kit (and the whole band for that matter) have to be considered. The right sound for the snare for a particular song might be really flat on its own.

High-pass filtering. A mic like the SM57 has the low-end rolled off, and I’ll roll off a bit more if I notice a positive impact on the sound. I’m not going to roll off more just to then flatten the snare sound. If you’re running an analog board, hit the HPF switch and listen for a difference.

It’s a good idea to remove low-end frequencies from all mics that aren’t focused on a low-end instrument. For instance,  use an HPF on vocal mics. Snare and cymbal mics, which aren’t focused on low-end kit pieces, are another good place for applying an HPF.

Gating. I’ve never been quite happy with the results of gating snares, at least as it applies to general snare sound. I’ve gated the snare for a song to get a specific sound, but for all-around mixing, I tend to skip it. (Your mileage may vary.

Out with the bad. I sweep the mid-range with a 6 dB cut and find the area of offending frequencies. You know, that area where you make the cut and suddenly think, “now that sounds much better.” In the case of my “frog EQ,” I didn’t find that spot but rather found a huge boost was needed. Some days it’s like that.

Sculpt To Fit
It’s of real benefit to have a sound in your head that you want the snare to match. It’s that internal reference sound. You know what sounds good, now make it a reality. Is there too much snap? Not enough? Perfect the way it is? (This is where that snare/mic pairing pays off.)

There’s no magic formula for exactly what to boost or cut and where to do it. It all depends on the snare and the mic and the drummer and the room and…eh, you get the point. That said, here are a couple of places to start:

—Snap and presence, 3 kHz to 12 kHz. The higher you go, the less presence and more snap.
—Body, sub-500 Hz. If more substance is needed.

The take away is to do the homework: know the tone of the instrument, pair it with the right mic, and then step behind the mixer. A good snare sound is the very likely result.

Chris Huff is a long-time practitioner of church sound and writes at Behind The Mixer (www.behindthemixer.com), covering topics ranging from audio fundamentals to dealing with musicians – and everything in between.

Posted by Keith Clark on 02/06 at 10:44 AM
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Thursday, February 05, 2015

The Ins & Outs: Detailing The Latest Digital Console I/O

PSW Top 20 presented by Renkus-Heinz

When shopping to add a new console to my company’s inventory, I look at three primary aspects: reliability, sonic quality, and routing options.

Most consoles from the major manufacturers are reliable and sound great, so the choice often comes down to the inputs, outputs, networking and overall routing capabilities.

There needs to be the required number of inputs necessary to handle most/all of the gigs my company works, and further, we want extensive routing options and outputs to both send feeds everywhere and patch things easily.

The corporate events that are our specialty require a multitude of output sends, including feeds for recording, video world, podium and backstage monitors, dressing rooms and show intercoms, overflow and/or breakout rooms, and of course, feeds to the main PA, delay zones, front fills and possibly tie-ins to the house and lobby loudspeakers. 

On smaller budgeted shows, the front of house console might also have to supply monitor and stage fill mixes for the entertainers onstage. Live remotes add a few more sends that could include splitting off certain inputs or crafting sub mixes to send to a broadcast truck. It’s surprising that even small gigs can have a lot of routing requirements, including stage and in-ear monitors, aux fed subs, a feed to a videographer, and again, the main system.

And it’s just not outputs that need to go everywhere. Inputs can be located on stage, off stage, backstage, in another room, in a remote truck, or right at the console. The ability to patch inputs has taken on increasing importance, even when a gig doesn’t appear to be all that complicated at first glance. Another concern can be a console’s networking ability, and how it can connect to various digital network protocols as well as analog systems.

My days of running large, heavy analog multi-core splitter snakes are coming to an end, replaced by networked audio over coax, fiber or Cat cables. Even many smaller consoles feature the increasing capability to interface with a variety of network protocols like Dante, MADI and/or the manufacturers’ own protocol. Networking lets us to locate stage boxes anywhere without having to run obtrusive copper snakes, and it also allows the console to interface with additional consoles and recording devices.

Remote live access is another item that has been added to many smaller consoles/mixers. Being able to walk around a venue and mix the show on an iPad or stand onstage next to a performer during sound check and dial in their wedges has made my job easier. Another benefit of wirelessly accessing the console lets us place it at the stage and not even run a snake at all.

An increasing number of models also allow wireless access via tablet or phone so that performers can adjust their own stage mixes. This can save a ton of money for a band or church that can’t afford a stand-alone personal monitoring system, and it can also save engineers and techs a lot of time and hassle. With so many new models and technologies hitting the market recently, let’s take a look connectivity capabilities.

Compact rack-mount consoles/mixers are becoming quite popular and Allen & Heath unveiled a new one at last month’s NAMM show. Qu-Pac can be used on a tabletop or mounted in a rack, with full control of the mixer from the front-mounted touch screen as well as with wireless iPad and iPhone remotes.

The 16 onboard XLR mic and TRS line inputs can be expanded up to 38 inputs via the Allen & Heath dSNAKE network, and the unit also offers a 32 × 32 USB audio interface, 2 stereo TRS input channels, 12 mix buses and AES output. A cool feature is the ducking circuit, which can come in handy for making announcements over music or allowing one person’s mic to override others at a meeting.

Soundcraft Ui 16

Also new at NAMM was the Soundcraft Ui Series of compact mixers that can be placed anywhere, like a stage box. (The larger unit can also be rack-mounted.)

Both models have built-in Wi-Fi and the ability to be controlled by any connected device via a standard web browser, including iOS, Android, Windows, Mac OS, and Linux devices.

The smaller Ui 12 has 4 XLR inputs, 4 XLR/TRS combo inputs, dual 1/4-inch line inputs, stereo RCA line inputs, 2 XLR aux outputs and a pair of main XLR and 1/4-inch outputs. The larger Ui 16 adds 8 XLR/TRS combo jacks to 4 XLR inputs and 4 aux XLR outputs.

Staying in the rack-mount genre, PreSonus recently added a couple of new compact models to its StudioLive mixer lineup. The RM16AI and the RM32AI offer the same features, with the only differences being the number of channel inputs and mix outputs. Both are operated using UC Surface control software for Mac, Windows, and iPad. The RM16AI has 16 XLR inputs, 8 XLR aux buses, and 3 XLR main outputs for LCR.

PreSonus RM32AI

Meanwhile the RM32AI offers 32 XLR inputs, 16 XLR buses and 3 XLR main outputs. Both also provide stereo RCA inputs and an option card slot.

A S/PDIF digital output option card, as well as FireWire S800 and Ethernet cards, are available now, with Thunderbolt, Dante and AVB cards announced. Using the same one-click recording software as the rest of the StudioLive line, these new units foster easy virtual sound checks.

Let’s shift gears to larger models. Yamaha Commercial Audio just introduced a new flagship console, the RIVAGE PM10. The control surface offers 8 analog inputs and outputs, 4 AES inputs and outputs, and a pair of MY option card slots.

Yamaha RIVAGE PM10 DSP-R10

The DSP-R10 engine provides an additional 2 MY slots along with 4 HY slots (TWINLANe and Dante). The main connectivity is via RPio622 stage boxes, which can provide up to 96 mic preamps per rack along with 2 HY slots.

I’m also anxious to hear the newly-developed RY16-ML-SILK hybrid mic preamp with digital modeling based on Rupert Neve Designs transformer circuitry, with users able to choose between a transparent audio path or one enhanced by the Silk processing, which provides a bit of color and character to each input.

The new mc²36 is an “all-in-one” console from Lawo, offering 32 mic/line inputs, 32 line outputs, 8 digital AES3 inputs, 8 digital AES3 outputs, MADI and 3 RAVENNA network ports that can provide additional connectivity for up to 384 external inputs and outputs.

Up to three mc² compact I/O stage boxes can be connected, each with 32 mic/line inputs, 32 line outputs, 8 AES inputs, 8 AES outs, and MADI.

Also of note, the console has 21.5-inch HD touch screens and touch-sensitive color-illuminated rotary encoders, helping the operator keep track of mix functions.

The new Roland Professional A/V M-5000 console offers the company’s proprietary Open High Resolution Configurable Architecture (OHRCA), which basically means that the internal mix architecture is not fixed and can be freely defined within a range of up to 128 input/output channels/buses.

Onboard connections include 16 XLR inputs, 16 XLR outputs, 2 AES inputs, 2 AES outputs and REAC networking ports. Two option card slots allow interfacing with Dante, MADI, additional REAC streams and Waves SoundGrid.

Roland Professional A/V M-5000

Additional connectivity can be had via Roland digital snake boxes, including the latest, the S-2416, offering 24 XLR inputs and 16 XLR outputs.

A highly readable, well-defined 12-inch color control screen, color changing encoders, and wireless iPad control are very operator friendly.

DiGiCo has a new version of its popular compact mixer, the SD11i, which can be used as a desktop unit or rack-mounted. All 32 mix channels are now Flexi channels (configurable as stereo or mono).

The SD11i offers 16 XLR inputs, 8 XLR outputs, AES input and output, 3 USB ports, MADI and a D-Rack interface so stage boxes can be interfaced to gain additional inputs. Waves integration is also an option. In addition, DiGiCo recently released the SD app for iPad that can control all of the mix parameters for the entire SD line.

Just a couple of months ago, a second model joined the Solid State Logic Live Series, the more compact Live L300. While smaller than the original L500, it still offers 128 mix paths (96 fully processed, 32 dry) and up to 568 inputs and outputs.

Onboard there are 16 XLR inputs, 16 XLR outputs, 4 pairs of AES ports and 8 MADI ports (4 redundant pairs). Optional stage boxes can be used to increase the input count via analog, AES or MADI inputs.

Rold Professional A/V M-5000

SSL has a nifty network called Blacklight II, a high bandwidth multiplexed MADI approach that can be used to reduce the number of interconnecting cables. Blacklight II carries 256 (at 96 kHz) audio signals, equivalent to 8 MADI connections, bi-directionally down a single multimode fiber optic cable.

