Tuesday, September 01, 2015

Carry-On Mixing

Yes, I know. When I first saw TouchMix at NAMM two years ago, I thought it was a toy, but while I was at Summer NAMM’s new TEC Tracks, across the aisle QSC was presenting “the world’s most powerful and portable easy-to-use compact digital mixer” with hourly live performances.

TouchMix is well into its second year and second software revision, so I knew it was stable. Greg Mackie and Peter Watts consulted with QSC to design this unique little mixer, so I knew it was well thought-out. What surprised me was TouchMix’s sound quality.

I’d already begun my summer tour with New Orleans’ most legendary doctor using a different monitor desk and a different set of wedges every day, with wildly varying travel – literally planes, trains and automobiles – so TouchMix seemed like a solution that could provide day-to-day consistency, in a package small enough to be hand carried.

There are two models: TouchMix-16 is bigger, with twice the inputs of the TouchMix-8. The TM16 also has six mono auxiliary sends on XLRs instead of the TM8’s four, and two stereo aux sends for hard-wired in-ear monitors on TRS instead of one. The TM16 has both a stereo cue and a stereo monitor on TRS, as well as an independent XLR talkback input, while the TM8 simply has a cue output.

Full Package
TouchMix puts pro desk features in a laptop form factor: four-band fully parametric EQ, variable high- and low-pass filters, compressors and gates on every channel, plus class-A mic preamps, pro-grade converters, output EQ, limiting and delay, as well as eight DCA and eight mute groups.


In addition to all those aux sends, there are four dedicated FX sends with multi-parameter digital effects optimized for live sound, emulating popular L-word (“Lush”) and Y-word (“Dense”) plate, room and hall presets, plus delay and micro-pitch shift.

Both models employ 32-bit floating point processing, with 44.1 or 48 kHz sampling. Specs include a S/N ratio of 95 dB, dynamic range of 105 dB and latency of 1.6 milliseconds, comparing favorably not only to budget digital consoles, but to many midrange professional touring products as well.

Each input XLR has its own analog gain “trim” pre-amp control – two rows of eight trim knobs below two rows of eight XLRs. There’s up to 45 dB of gain in the full clockwise position. I disliked them at first, as they can be bumped and are obviously non-recallable.

However, unlike most other consoles, TouchMix has no faders or encoders except for its single Master Encoder, making dedicated gain controls necessary. The seven Shure BETA 57 and 58 vocal mics sit nicely at “12 o’clock” while SM58s might be at “2 o’clock.”

The tour’s file-based front of house engineer Andy Loy uses a 32-channel input list that fits a wide variety of digital consoles, so it seemed like the TouchMix-16 might not be big enough. However, only 16 channels are needed in the wedges.

Obviously all seven vocal mics are needed. In addition to the trombone’s wireless BETA 98 and the effects pedals fed from its UHF-R receiver, we mic two guitar amps, use the XLR output from Roland Guerin’s Aguilar DB 751 bass head, a Nord keyboard’s JDI, and a Barcus-Berry CS-4000 piano pickup that’s used with a Countryman Type 10 DI. That makes 14.

Herlin Riley’s Mapex drum kit uses a total of 13 mics, an input list that would satisfy most festivals. However, the majority of bands I’ve worked with need no more than kick and hi-hat in their floor monitors. While some musicians ask for snare drum in their wedge, most get more than enough snare from the hi-hat mic, which is always needed (Figure 1).

Figure 1 (click to enlarge)

The Shure KSM137, like the KM184, is particularly smooth and accurate, and when placed a half-foot above the hi-hat cymbals, is shadowed from the snare drum enough to balance against it, but also gets enough toms to sound natural when combined with a dynamic kick drum mic, for a total of two more monitor inputs.

By varying the hi-hat mic’s height, high-pass filter and EQ, a variety of drum sounds can be supplied, from lots of hi-hat to an even blend of hat, snare and toms. Without the pounding of snare drum mics in monitor mixes, everyone can monitor at lower levels, helping conserve hearing, though it’s not for everyone. Check it out.

On The Road
We began our run on smaller stages, starting with a Philadelphia parking lot gig. Moving into New England we played the Bull Run in Shirley MA, the Flying Monkey in Plymouth NH, The Space in Westbury NY, the Westhampton Beach PAC, Infinity Hall in Hartford CT, the Blue Ocean Music Hall in Salisbury MA and Dartmouth College’s Spaulding Auditorium.

Each venue offered a new monitor console and a different make and model of floor monitor. The time needed to build a new file is not always available, unless you’re able to carry a wide variety of off-line editors.

Even so, moving from one monitor console to the next means switching from one show file to another, so the previous file on a similar console may be from a show that’s weeks or even months old, requiring substantial tweaking during sound check.

Console choices over four weeks in no particular order ranged from Yamaha M7CL, Soundcraft Vi6, Allen & Heath GL2200, Avid D-Show, Avid SC48 (twice), Midas PRO2, Soundcraft MH3, Yamaha PM5D (twice), Yamaha CL5, and Allen & Heath GLD. Even using Yamaha’s file conversion software, we still would have been on other consoles more than half the time.