Midas recently added the 40-input M32R desktop/rack mixer to the M Series. It has 16 XLR inputs, 8 XLR bus outputs, 6 (six) 1/4-inch aux inputs and outputs (and a pair of RCA jacks also on aux 5/6), AES50 and Ultranet network ports, and USB interface port.

Additional input counts and routing can be had by adding stage boxes, including the newer DL32 (32 XLR inputs and 16 XLR outputs) and DL16 (16 XLR inputs and 8 XLR outputs).Of note is that the DL16 offers dual AES50 ports so you can cascade a pair of them with no merger or router unit required.

Mackie just came out with the DL32R rack-mount mixer that’s wirelessly controlled via iPad and iPhone. The unit has 24 XLR inputs, 8 XLR/TRS combo inputs, 14 XLR bus outputs, and a stereo AES output. There’s an option card slot, and Mackie has just announced that a Dante card is now available.

A neat feature is that USB ports are also provided for feeding audio to a computer DAW or directly into a hard drive, with the Master Fader software providing the recording and playback controls.

QSC recently entered the mixer market with the TouchMix line, comprised of two very compact models. The TouchMix-16 has 12 XLR and 4 XLR/TRS combo inputs, 2 stereo TRS inputs, 6 aux XLR outputs, 2 stereo aux TRS outputs, and main LR XLR outputs. The TouchMix-8 (now shipping, by the way) offers 4 XLR and 4 XLR combo inputs, 2 stereo inputs, 4 aux XLR outputs, 1 stereo aux TRS, and main XLR outputs.

Both models sport a color touch screen for control instead of faders, and they can also be operated wirelessly via an iPad or iPhone. A new firmware update (2.0) provides added functionality, including support for password-protected, multi-level security access, expanded Wi-Fi options (including wired connection to an infrastructure router), and programmability of user buttons.

Behringer just added a desktop-style and three compact stage box-style mixers to the X AIR digital lineup. The new X AIR X18, XR18, XR16 and XR12 incorporate an integrated Wi-Fi module and are controlled using X AIR software for both iPad and Android tablets. The XR18 has 16 XLR/TRS combo inputs and 8 XLR bus outputs, while the XR16 offers 8 XLR combo inputs, 8 (eight) 1/4-inch line inputs and 6 XLR bus outputs.

The XR12 provides 4 XLR combo connectors and 8 (eight) 1/4-inch line inputs, plus 1/4-inch aux outputs and main output XLRs. All models have a USB connector for file storage or uncompressed stereo WAV recording and playback. A future planned firmware update will add Dugan auto mixing to the feature set. 

Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.

Posted by Keith Clark on 02/05 at 03:35 PM
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Wednesday, February 04, 2015

Ribbons Mics For Live: Adding Another Dimension

A range of options for adding some sonic creativity in the live sound realm...

It used to be that the fragile nature of ribbon microphones made them unsuitable for most live sound applications. But not any more – many recent models have been beefed up for added ruggedness, which is great because it allows us add their special sonic qualities to our live mixes.

Let’s explore the unique devices that are ribbon mics, including their design, technology, application, and techniques of use.

Physical Design
The majority of dynamic mics are based on moving coil designs, but ribbon mics are also a type of dynamic. They both convert sound into electricity using a magnet and a moving conductor.

In a ribbon mic capsule, a thin strip of aluminum foil (the ribbon) is suspended in a magnetic field (Figure 1). Sound waves vibrate the ribbon in the field and generate an electrical signal. The magnetic lines of force are cut by the ribbon as it moves, so a voltage is induced across the ribbon proportional to the velocity of the ribbon. So the ribbon is both the diaphragm and the transducer.

Typical ribbon dimensions are 1 to 2.4 inches long, 0.2 inch wide, and 0.6 to 2.5 microns thick. (A micron is 1/1000 of a millimeter!) Because the ribbon is so small and thin, it has an extremely low mass, which gives it a very fast transient response – producing a detailed yet warm sound.

The ribbon is corrugated like an accordion to reduce resonances and to prevent curling at the edges. If the corrugations are few and shallow, the ribbon tends to fatigue easily.

Figure 1: A diagram of ribbon microphone structure, along with a corrugated aluminum ribbon element with damping screen.

Being a short conductor, the ribbon has a very low impedance. A step-up transformer inside the mic raises its output impedance – and its signal level – to usable values. Older designs tended to have very low sensitivity because of the weak fields of their alnico magnets. Modern designs use neodymium magnets with higher flux, which, in turn, provides higher output.

Active models further boost output level by using a built-in, phantom powered mic preamp. The preamp also isolates the ribbon from loading and from miswired phantom power. The typical self-noise of a ribbon mic is less than 17 dB (A-weighted).

Ribbon mics tend to have a flat frequency response and a high-end rolloff because the ribbon is stretched very loosely so that it has a low resonance frequency, around 30 Hz or higher.

The resonance is damped by resistive fabric or screen near the ribbon, and by the resistance of the air gap between the ribbon and magnetic pole pieces. It’s also damped by the mic preamp’s input impedance. Above resonance, the velocity of the ribbon vibration drops 6 dB per octave.

The ribbon is open to sound waves on front and back. It responds to the pressure gradient (difference in sound pressure) between the front and rear of the diaphragm. At low frequencies with long wavelengths, the pressure gradient is very low.

At high frequencies with short wavelengths, the pressure gradient is high. The pressure gradient rises 6 dB per octave, then falls off at very high frequencies (small wavelengths) because both sides of the ribbon are exposed to successive sound-wave compressions.

Figure 2: Two factors that combine to create the frequency response of a ribbon microphone.

Figure 2 shows how the frequency response of a ribbon mic is produced. The ribbon’s velocity versus frequency falls 6 dB per octave above its resonant frequency. The pressure gradient across the ribbon rises 6 dB per octave.

Add those two effects together, and you get the net frequency response of the ribbon mic. It tends to be flat above resonance, then rolls off at higher frequencies, giving a smooth, mellow high end. At high frequencies, sound-pressure buildup on the ribbon and pole pieces helps to lift the highs. Then there’s the mic grille, which introduces small peaks and dips in the overall response, just as in any other microphone design.

Both moving coil and condenser designs have a resonant air chamber in front of the diaphragm. This Helmholtz resonator (air resonance in a cavity) boosts upper frequencies to create a more extended high-frequency response.

However, ribbon designs omit the resonator, so there is less ringing and phase shift in its signal at high frequencies – a “rounded off ” sound that is easy on the ears. Some mix engineers say the sound is particularly complementary to digital recording.

Polar Pattern
Because the ribbon is open to sound on both sides, it has a bi-directional or figure-8 polar pattern. As a result, it’s most sensitive to sound in the front and rear ofthe mic, and is almost “dead” on the sides (90 degrees off axis to the left, right, top and bottom).

That’s because side-arriving sound waves strike the front and rear of the ribbon equally and push it in opposite directions, so the ribbon doesn’t move. As a result, ribbon mics offer excellent isolation if their sides are aimed at unwanted sound sources. Some manufacturers also partially close the back area to create a hypercardioid pattern.

In addition, because the ribbon is very narrow, the polar pattern in the horizontal plane tends to stay constant with frequency – unlike with moving coil and condenser mics. So ribbon designs have very little off-axis coloration, and any leakage that does get in typically will tend to sound natural as opposed to colored.

What’s more, a figure-8 pattern has more proximity effect (up-close bass boost) than cardioid or supercardioid patterns. That huge proximity effect can be used to advantage. Place a ribbon mic close to the source to boost the lows, then roll them off with EQ – you’ve just created a noise-cancelling microphone!

Figure 3: Figure-8 polar pattern (2 kHz – 8 kHz) and frequency response of the Shure KSM313/NE.

Figure 3 shows a typical polar pattern and frequency response of a ribbon mic, while Figure 4 offers another polar pattern, frequency response and proximity effect measurement. Note that most ribbons are side-addressed, but models by beyerdynamic are end-addressed.

Getting Tougher
A ribbon is extremely thin and loosely stretched, so it can be deformed easily – even if corrugated. A sharp blow or air puff can damage the ribbon element.

However, new roadworthy designs use thicker material. For example, Royer developed a new ribbon thickness for the R-121 Live microphone that increases durability with little effect on gain and transient response. According to Royer, the sonic difference is negligible.

Figure 4: The polar pattern of the end-addressed beyerdynamic M 160, alongwith its frequency response and proximity effect.

Further, Audio-Technica’s proprietary MicroLinear ribbon imprint protects ribbon elements from lateral flexing and distortion, while the Shure KSM313 uses a “Roswellite” ribbon material that replaces the traditional foil ribbon. The benefits are additional resilience at extreme SPL, high tensile strength, toughness, and shape memory.

Some new models also have thicker grille wires, stronger housings, and heavy-duty shock mounts for added protection. Ribbon mics, in general, are not affected by heat or humidity, and current models can withstand very high sound levels, typically 146 dB SPL.

Does the propensity to pick up sound from the rear cause problems for ribbon mics on a live stage? Not really. Close mic’ing provides excellent isolation from leakage and feedback in spite of the rear lobe. That isolation comes from two factors:

1) The deep nulls in the polar pattern at the sides, and

2) The strong proximity effect, which, if rolled off, removes the lows in any leaked signal.

Because ribbon mics take the “edge” off the sound by rolling off the highs, they tend to work great on guitar amps, upright bass, strings and horns. By mellowing the sound of guitar amps, they can help guitars not compete for space with the vocals.

Ribbons also find use as drum overheads and vocal mics because of their smooth highs, which can be boosted with gentle EQ. Further, they can reduce excess sibilance in a vocal.

Many ribbon mics are not completely symmetrical. They might have a warmer sound at the front and a brighter sound at the back, offering some sonic options. The back side produces a signal that is opposite in polarity to that of the front, so if you mic an instrument or vocal with the back of a ribbon mic, be sure to invert its polarity.

Also, use a shock mount because ribbons are sensitive to mechanical vibration. It’s good practice to handle these mics carefully. Store them in their protective cases with the ribbon positioned vertically to prevent sagging over time due to gravity.

Don’t blow into them, and keep them away from power transformers to avoid hum. And, don’t put a ribbon mic in a kick drum because the force of the air puff can deform the ribbon.

When using a side-addressed ribbon mic with a figure-8 pattern, aim the 90-degree side of the mic at nearby sound sources that you want to reject.

For example, suppose you have several guitar amps side-by-side on stage. Aim a ribbon mic at each amp to reduce pickup of adjacent amps to the sides. Although close mic’ing provides the most isolation, you might prefer to place the mic 6 to 8 inches from the amp to get a less boomy sound.

Figure 5: Stage monitor placement near a hypercardioid microphone.

Ribbon mics do have some off-axis coloration (peaks and dips) in the vertical plane. So for the flattest response, aim the mic straight at the sound source, rather than tilting it up or down. This also results in the best transient response because the sound wave strikes the entire ribbon at once. However, Royer notes that angling the mic by 15 degrees or more protects the element in extremely high SPL situations.

With horns, place the mic near the center of the horn bell for a brighter sound, and away from the center for a darker sound.

A drum kit can be miked overhead with two ribbon mics. The side nulls will reject instruments, vocals and loudspeakers all around the kit. You might apply some high-end boost to compensate for the mic’s high-end rolloff, or just leave the EQ flat to soften the edgy tone of some cymbals.

When using a hypercardioid pattern with a singer, put the monitor loudspeakers at 110 degrees to either side off axis, where the pattern nulls are located (Figure 5). This will provide maximum gain-before- eedback. If a monitor is directly in front of the singer, aim the mic horizontally to point it’s null at the monitor.

Suppose you’re miking a singer/guitarist with a vocal mic and guitar mic—there could be phase cancellations because the guitar mic will pick up vocal leakage, but with a delay. Prevent that with a coincident pair of bidirectional ribbon mics. Place the mics horizontally with tops touching. Aim the null of the vocal mic at the guitar, and aim the null of the guitar mic at the singer. You just solved an otherwise “sticky wicket.”

The new breed of rugged ribbons can handle life on the road while providing the warm, smooth sound that many engineers love, as well as a wide range of options for sonic creativity.

Bruce Bartlett is a recording engineer, live sound engineer, audio journalist and microphone engineer (www.bartlettaudio.com). His latest books are Practical Recording Techniques 6th Edition and Recording Music On Location 2nd Edition.

Posted by Keith Clark on 02/04 at 04:07 PM
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Church Sound: The Keys To Improving The Mix Prior To Sunday

Great worship audio mixes start well before the first service of the week...

The worship audio mixer’s job is executed in the mix position during services, but success is mostly established outside the mix position prior to worship.

1) Rehearsal Music. Get whatever rehearsal music media is available to the worship team for review (legally). Learn the arrangements by listening during the week.

Not only will your mixes come together quicker for each song, you’ll also anticipate things like guitar solos or false endings before they happen – not just after they’ve already begun. Does it really make sense when everyone on the stage knows the songs and arrangements thoroughly, but the sound tech does not?

2) Pre-Production Meeting. Meet with the music/worship and production teams well in advance of each planned service. Reviewing plans and expectations can ensure an appropriate audio set up, and can avoid potentially tough sound reinforcement surprises.

Example: The worship department requests three wireless lavalier or headworn systems for a worship service. At sound check, they are placed on three actors and the tech quickly finds they’re not actors at all…they’re singers, and they’re asking for their vocals in the monitors!

If the mics omnidirectional it’s a tough situation at best, and practically impossible in many environments. Now, the worship department may have requested the drama style mics because the presentation or mood doesn’t suit the normal handheld vocal miking approach. But they didn’t anticipate the technical disaster that comes with their request. (Is it really their job to understand all of the tech stuff?)

Heading off this surprise at an advance meeting allows the audio tech to suggest a better miking technique, such as normal handheld vocal mics or possibly cardioid headworn mics. But the point here is not about which mic technique is right for this application, it’s that regardless of the chosen solution or compromise, it should be sorted out in advance – not at sound check.

3) RF Performance Check. If any wireless microphones, wireless in-ear monitoring systems, wireless assistive listening systems, or any other RF devices are used in the worship space, they must be properly installed and their frequencies coordinated for compatibility. Assuming proper installation, antenna orientation, and frequency coordination have been accomplished, it remains wise to periodically check RF performance. New sources of interference and other surprises are better found during testing – without an audience!

To properly check the systems, turn on all RF devices that will be on during worship, and turn on any equipment in close proximity to the RF devices. Portable transmitters and receivers should not be clustered together for the test—piling them together on a desk or other surface at the sound booth is convenient but a common mistake!  They should be at least several feet apart, and located on stage or in a general area where they will be used. The outputs of all devices should be auditioned over the PA or with headphones (RF mics), on headphones or earphones (IEM receivers), or the receiver/transducer that will be used by the worshipper (assistive listening device).

4) System Checks. Verify that the house system is in working order before Sunday morning. A brief walk/listen check a day (or a few) in advance can confirm that all PA zones/loudspeakers are working with no failures, and it’s wise to check other output zones too, like lobby, overflow, and monitor sends. A blown driver in the main PA cluster is not easy to resolve at 7:45 am on Sunday!

5) Optimize Mic Technique. Review microphone selection and placements on stage. Choosing appropriate mics and optimizing placement can influence the PA mix notably by reducing leakage, increasing gain-before-feedback, and capturing better sounding sources.

6) Cue Sheets. Get a copy of whatever cue/tech sheet or order of service outline is available, or draw one up if not. Clearly mark mic and roll-in cues, and any other important audio notes, in advance of sound check. Mixing notes can be added during sound check.

If mixing on a suitable digital platform, it may be possible to pre-program some or all of the cues and mix changes. But manual control should always be available, and the cue sheet should always be visible, whether in paper or electronic form. For very busy events, such as dramatic pageants, enlist an assistant to manage and announce the cues.

7)  Sound Check Is Not Set Up. Clearly distinguish between setup and sound check. Sound check is the time for audio personnel to dial in the mixes, with the elements (gear and musicians, etc.) working exactly as they will be during the worship service. Complete all audio setup work in advance of sound check so that sound check really is just that – sound check!

8) I/O Checks. Some worship audio techs add an input/output check procedure prior to sound check. This is highly recommended. I/O check takes a sound source (such as a CD), one person on stage, and one person at each mix position (two people in many church applications).

Every input and output is briefly tested over the PA system (inputs) and over wedges or earphones (outputs). It’s a 5- or 10-minute effort at most, and this procedure verifies the entire signal paths from sources to worshippers (front of house) and sources to artists (monitors). And the occasional I/O that doesn’t work is identified and hopefully resolved before the worship team hits the stage – preserving sound check.

9) Review Mixes. If you record your mixes, review them. If you’re making a classic “board tape” right off the console’s PA mix, listen to it with the knowledge that it is mixed for the house sound and does not include the live acoustic portion of the listening experience (which affects mix balance).

If you multi-track services, you’ve got a great practice and training tool – play the tracks back through the front of house console. And if you’re fortunate enough to own a digital mixing platform that offers “virtual sound check” technology, you’ve got the ultimate tool for practicing, training, and fine tuning the sound reinforcement mix.

10) Ear Training. Good mixing requires good listening skills, which require training and practice. Listen to great mixes that are relevant to your worship style, and “take them apart” mentally.

Discover the details that make good blends and mixes. Train your ears to identify frequency ranges. This skill is critical for sound reinforcement mixing, and there are a number of useful training tools on the market—opractice with a tone generator and RTA (real time analyzer). Some worship audio techs add an input/output check procedure prior to sound check. This is highly recommended. I/O check takes a sound source (such as a CD), one person on stage, and one person at each mix position (two people in many church applications).

Every input and output is briefly tested over the PA system (inputs) and over wedges or earphones (outputs). It’s a 5- or 10-minute effort at most, and this procedure verifies the entire signal paths from sources to worshippers (front of house) and sources to artists (monitors). And the occasional I/O that doesn’t work is identified and hopefully resolved before the worship team hits the stage – preserving sound check.

Kent Margraves began with a B.S. in Music Business and soon migrated to the other end of the spectrum with a serious passion for audio engineering. Over the past 25 years he has spent time as a staff audio director at two mega churches, worked as worship applications specialist at Sennheiser and Digidesign, and toured the world as a concert front of house engineer. Margraves currently serves the worship technology market at WAVE (wave.us) and continues to mix heavily in several notable worship environments including his home church, Elevation Church, in Charlotte, NC. His mission is simply to lead ministries in achieving their best and most un-distracted worship experience through technical excellence. His specialties are mixing techniques, teaching, and RF system optimization.

Posted by Keith Clark on 02/04 at 02:18 PM
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In The Studio: Did Einstein Do Audio Post?

This article is provided by the Fisher Creative Group.

There are some pithy quotes from uber-genius Albert Einstein where I swear he was talking about the trials and tribulations of audio post-production.

And whether you are building the DM&E to an indie feature or mixing your latest musical opus, ponder these thoughts.

“Three Rules of Work: Out of clutter find simplicity; From discord find harmony; In the middle of difficulty lies opportunity.”

There is always a tendency to throw everything into the soundtrack. I call it the kitchen sink approach. But as a consequence there can be too much going on and too much fighting for attention.

That’s when it’s important to step back and strip away all the unnecessary sound elements and find the root of the soundtrack, to find the message (and emotion!) of the piece that you are trying to convey via sound. And this is not an easy process.

As audio professionals, we often become enamored of our own work. And it’s painful to cut out the bits we worked so hard to include. But that is the creative process – working and reworking the audio until the story is best supported.

“We can’t solve problems by using the same kind of thinking we used when we created them.”

When you’ve acquired a significant amount of knowledge in a particular area, there is a tendency to rest on your laurels and take a similar approach to any new work that comes your way. This is a dangerous notion. You suffocate your creativity when you resort to the same old fixes for the same old problems.

For example, students and other would-be pros often ask me what EQ settings and such I use. And my reply is simple: whatever makes the track work. I never want to resort to ‘canned’ settings and instead rely on my ears to make the right choices to drive and support the story.

In fact, I don’t care what anything sounds like on its own as long as the mix as a whole works. Individual sounds can be rather thorny when soloed, but in the context of the entire DM&E mix, this thorny sound may fit in perfectly. In short, every problem is unique and every solution to said problem is equally distinctive.

“Logic will get you from A to B. Imagination will take you everywhere.”

This is a corollary to the above mentioned approach. Don’t rely too much on the way things should be, or worse “the way we’ve always done it ’round here’.”

Instead use your tools to make the sound come together as it should in the service of the project. It’s the soundtrack as a whole that must work, and only when expanding your approach (and your creativity) can you discover the right way.

“You have to learn the rules of the game. And then you have to play better than anyone else.”

I think what Einstein was saying here is that there are no shortcuts, no easy fixes. I have a client who is always looking for the easy fix: What ____fill in the blank___ will make my audio better, less noisier, and perfect? And the answer to that blank is simply this: hard work and even harder won skill.

Acquire good sound to begin with and then use your post audio tools to make it shine. There is no magic button for fixing audio—it takes effort to craft a seamless soundtrack that makes an impact on your audience. It is the raw skill that really matters and not the tools.

“Sometimes one pays most for the things one gets for nothing.”

This applies to all those free VST plug-ins you find on the web. Stick to the tried and true technology and you will have fewer issues. And never upgrade your DAW software in the middle of a project.

“Imagination is more important than knowledge. For knowledge is limited to all we now know and understand, while imagination embraces the entire world, and all there ever will be to know and understand.”

There is nothing I could say that could enhance this quote. Thanks A.E. for dabbling in audio post. You make the work easier.

Of course, there are dozens and dozens of quotes from the venerable Mr. E., so do yourself a favor and read up on the man. You just might discover he had far more insight into art than you might have thought.

Jeffrey P. Fisher provides audio, video, music, writing, consulting, training, and media production and post-production services for individuals, corporate, and commercial clients through his own company, Fisher Creative Group. He also writes extensively about music, sound, and video for print and the web and has authored numerous books and training DVDs.

Posted by Keith Clark on 02/04 at 10:32 AM

Tuesday, February 03, 2015

The Conundrum Of “Ears Versus Education”

For the best results in audio mixing, context is vital. But can it be taught?

I’ve been thinking quite a bit about the role musical education plays in audio mixing. There have been numerous threads about the subject in several on-line forums, and the responses seem evenly divided between “not needed but it doesn’t hurt” and “it’s actually a hindrance” and “it certainly helps.”

Because I earned a degree in music performance, I’m biased on the subject, with my opinion leaning toward the “it helps” camp. Still, I can’t help but wonder if it really does…

When evaluating the handiwork of mix engineers, there are plenty of guys and gals that indeed do not have formal musical training. An obvious example is Al Schmitt, who’s earned a stockpile of Grammy Awards for efforts with artists such as Frank Sinatra, Toto, Diana Krall and numerous others.

Even though he’s a studio engineer, I think his example can still be applied to sound reinforcement. One thing’s for sure – Mr. Schmitt has never been called “unmusical,” or at least I’ve never heard it said.

My hunch is although he doesn’t have “formal” musical training, he still has listening skills quite sensitive to musical aesthetics, an amazing sense not only for the technical but also for how all of the sounds relate to one another in context.

This leads us to a key point: for the best results in audio mixing, context is vital. But can it be taught?

Matter Of Style
With any art form, there are those who specialize in a particular style and then those who seem to be able to transcend their particular era and become “timeless.” Relating this to audio, I’ve heard mix engineers who seem to meld their style of mixing to the music itself, while others try to force the music into their mixing style.

Back when I was touring as a mix engineer for the Airmen of Note (U.S. Air Force Jazz Band), I found it was important to spend time with the band in rehearsal to get a sense of the issues at hand: arrangement, internal balance within sections and between sections, and the general “feel” produced by the music.

In the process, I came to the conclusion that the drums, along with the bass, generate a certain rhythmic element that actually drove the way the horn players stayed “in the groove.” It was an actual physical thing, where the acoustic wave from the kick drum had an impact on the diaphragms of the horn players. Stand close enough to this type of group while they’re playing, and you can pick up this sensation.

So I set about trying to bring some of that feel to the audience while I mixed, but without making it too overpowering or “rock ‘n’ roll” – which I felt would not be representative of the big band style. The approach involved how I mic’d the drums (three mics – kick, and two overheads), use of EQ (not much, except to bring out certain things and make sure other elements didn’t become overbearing) and setting the drum levels relative to the rest of the mix (supporting the sound).

I felt that the result was a convincing live portrayal of the band, bringing out the dynamics and impact they worked so hard to do attain, but without too much power from the rhythm section. But did my music education help me attain this, or was it some innate musical sense that can’t be taught?

The Inner Voices
Another aspect of mixing, and it was clearly important in big band work, is the inner voices. No, I don’t mean the little voices in my head saying, “check out that woman in the third row.” Rather, I’m referring to the relationships of all the instruments between the bass and cymbals.

Any arrangement - rock, jazz, classical, or whatever - relies on specific voicings. I’m talking about the order of notes from the lowest to the highest within a chord. As a mixer, if you’re not aware of this, then you likely don’t realize that the third of a chord determines whether it’s major or minor, that the fifth along with the root make up the “frame” of the chord, and that everything above the fifth is harmonic embellishment but nevertheless important in terms of leading notes, harmony, and what kinds of scales might be used for melodic material.

And perhaps the mixer might miss (or not know) that inversions (chords where the root, third, etc. are stacked out of order) are extremely important to musical harmony, and thus are a critical element of a musical style like jazz. An example is the horn section for a swing band (think Brian Setzer’s Dirty Boogie), where if one of the horn mics is turned up too “hot,” then the wrong note in some chords may be emphasized. The difference might be subtle, but it may also throw a certain amount of “aural sand” into the musical experience for at least a portion of the audience. And let’s face it – it’s just not right.

But these are “rules of thumb” taught by the educational process. Another way to figure out “who’s playing what” might be to listen and think, without cluttering up the works with confusing terminology. In other words, how do you think it sounds?

The New Response
“The most important tool in audio is… ?” I ask this question often when giving presentations. It used to be that the answer I wanted to hear was “our ears.” Recently, however, I’ve preferred the response of “our brains.”

Of course, good ears are a critical component in mixing, and without them, there wouldn’t be much of a purpose for audio systems. (Although I’m sure that marketing departments would find a way to put a spin on that!)

But my thinking began to change as I realized that without the brain, what the ears are telling us can’t be interpreted and no plan of action can be developed. In other words, we may hear a problem, but if we can’t produce a solution, then what’s the point?

For example, if there’s a buzz in the system, is it at 60 Hz? 120 Hz? 180 Hz? And if it’s indeed at 60 Hz, where to start in looking for a solution?

On the flip side, those without the sense to apply their knowledge in order to generate an aesthetically pleasing mix lead me to question the value of any understanding of things like gain structure and signal flow, let alone voicing and spatial relationships. In other words, it may be technically “right” but does it sound good?

Perhaps their mixes are “good enough,” and certainly any situation involving art and technology must by nature be a form of compromise. However, if you knew of a way to improve your mixes, wouldn’t you want to employ it?

Equal Value
My resolution to these conundrums has been to settle on the theory that both musical ears and musical education have relatively equal value, and therefore, for better mixes, the focus should be on both. My theory guidelines track along these lines:

- If considering attending an audio school, see if the curriculum includes courses in musical training (ear training, theory, etc.). Purely technical audio training can result in a set of skills, but musical training allows you to “speak the language” with musicians and within your own mind.

- Spend a lot of time listening to a wide variety of music, and try to determine the common elements between them as well as those things that distinguish between different styles. It’s also vital to listen to acoustic music as much as possible – if you don’t know what instruments sound like un-amplified, where is your frame of reference?

- Come to terms with your own mix style and types of music. There are even differences between punk music from New York and L.A., right? (I suppose I’m showing my age with that one.)

- If the music you’re mixing was developed before amplification (classical, big band jazz, etc.), understand the context, both musically and in terms of acoustics. For example, what types of rooms originally hosted these types of performances? In other words, why put major amounts of reverb on a baritone sax solo in a big band performance? It just doesn’t fit. Not only that, but the players and the audience will expect to hear it as it is supposed to sound.

The track record of many successful folks working as mixers in pro audio without a formal musical education makes a persuasive argument that such an education may be largely irrelevant towards enhancing mix skills. Perhaps their abilities and success are a matter of an innate, natural musical sense, along with great ears and a lot of real-world experience.

Yet it also begs the question: would they be even better at what they do with further learning? Aren’t we all usually better for having learned more?

Karl Winkler is director of business development for Lectrosonics and has worked in professional audio for more than 20 years.

Posted by Keith Clark on 02/03 at 03:43 PM
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Wednesday, January 28, 2015

A Look At Top New Gear At 2015 NAMM—Part 2

Yesterday I presented the first portion of a list of noteworthy products making their debut at this year’s 2015 NAMM show in Anaheim, which concluded on Sunday. (Go here for part 1.) I’m continuing the report here with part 2.

As I noted, the NAMM convention brings announcement after announcement of new products, from pro audio gear to percussion pieces—and the 2015 edition of the show proved no exception.

For the lucky few of us who were there, we got to see the new gear up close, and even put our hands on it. Wading through the sea of press releases, amazing new devices—and a few eyebrow raisers—I’ve culled a list of gear that caught my eye. (In a couple of cases, the products had been previously introduced, but this was their first show appearance.)

So here’s part 2. I’ve called out the facets that are the most interesting to me, but be sure to click on the link for each to get additional information.

VUE Audiotechnik Ultra-Compact h-5 Loudspeaker. A problem-solver for stage lips and underbalcony locations that require loudspeaker coverage. The h-5 provides that coverage with the same audio quality seen in the larger h-8, h-12 and h-15 loudspeakers. It offers 120- x 40-degree coverage, a neodymium HF compression driver with proprietary Truextent beryllium diaphragm, and dual channel high-efficiency amplifiers devoid of cooling fans. 

The enclosure includes M10 hanging points with the option of additional rigging hardware for those stage lip and under-balcony locations. The surprising feature is the transparent candy-apple red grill finish which adds a nice splash of subtle color. Call me superficial but it’s nice to pair functionality with appearance.

Pro Tools 12 and Pro Tools | First. Avid is making a significant change in how it sells Pro Tools with the release of version 12 and Pro Tools | First. Pro Tools 12 can be purchased outright or can be licensed per month for as low as $29.99 with all software updates provided immediately via the cloud. Pro Tools | First is a new free version of Pro Tools that includes limited cloud storage and a subset of Pro Tools effects, sound processors, and plug-ins. 

In addition, Avid’s Cloud Collaboration provides cloud-based storage for working from anywhere and working with others on the same project. The software-as-a-service model is perfect if you only use Pro Tools a few times a year.

dbx DriveRack VENU360 Loudspeaker Management System. Places loudspeaker optimization in the palm of your hand – pardon the cliché.  Successor to the dbx DriveRack 260, it adds mobile device control, additional input channels, improved DSP and includes dbx’s useful Advanced Feedback Suppression. 

It can be accessed via any iOS, Android, Windows or Macintosh device via an easy-to-use app and a standard Wi-Fi router connected to the rear-panel Ethernet port. Loudspeaker optimization is a science and having the mobility to make changes at any location in the venue makes that job a bit easier.

Blue Microphones Hummingbird. A cardioid condenser mic based on the Blue Bottle B1 capsule that places the capsule on a 180-degree swivel mount making it a great low-visibility snare microphone. The Hummingbird is rated at a maximum SPL of 130 dB with a frequency response from 20 Hz to 20 kHz. 

According to Blue, it offers plenty of sparkling high end for use with drum overheads, acoustic guitars, and other stringed instruments. At first glance, the pivot head seemed a novelty since the NAMM display had it hanging over a tambourine. However, it does provide the ability for sound amplification with a cleaner-looking stage.

PreSonus StudioLive 48AI & 64AI Mix Systems. There was previously some connectivity between StudioLive consoles, but PreSonus has taken that to the next level with the 48AI and 64AI. Using two StudioLive 24.4.2AI mixers or 32.4.2AI mixers and a joining adapter creates an expansive mix system with the centrality of master setting controls through the primary mixer. 

Both mixers can be operated wirelessly and controlled through PreSonus’s StudioOne software. The 64AI system runs just under $7,000 and both systems include a PRM1 reference microphone for use with integrated Smaart software. If you’re StudioLive console lover, this should make you even happier.

Rane DR6 Touchscreen Remote Control. A new interface for the company’s HAL system with a simplified LCD wall-mounted interface for conference halls, schools, and churches. The 7-inch LCD display is easy to use and Rane recognized the need for limiting controls based on the logged in user.

Therefore, with User Access secure management control, people renting a conference hall can use only the components they’d need to use. Having worked with novice users on auto mixing systems, the DR6 is something they could easily use with limited-to-no instruction.

AKG DMS800 Digital Wireless Microphone System. The next step up from the DMS700 V2, is taking the word digital to a new level with audio outputs for both Dante and AES EBU as well as existing HiQnet support for remote control and monitoring.

The handheld microphone is built with interchangeable heads with the AKG D5 WL1, D7 WL1 and C5 WL1. The DMS800 isn’t for everyone with a base price of $1,899 but considering the flexibility and Dante integration, it might be exactly what you need. 

Behringer X AIR Portable Mixers. Available with 12, 16, and 18 channel inputs, they’re the portable baby brother(s) of the X32. Allowing for all types of inputs, from hi-Z to MIDI to the standard array of input connectors, the brick-shaped I/O box also includes an Ethernet jack and wi-fi for remote access via the X AIR for iPad app. 

Tack on the Ultranet port for connecting to Behringer’s P-16 personal monitoring system and you get a small-form factor, high-ability piece of hardware. The X AIR portable unit can even perform multi-track recording via the USB jack. Considering the 12-channel X AIR runs around $299, it’s more than worth considering.

Shure PG ALTA Microphones. The company has replaced the black and silver entry-level PG microphones with this new all-black series. An addition is the PGA181, a side-address condenser microphone great for acoustic instruments, as well as the PGA98D, a gooseneck condenser microphone for drums.

Shure also made a simple yet-brilliant change to the PG56 drum mic (now the PGA56) clamp by replacing the hard-to-use wing nut-style handle with a quick-release clamp common to bicycle seats and wheels. Using the new clamp enabled me to set the microphone angle and keep it after closing the clamp. No more of the set, twist, reset, twist. Sometimes it’s the little things…

Allen & Heath ZED Power 1000 Mixer. An 8-input, 2 x 500-watt powered mixer with USB stereo recording and playback, onboard effects, and 9-band EQ. It can route outputs to L+R speakers, mono LR and foldback, or even mono LR and sub.

Though I usually shudder at the thought of a powered mixers based on my experiences with cheap models in the past, the Power 1000 is built on the ZED platform with Neutrik jacks and secured controls, so it’s a unit that I would confidently use. 

Soundcraft Ui Virtual Mixers. The Ui12 and Ui16 provide the ultimate in “plug-and-play.” Not only can up to 10 remote controls (tablets and smartphone) be used for mixing in ear monitors, but the rack-based system includes its own wi-fi router.  They also include signal processing from dbx, DigiTech and Lexicon as well as recallable and remote controllable mic gains.

Another feature is the real-time frequency analyzer (RTA) on inputs and outputs. Either of these mixers would be a welcome addition for any roving band. Soundcraft even includes a virtual demo of the remote control via the company website.

Midas PRO X Console. Built around the new Neutron Audio System Engine, it’s capable of processing 800 audio channels. The Neutron system also allows for channel routing on a point-to-point basis and changeable even on individual automation scenes.

The PRO X offers 99 mix buses that can be simultaneously displayed as 24 mono or stereo mixes on the console surface. Each displayed mix has its own LCD select switch with color coding and scribble strips, plus LED metering.

Mackie FreePlay. The perfect gift for the singer/songwriter in your life. This 300-watt personal PA system with an 8-inch woofer can take in two inputs (mic or line) and an auxiliary stereo signal, feed a monitor, and run for 10 hours on the rechargeable battery.

The real beauty of FreePlay is in the remote mixing capability. Users can perform a sound check or just start performing while another person remotely controls the mix, including the effects, and they can also stream music directly to the unit via Bluetooth for adding loops and backing tracks. 

Go here for part 1 of this report.

Chris Huff writes about church audio at Behind The Mixer, covering everything from audio fundamentals to dealing with musicians.

Posted by Keith Clark on 01/28 at 03:15 PM
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Tuesday, January 27, 2015

Church Sound: The Impact Of Acoustics On Worship Music Styles

If you asked people how they fell about “room acoustics,” you’ll find that many don’t even know what it is. We’re immersed in reflected sound every minute of every day, but it’s largely taken for granted unless it becomes very annoying.

In a given congregation, it’s likely that the majority don’t really care about the acoustics of the proposed new assembly hall. A heavily padded room that’s suitable for electronically amplified music and speech production is fine with them.

However, it’s just as likely that there will be some people who are passionate about how the room sounds. They may prefer traditional music and have an appreciation for the acoustic environment required to enhance its presentation.

Indifference and passion are not a good mix, especially when you consider that the possible solutions are potentially very expensive. Let’s look at both sides of this issue rather than present a singular viewpoint.

Traditional Western music finds its roots in Europe, where it evolved and developed over many centuries. Many of the great composers were full-time employees of the state, and the composition and performance of musical works had great social and political significance.

Even today, every major European city (and most smaller ones) have one or more concert halls where music aficionados can enjoy a classical performance.

Music has been around much longer than sound reinforcement systems. For most of human history, the acoustic environment was looked to for amplification and tonal enhancement.

Many of the most popular musical pieces in history were written to syncopate with reflections from the room, and could only be presented properly if the room volume was correct. Room reflections were the only way to amplify music instruments.

Wind, string, and brass ensembles used shells to provide amplification so that larger audiences could be entertained. Reflected sound plays a vital role in the presentation of traditional music.

Another source of amplification was the use of ensemble of the same instrument. A violin section is much louder than a single violin, and the inherent timing and pitch differences produced a sound rich in harmonic content.

As sections were combined to produce larger ensembles, a conductor was required to keep everyone playing together, and to provide visual cues for loudness and tempo. It became apparent that a skilled conductor was vital to a good performance, and many of them gained greater notoriety than the musicians themselves.

Any classically trained musician is fully indoctrinated into the importance of room acoustics in live performance.

Music schools across the country have rehearsal and performance halls that have been carefully sculpted to envelop both the musicians and audience with reflected sound.

While virtually all traditional musical instruments depended on the stage and room for acoustic support, one instrument developed that took it even further. For a pipe organ, the room is part of the instrument. A dense reverberant field envelopes the listening with sound from the many pipes used to generate the musical tones.

Rooms that are deemed suitable for pipe organ (and the human equivalent - a choir) have reverberation times in the 4- to 5-second range. This means that the reverberation time must be even longer if the room is empty. Since communication was next to impossible in these spaces, liturgical services were sometimes sung rather than spoken, often in a language unfamiliar to the congregation.

Here And Now
The era of electronic music began in the mid-1960s with the invention of the electric guitar. The first versions were simply acoustic instruments outfitted with electronic pickups. They could be played with or without electronic amplification.

The solid body electric guitar had very little acoustic sound, and could only be heard when plugged into a guitar amplifier. They had a characteristic sound determined mainly by their pickup design and configuration, and the density of the wood to which the pickup was mounted.

The guitar amplifier had such a profound effect on the sound that it was soon recognized as port of the instrument itself. Various makes and models became preferred for their characteristic sound, and a good amplifier technician could modify the electronics to produce a sound pleasing to the musician.

Because the guitar and amplifier could product almost any sound level, there was no need to rely on room acoustics for support. In fact, since the instrument was plucked rather than bowed, the staccato sound could easily be aggravated by echoes in the auditorium, and the desired sound of the instrument lost in a sea of reflections.

Add to this the fact that acoustic drums almost always accompanied the guitar. These were quite loud and also staccato, and their sound was equally aggravated by echoes from the room.

Musicians and club owners quickly learned that acoustically live rooms could dramatically detract from a performance. Musical venues were draped with curtains and padded seats in an effort to reduce the room reflections.

Of course, non-electric sources like the human voice and brass instruments required amplification to keep up with the amplified sources. Thus, the sound reinforcement system was the natural solution to the problem.

Before long, everything was electronically amplified, and a band could play at virtually any sound level and for an audience of almost any size.

Obvious Advantages
There were obvious commercial advantages to amplified music. A chamber orchestra can only play to a few hundred people, and the best concert halls could hold less than two thousand. A larger hall produced reflections that were too late to provide the proper support.

On the other hand, an electronically amplified band could play to packed stadiums of ticket buyers. After the show, the sound system could be loaded into a truck and moved to the next city on the tour.

The only way to get consistent sound quality was to make the production independent of the room acoustics—exactly the opposite objective of traditional music.

So it’s not surprising that both viewpoints have carried forward into modern assembly spaces. The formally trained musicians and musical directors see the space as an amplification system for the organ, choir, and acoustical instruments. They want a live space with carefully sculpted reflection support, including an orchestra shell and reflecting clouds over the audience.

The contemporary music camp wants an acoustically dead space where the sole source of sound are the amplifiers and sound reinforcement system. A sound system is in control of everything—for better or for worse.

The operator plays a role similar in ways to the orchestra conductor, with the musicians playing and the operator determining the relative balance between the instruments as they are amplified through the reinforcement loudspeakers.

Whereas classical musicians sit in close proximity to hear each other, a contemporary band uses foldback (stage monitoring) to hear what is going on, again under the control of a technician. (Modern in-ear monitoring systems offer control to the musicians.)

Increase Early Energy
The formally trained acoustician shares the mindset of the classically trained musician. Their objective is to increase the early acoustic energy in the space to provide support for acoustic sound sources.

The room geometry and surface coverings are selected to provide natural amplification. Hardwood, plaster and even glass are popular choices.

But if the space is to be used for contemporary amplified music, these room reflections are unnecessary and even detrimental. What is usually needed is large amounts of absorption in the space—an idea that is anathema to the classical mindset.

The surface coverings of choice are velour drapes, padded seats, and carpet. The idea is to “kill off’ the room and place full control of the sound in the hands of the mixer operator.

It should come as no surprise that the acoustic advice provided by one “expert” may completely contradict that given by another. It depends on their mindset, and which musical type they are trying to enhance. This power struggle has raged for years, and is probably getting worse rather than better.

We have witnessed the development of the need for the “sound reinforcement acoustician” that understands the acoustical requirements of electronically amplified sound.

Have It Both Ways
One solution is to create a variable acoustic environment. This can be accomplished physically by mechanically changing the surface coverings of the space by installing reversible panels of retracting drapes.

It’s a viable solution, albeit potentially expensive. A more modern approach is to vary the acoustics electronically. This requires a large number of auxiliary loudspeakers that are used to produce the “reflected” sound field. Done properly, these systems can be quite convincing and represent a method of having both a dead and live room that can be selected with the push of a button.

The most important thing to understand is that the room acoustic criteria for contemporary and traditional music are mutually exclusive - one can’t be improved without sacrificing the other. Attempts to reach a “middle ground” can make a space non-optimal for either type of music.

The greatest potential for “having it both ways” lies in properly designed and implemented electronic acoustic enhancement systems. These can often be implemented at a total cost similar to the expensive acoustic treatment required to make a room suitable for only one type of music.

Suppliers of these systems can arrange demonstrations to allow those interested to hear the system in use so that they can judge for themselves if such a system will meet their needs.

The real key to walk into any discussion of acoustic with a basic understanding, and to work with an acoustical professional who seeks to best serve the musical styles offered at your church.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops online and around the world. For more information go to www.prosoundtraining.com.

Posted by Keith Clark on 01/27 at 07:14 PM
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A Look At Top New Gear At 2015 NAMM—Part 1

The NAMM convention brings announcement after announcement of new products, from pro audio gear to percussion pieces—and the 2015 edition of the show last week in Anaheim proved no exception.

For the lucky few of us who were there, we got to see the new gear up close, and even put our hands on it. Wading through the sea of press releases, amazing new devices—and a few eyebrow raisers—I’ve culled a list of noteworthy products making their debut at this year’s show. (In a couple of cases, the products had been previously introduced, but this was their first show appearance.)

Here’s the first dozen, with another 12 or so to follow tomorrow. I’ve called out the facets that are the most interesting to me, but be sure to click on the link for each to get additional information.

Audio-Technica System 10 PRO Rack-Mount Digital Wireless System. The new System 10 PRO Rack-Mount (pictured here with Scott Shaw of A-T) solves the problem of limited wireless range. Mounting receivers on stage removes front of house monitoring, and traditional antenna extenders are expensive.

It allows for extending the range by pulling out an antenna cartridge from the receiver, mounting it elsewhere (up to 300 feet), and running a Cat-5 cable between them. Add in daisy-chaining up to 10 of the receiver units via RJ12 cable, and this was my favorite new release at NAMM.

Electro-Voice EKX Portable Loudspeakers. Available in both passive and powered loudspeaker configurations. Powered models have QuickSmart DSP processing, high-efficiency Class D power amplifiers, and intelligent heat management.  Both types come in 12- and 15-inch 2-way versions and 15- or 18-inch subwoofers.

EV provided a 5-minute demonstration of multiple configurations, including 2-way only as well as full 2-way and subwoofer pairings. These loudspeakers provide solid mids and crisp highs, with a nice amount of low end, while the subs deliver deep lows with clarity and substance. It was hard to leave the demo. 

Allen & Heath QU-PAC Digital Mixer. A slim-lined unit that provides the remote controls for complex mixing while providing a tactile interface, with touchscreen, appropriate for schools, hotels, and conference centers where simplicity is key.

The touchscreen interface is easy to use and understand. As an added bonus for multi-use venues, the QU-PAC includes permission setting to control what a user may or may not do. This enables the venue’s tech to set up the system but prevents anyone else from screwing it up. (And yes, that’s a technical phrase.)

Shure MOTIV Digital Microphones & iOS App. The MV88 is a stereo condenser mic capable of connecting to any iOS device with a Lightning plug. The MV5 is a USB microphone with three onboard DSP presets (vocals, flat, instrument).

The show-stealer is the rugged metal-cased MV51, which combines a large-diaphragm condenser mic with the new MVi touch-panel interface with quick access to gain, mute and headphone monitoring volume adjustments as well as five onboard presets. I could have dropped it with no fear of damage. The MV5 and MV51 can also connect to iOS devices and work with the ShurePlus MOTIV app for real-time adjustments and the sharing of files. 

Line 6 Relay G70 & G75 Wireless Systems. Multi-instrumentalists give audio engineers a headache every time they swap instruments while using the same pedalboard. Line 6 has come to the rescue with the G70 and G75 wireless systems, which feature hands-free switching between up-to 16 different color-coded transmitters with locking cable connectors and either amp top or pedalboard-designed receivers.

Gain control is set at the transmitter so all instruments send the same signal level through to front of house. Tap the receiver’s button until the button ring color matches that of the transmitter. With a latency time of 1.48 ms and zero signal compression, bass players can hear the lows as if they were running straight through their amplifier. As one who has to ride the fader every time a musician switches instruments, I’d love to have a Relay system in my toolbox.

Mackie DL32R Wireless Digital Mixer. An impressive 32-channel remote mixing system that allows for offline setup, multi-track recording, individual personal monitor mixing, and virtual sound check all through the easiest interface. Just announced at the show is the new DL Dante expansion card that provides 32 x 32 channels of network audio I/O.

Another great feature is the metering view of all channels at once – the perfect way to see which channel might have a problem. According to the app designer, it’s the smoothest metering display on the market. Having examined several app-based mixers, the Mackie app is my favorite. 

Sennheiser evolution wireless (ew) D1. The D1 could be considered the ultimate plug-and play wireless microphone system. Through automatic unit pairing and transmission frequency selection, any number of units could be set up in moments and the built-in frequency scanner keeps them on the clearest of frequencies.

Then come the more unusual features of the microphone: automatic gain control, 7-band graphic EQ, de-esser, and low-cut filter. It’s not a microphone I’d use in conjunction with a digital console where EQ work would better reside. However, the seven-band graphic EQ could be used to tailor the microphone to a person when gigging and using analog consoles.

JBL SRX 800 Series Powered Loudspeakers. Equipped with JBL transducers, Crown’s DriveCore Technology, user-configurable DSP, and full HiQnet Network control. The onboard DSP includes 96 kHz FIR (Finite Impulse Response) filters for crossover tuning and delay adjustment.

Available is the 12-inch 2-way SRX812P, 15-inch 2-way SRX812P, and dual 15-inch 3-way SRX835P, as well as the single-18-inch SRX818SP and dual-18-inch SRX828SP subs. Output ratings are stated as between 135 dB and 141 dB, depending on model. Expect the distinct bass feel common to JBL.

AKG Session I Drum Microphone Set. The starter Groove Pack drum microphone kit, containing six mics with three for toms and snare, has gained a microphone while keeping the same price.

The new Session I kit (picutred here with Erich Gaertner of AKG) adds another P4 dynamic cardioid mic to the mix. While a drum kit can technically be captured with only a couple of microphones, I’ve found seven a requirement for most live situations. Look to the Session I for a good, basic but complete drum mic kit.

Midas M32R Digital Console. The M32 has shrunk in terms of size but not in capability. The new M32R has 40 input channels controlled over one 16-fader surface. The upper console area, with channel controls and display screen, does a mixed job with the smaller area of real estate.

The controls are condensed into a smaller space but without any impact to usability. Where the M32R does fall a tad short is in the reduced size of the display screen. The saving grace is that like it’s big brother, it includes iPad, Android, Mac and PC remote mixing. 

Countryman Isomax 2 (I2) Instrument Microphone. The flat-response, low-noise I2 is a low-profile microphone that can be added to a number of Countryman-designed instrument mic clips for optimal placement on brass and string instruments as well as pianos.

It also comes in the Isomax 2-H hanging mic with a non-twisting cable with bendable end for exact positioning – important when using a non-omni microphone. Both come in omnidirectional, cardioid, and hypercardioid polar patterns. Testing the I2, the deep nulls in the hypercardioid demonstrated rejection of stage noise so often a concern with live engineers.

Soundcraft Signature Series Analog Consoles. Actually more than a line of analog consoles. As analogs go, ranging from 10-input consoles to 22-input consoles, these have everything expected plus two sweeping mid-range EQs per channel and built-in Lexicon studio-grade effects.

Then comes the bonus. The 12- and 22-channel versions include USB multi-track recording, a regular feature of digital consoles. My only complaint is the top-mounted plugs on the larger 22-channel console, but it’s sacrifice to keep the price down, the Soundcraft rep told me. For a 20-plus channel multi-track recorder, I can live with it. 

Radial Engineering JDI Stereo & J48 Stereo. The popular JDI passive and J48 active direct boxes offered in stereo, both available in February for about $299.

Additionally, Radial announced the new IceCube IC-1, a compact balanced line isolator for elimination of hum and buzz caused by ground loops in audio systems. It’s rated as providing a linear response from 20 Hz to 18 kHz. When ground loop problems can’t be eliminated, it’s good to know the IceCube, at only $69, can solve the problem.

Be sure to check back in tomorrow for Part 2!

Chris Huff writes about church audio at Behind The Mixer, covering everything from audio fundamentals to dealing with musicians.


Posted by Keith Clark on 01/27 at 10:47 AM
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Monday, January 26, 2015

Mic Techniques For Taming The Live Stage

Approaches for controlling feedback and leakage as well as fostering delivery of clean, natural sound

Let’s face it—the live sound reinforcement realm presents some microphone challenges that regularly threaten sound quality.

Look at the conditions. The monitors feed back. They leak into the vocal microphones and color the sound. The bass sound leaks into the drum mics, and the drums leak into the piano microphones.

And then there are the other mic-related gremlins breath pops, lighting buzzes, wireless-mic glitches, and even electric shocks.

So let’s have a look at solving at least some of these problems. Based on the experiences of live sound mixers and technicians, these suggestions will help control feedback and leakage while also fostering a clean, natural sound to the audience.

Get In Close
The first tip is to try to get in close to sources with directional mics. To start, place each mic within a few inches of its sound source. Close miking increases the sound level at the microphone and makes the sound system louder.

Use unidirectional mics to reduce feedback and leakage. They reject sounds to the sides and rear of the mic, such as floor monitors. Some examples of unidirectional patterns are cardioid, supercardioid, and hypercardioid.

Sometimes locating a mic right at the source can help. (By the way, that’s a SP25B condenser from Applied Microphone Technologies.)

Most directional mics boost the bass when you mic close. This is called the proximity effect. At low frequencies, it provides free gain (extra volume without feedback). If you want to roll off this excess bass with your mixer EQ, you also reduce any low-frequency leakage picked up by the mic.

Next, here’s an extreme way to get plenty of level into the mic: place the mic near the loudest part of the musical instrument. Some typical positions are near the sound hole of an acoustic guitar, in the bell of a sax, or inside the shell of a tom-tom.

Use this method as a last resort because close miking tends to color the tone quality, giving an unnatural sound. Here’s why: most musical instruments are designed to sound best at a distance (say, 1.5 feet or more away). So a flat-response mic placed there tends to pick up a natural or well-balanced timbre.

But when you get close, you emphasize the part of the instrument that the mic is near. The tone quality that is picked up very close may not reflect the tone quality of the entire instrument.

For example, the sound hole of an acoustic guitar resonates strongly around 80 Hz to 100 Hz. A mic placed close to the sound hole hears and emphasizes this low-frequency resonance, producing a bassy, boomy timbre that does not exist at a greater micing distance.

This placement likely emphasizes low-end resonance.

The close-miked sound is harsh, too. To make the guitar sound more natural when mic’d close to the sound hole, you need to roll off the excess bass on your mixer, or use a mic with a bass roll-off in its frequency response. Also dip out some 3 kHz to reduce harshness.

A sax miked in the bell sounds like a kazoo. To mellow it out, cut around 3 kHz and boost around 300 Hz. And if you can get adequate gain-before-feedback with mic positions that sound more natural, by all means do so.

Making Contact
Another approach is to use contact pickups in tandem with the mics, which can help solve feedback problems because it’s sensitive to mechanical vibrations, not sound waves.

A pickup for an acoustic guitar usually sounds good near or under the bridge. Unfortunately, the guitar sounds electric with a pickup because it misses the acoustic string sounds.

Many engineers have had success with a hybrid method that combines a pickup with a mini mic. A pickup mounted under the bridge picks up the lows and provides volume and punch. A mini hypercardioid mic is mounted just inside the sound hole facing in. It provides the treble and the clean acoustic string sound.

The pickup and microphone are mixed in a small two-input mixer provided as part of the system. The combination of the pickup and microphone provides a loud, punchy, yet natural sound with all the crispness of a real acoustic guitar.

It often helps to send the pickup signal just to the stage wedges (where feedback is worst), and send the mic signal just to the house speakers. Using as few mics as possible can also be helpful.

The more mics in use, the more likely you are to run into feedback. The gain-before-feedback ratio decreases 3 dB each time the number of open mics doubles. Two mics at equal levels have 3 dB less gain than one mic; four mics have 3 dB less gain than two mics, and so on.

To reduce the number of open mics, turn off any mics not in use at the moment. You might prefer to turn them down about 12 dB, rather than off, so you don’t miss cues. Instead of using 10 mics on a drum set, try using a single miniature omni mic in the center of the set. A mini mic is recommended because it has excellent high-frequency response in all directions ­ unlike a larger microphone.

Clip the mic to the right side of the snare drum rim, about 4 inches above the drum, and centered in the set. It will pick up the toms and cymbals all around it. You’ll be amazed how good that single mic can sound. Boost the bass to add fullness. If the cymbals are too weak, lower them a few inches. You can hang another mini mic in the kick drum, and it will sound full because omni condenser mics have deep bass response, no matter what their size.

A drawback of this system is that you can’t control the balance among the toms and snare except by mic placement. On electric guitar and bass, try using direct boxes instead of mics. Direct boxes pick up no feedback or leakage. You can plug the direct box into a connector following the musician’s effects boxes. This method, however, misses the distortion of the guitar amplifier, which is often an essential part of the sound.

Could a DI box be a better approach than what’s being done here?

Cancel At Distance
Finally, try noise-canceling mics. A noise-canceling (or differential) mic for vocals is designed to cancel sounds at a distance, such as instruments on stage or monitor loudspeakers. Such a mic provides outstanding gain-before-feedback, and almost total isolation.

The differential mic was designed to cancel sounds beyond a few inches away. As a result, many users have reported that their house mixes have improved because the mic’s isolation is nearly complete. In other words, “Mic 1” is no longer vocals and some drums, guitar and bass; “Mic 1” is vocals only.

Singers must use a differential mic with their lips touching the grille; otherwise, their voice gets canceled. This restriction is not a problem because many singers already kiss the mic. But it can be a drawback if the singer likes to work the mic for effect.

A cardioid differential mic also rejects sound behind the microphone, say, from a floor monitor. Not only does this prevent feedback, it also reduces the sonic coloration caused by monitor sound leaking into the vocal mic.

Give these techniques a try, and you’re likely to find improved results by using one or more of them.

Bruce Bartlett is a recording engineer, live sound engineer, audio journalist and microphone engineer (www.bartlettaudio.com). His latest books are Practical Recording Techniques 6th Edition and Recording Music On Location 2nd Edition.

Posted by Keith Clark on 01/26 at 08:50 AM
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Thursday, January 22, 2015

Church Sound: Five Ways To Improve Your Sound In 2015

This article is provided by ChurchTechArts.

It’s a new year, and now that we’re all rested up from Christmas, it’s time to start looking at how we can improve our systems—specifically audio—this year.

Certainly big-ticket items like new PAs, new consoles or new bands (just kidding) are nice, but sometimes we have to make incremental improvements.

Oddly enough, sometimes these small improvements add up to a big improvement that sometimes negate the need for a big spend.

Here’s a non-exhaustive list of five things you can do this year—without breaking the bank—that will improve your sound.

1) Test & Repair Loudspeaker Components
I once inherited a sound system that had two subs. One driver was completely blown, the other was torn. The main boxes had three bad HF drivers. As you might expect, the sound in that room was not good.

While it did take the better part of a day to diagnose the faulty drivers, and then another half day to replace them, once that was done, we actually had full-range sound again.

Testing your loudspeakers is relatively easy. If you have a bi- or tri-amped system, isolate each loudspeaker either by unplugging the amplifiers or the loudspeakers so that just one cabinet is running at a time. Then play some pink noise through the system. Get right up to the box and listen.

If you have access to an oscillator that can be swept from 60 Hz to 15 KHz, that’s even better. Just be careful with the levels; start low and work up to a comfortable level. If you find one box that produces next to nothing above 3K, you probably have a blown HF driver.

If you’re uncomfortable doing this or are unsure, contact a local dealer. This is a fairly simple process for them, and will likely lead to either a thumbs up or a list of new components to replace (and by components, I mean drivers, not an entirely new PA). Replacing the HF drivers in a system can have a great impact on the sound, and it’s not that expensive.

A test like this can have other benefits. I once was hired to mix in a room with a fairly complex PA layout. After struggling to get a good sound for a few months, I came in to test the drivers. I discovered the processor was wired incorrectly, sending the wrong signals to the wrong drivers. A quick re-patch made it sound like a new PA.

2) Get Your System Tuned
Once your loudspeakers are all producing full-range sound again, it’s a good time to have the system tuned.

A lot of people refer to this as “EQ’ing the room,” but it’s really not. We don’t EQ a room, we EQ a PA to work well in the room.

If you feel competent with using a measurement system, you can do this yourself. If not, hiring someone who is qualified shouldn’t be a huge expense.

Often, people who don’t really know what they are doing will try to “improve” on the sound of a PA by adjusting the system’s EQ. I’ve seen smiley faces, fish and other strange patterns on graphic EQs of systems I’ve worked on. None sounded good. Having someone come in to take measurements, set delays and EQ will often make a less than ideal PA sound decent again.

Once the PA is properly aligned and tuned, lock the processor or EQ either in software or by using vented security covers on the rack. Just remember to write down the passwords and put them somewhere safe—and where at least one other person knows where they are.

Sometimes, a simple tuning can extend the life of an old PA by a few more years. Often, the system was tuned years ago for one style of worship and the church has moved on. A re-tune can help optimize the system for the current sound you’re going after. It may not be a complete solution, and a new PA may still need to be in the long-term plans, but quite often spending a few hundred to a few thousand dollars on the tuning of the system will give you more time to save for the new one that is needed.

3) Upgrade Mic Package
Microphones are mechanical, and like all things mechanical, they can wear out. They’re also dropped and abused in other ways over time. If you’re using really old, beat up mic’s every week, changing them out is a cost-effective way to improve sound.

Sometimes it’s a matter of matching a mic to the source; a better fit for a vocal is a great example. Other times you may be using a mic on a source because you had it, not because it was the best choice. Finding the right kick drum mic for your drum kit, PA, room and sound can make a big difference.

Outfitting your stage with all-new mics might be cost prohibitive to do in a single year, but perhaps you can start down the road. Pick up a few new vocal mics that will help your singers sound better. Then move on to drum mics, and finally other instruments. Get recommendations from people you trust and try them first if possible.

4) Optimize Gain Structure
System gain structure is one of those things that we don’t talk about enough in audio.

I’ve seen all manner of sins in this area; consoles that are way overdriven with amps turned way down, and others with the amps all the way up and the faders all running at -40.

Optimizing your gain structure is critical to getting the best sound possible from your system.

Start with the source, and make sure your input channels are running at good levels with your faders around unity. Then move onto your mix buses (either groups or main L&R bus). The main output should be running somewhere close to where the green lights start to turn yellow (the exact, optimum point will vary from console to console, so this may take some experimentation).

You will hear it if your console is running too high or too low; it will either be noisy or distorted. Avoid both.

Next, move on to the system processor (or EQ) and the amps. You want healthy levels coming into and leaving the processor, then adjust the amps to achieve the level in the house that you want. If you have to turn the amps way, way down, you may want to drop the level coming out of the processor a little bit and leave the amps up.

Again, if you’re not quite sure how to do all of this, there is no shame in bringing in someone qualified who is. This is another area where big improvements can be made by making some small changes.

5) Practice
We typically expect that the worship leader, vocalists and musicians are practicing their parts throughout the week. But when does the sound guy or gal get to practice? Practice is the only real way to get better, so how do we do that? Unless you have a band that really enjoys playing for hours on end, the best answer is virtual sound check.

There are many systems available now that make it fairly easy to record each input on the board and play it back in place as if the band were still there. With a virtual sound check system, you can mix a song over and over, trying out new things, adjusting EQ, compression, FX and other techniques until you get it just right. And the only person you need in the room is you.

Or, try this one. How about recording the rehearsal, then coming in the next day with the worship leader and work on the mixes? Find out what he wants to hear, and work toward getting there. Sometimes, it will be clear that the problem is not a mix issue, but an arrangement one; in that case, everyone wins when the band gets better.

Virtual sound check might be the most expensive item on this list, but it’s still less than a new PA and will often have greater benefits. Go here for some help on how to get started.

As I said at the outset, this is not an exhaustive list, nor did I try to go into great detail on each topic. Do some research and find out how to implement these steps and you will have better sound at the end of the year than you do now. And you may even have budget left over!

Mike Sessler now works with Visioneering, where he helps churches improve their AVL systems, and encourages and trains the technical artists that run them. He has been involved in live production for over 25 years and is the author of the blog Church Tech Arts.

Posted by Keith Clark on 01/22 at 06:09 AM
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Tuesday, January 20, 2015

The Votes Are In And The Readers Have Spoken!

The votes have been tallied and the results are in for the Sixth Annual ProSoundWeb Readers Choice Product Awards.

The participation of both manufacturers as well as our audience proved exceptional, with thousands of ballots cast.

The races in each category were close and competitive, owing to the overall strength of every product entered combined with the distinct yet varied preferences of the pro audio industry’s largest online community. In fact, many races were so tight, separated by just a few votes, that multiple winners were awarded.

The Readers’ Choice Awards is unique for a number of reasons, chief among them (and as the name says), all voting is the exclusive domain of the readers of ProSoundWeb.

Our sincerest thanks to everyone who entered and voted in this highly successful sixth run of the PSW Readers Choice Awards.


Consoles & Mixers—Large Format
Soundcraft Vi3000
DiGiCo SD9 Rack Pack

Consoles & Mixers—Small Format
Yamaha Commercial QL Series

Line Arrays—Large Format
EAW Anya
Meyer Sound LYON

Line Arrays—Small Format
NEXO GEO M620 (Yamaha Commercial)
dB Technologies DVA-MINI M2M + M2S

Loudspeakers—Column and Line Source
Meyer Sound CAL

Loudspeakers—Drivers and Transducers
Eighteen Sound 18TLW3000
Acustica Beyma 10MC500

JBL Professional EON615
Electro-Voice ETX
Mackie SRM450v2

EAW QX Series

Loudspeakers—Stage Monitors
EAW MicroWedge MW12

Adamson Systems Energia E219
EAW Otto

Loudspeakers—Portable PA
Martin Audio DD12

Microphones—Condenser Type, Performance
Shure KSM9

Microphones—Wireless Systems
Shure QLX-D Digital

In-Ear Monitoring Systems
Sennheiser EW 300-2 IEM G3

Networking Technologies
Yamaha Commercial Dante-MY16
Audinate Dante Ultimo

Power Amplifiers
Powersoft X Series
Lab.gruppen PLM Series

Power Amplifiers—Control & Monitoring
Lab.gruppen Cafe

Signal Processors
Lake LM 44

Outboard Gear
Avalon VT-737SP
Waves MaxxBCL

System Engineer/Tech Tools
Rational Acoustics Smaart v.7 Di

Interconnect Products
Digiflex D-UX PowerCon Hybrid Cables

Live Recording Hardware & Software
Audinate Dante Virtual Soundcard

Audio-Technica ATH-M50x
Shure SRH1450

Posted by Keith Clark on 01/20 at 09:46 AM
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To Chase New Clients Or Tap Into Existing Ones?

This article is provided by Commercial Integrator

The 80/20 rule applies to many, many things in business. One is that 80 percent of your business comes from 20 percent of your clients — so what does this mean for your future business?

Just like your existing business, your future business (read: profits) will also come from a small percentage of clients. But are those future profits going to come from the clients you already have, or the ones you have yet to secure?

Having worked closely with dozens of AV system integrators big and small, I have consistently heard leadership teams call out the need for more business; more new clients, more revenue generation and more diversity in revenue sources.

And while all of these outcomes are ideal, what I don’t hear about nearly enough is more customer retention, more customer satisfaction and more business diversity with the clients already in house. Leaving me to ask two questions:

Why so little talk about the current clientele?

What are you thinking not doing more to grow the business you already have?

If I told you it was six times more expensive to acquire a new customer than it is to keep one you have, would you believe it? The research is in and it supports that data point. Turnover is very expensive!

Furthermore, for most small businesses, the best method of customer acquisition comes from the clients they already have. So when a client walks out the door, not only does the associated revenue vanish, so does the potential connections of a satisfied customer.

Time and again I hear about big accounts changing hands from one integrator to another. This can certainly be a byproduct of another integrator knocking their socks off, but if the experience is anything like the one I know so well, the changeover took place at an inflection point.

You know, one of those moments where the integrator of record dropped the ball and the disappointed client started shopping around. Now, at some point this will happen to every business, but as a whole, the reason this most often happens is because the relationship between the integrator and the client was never cemented.

It was merely too transactional.

For any and all growing businesses, it is important that business development is a focus. But developing business needs to be both of the following things. It should be something that is done in addition to taking care of the customers you have.

Invest in resources to serve your clients before finding people to identify new ones. It can come from the clients you have. Meaning, do you truly believe that the clients you already have are doing all of the business they can with you?

Ask yourself, how many of your current clients do you even consistently make time for when there isn’t an active project? If there aren’t regular meaningful touch points in between sales, then you are further perpetuating low-value sales relationships that will likely cost you down the line.

The more you secure the relationship between projects, the more likely some type of mistake can be overcome without customer attrition. In short, there is nothing wrong with aspiring to grow your business, but it should never come at the expense of the clients you already have.

If you’re seeing too many clients exiting on a year-over-year basis, maybe more sales isn’t the answer; maybe the answer is more customer retention.

Daniel L. Newman currently serves as CEO of EOS, a new company focused on offering cloud-based management solutions for IT and A/V integrators. He has spent his entire career in various integration industry roles. Most recently, Newman was CEO of United Visual where he led all day to day operations for the 60-plus-year-old integrator.

Go to Commercial Integrator for more content on A/V, installed and commercial systems.

Posted by Keith Clark on 01/20 at 07:03 AM
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