And while it’s possible to completely chart a desk, that also takes time. The consistency of staying on the same desk or file, night after night, allows it to mature with daily input channel and mix refinements, allowing musicians to feel instantly comfortable - like they’re stepping onto the same stage over and over.

Figure 2 (click to enlarge)

When working with “wedges du jour,” a consistent set of mics and console are required for dependable results. Using TouchMix’s graphic EQ with Rational Acoustics Smaart DI to systematically flatten and contour the frequency response of each day’s new wedges is the beginning of a preset that can be copied to every mix. Mix EQ touch-ups are easily made with the iPad at each mix, compensating for double wedges or wedges hitting vocal mics from the side (Figure 2).

TouchMix provides a 28-band third-octave graphic equalizer on its six mono mixes and on the main stereo mix, with eight XLRs for easy interfacing with power amps or self-powered loudspeakers. While fully parametric EQ is considered best for tuning speakers, many agree that for wedges, a graphic EQ is better for balancing frequency response while also managing gain before feedback.

Third-octave ISO frequencies don’t often fall where they’re needed, however today’s manufactured floor monitors exhibit fairly flat frequency response so that their response can be managed with a GEQ. There are sometimes a few sharp peaks that fall between ISO frequencies, but can be tamed using TouchMix’s four notch filters (Figure 3).

Figure 3 (click to enlarge)

WiFi & App
TouchMix comes with a simple USB WiFi dongle that works in uncluttered 2.4 GHz environments and performed well in 500-seat venues during our first week. At the first larger venue, a greater number of smartphones in the audience overwhelmed the dongle, forcing operation from the TouchMix itself.

The following day a dual-band 2.4/5 GHz “a-n” router was installed, connecting to TouchMix on the low band, with it’s SSID hidden, and connecting to the iPad using its less congested high band, where it worked flawlessly in Manhattan, at the Newport Jazz Fest and in Europe. A USB-to-Ethernet adapter is an even more secure way to get into a WiFi router.

iPad apps are great for monitor mixing, allowing operators to stand eye-to-eye and ear-to-ear with performers on stage. The TouchMix app is my new favorite. it closely mimics TouchMix’s touch screen and controls. The main difference is that its Master Encoder jog-wheel is replaced by up and down 1 dB fader “nudge” controls that I enjoyed, good for both mixing IEMs where smaller moves are better and double or triple-tapping for wedges.

The iPad mini is slightly larger than TouchMix’s 7-inch screen and the app’s on-screen faders are longer than the mixer’s, though I prefer iPad’s even larger surface for my big hands.

The app and the mixer’s GUI operate independently, so the iPad can act as an extra user interface that shows and controls different functions from those controlled via the screen.

This can be extended to multiple iPads, so that one (or more) could mix monitors, while another might be used for the FOH mix even though TouchMix might be located at the side of the stage. It would even be possible to tile several tablets together to display the entire TouchMix console!

Substitute drummer Derrick Phillips, who plays for Hank Williams, Jr., brought his custom IEMs and it was easy to give him a hardwired in-ear mix that he could control from his iPhone from one of the TouchMix-16’s two stereo auxiliaries by just using a headphone extension cable. The operator can allow or restrict access to functions on a per-device basis, keeping users from adjusting anything but their own mix, allowing the iPhone app to operate as a personal monitor mixer.

For non-Apple users, QSC just released the Android version of the app for tablets and smartphones that requires Android OS 4.4.4 or newer and TouchMix firmware V2.1.4922.

The author mixing stage-side via the TouchMix app.

Size Matters
At 12 by 15 inches and 6 pounds, TouchMix-16 is no bigger and slightly lighter than my last Windows laptop. Its cardboard retail box holds TouchMix in a slim 13-inch by 22-inch fabric and hard foam zippered case with a side compartment for its international voltage power supply. Our trip to Europe simply required using an IEC cable with a Schuko plug instead of an Edison.

SKB makes the injection molded 3i1813-7-TMIX waterproof case for TouchMix, with a custom foam insert that accommodates either model by removing a perforated foam insert, plus a cutout for its power supply, holding a standard or mini iPad underneath. Not only does either case meet carry-on restrictions, they even fit overhead in smaller, three-across regional planes.

The latest firmware upgrade, version 2.1, includes options for Mandarin, French, German, Russian and Spanish in addition to English, as well as the ability to assign aux buses to the left and right main outputs to act as sub-groups.

TouchMix in its SKB compact waterproof case.

TouchMix-16 is also capable of direct recording all 20 (16 mic plus 2 stereo line) individual channels plus one of its three stereo buses to a 7200 RPM external USB hard drive in 32-bit broadcast wave format, and then play those tracks back for virtual sound check or virtual rehearsal.

If you’re looking for a mixer that can take your band from rehearsal to promotional appearance to support slot to festival stage and perhaps even headlining, TouchMix is the smallest professional solution you can carry with you in the van and on the plane.

Mark Frink is a long-time monitor engineer and professional audio editor and writer. He’s hosting the Live Sound Expo at the 139th AES Convention in New York this October.

Posted by Keith Clark on 09/01 at 12:38 PM
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Audio Visualization

PSW Top 20 presented by Renkus-Heinz

It is said that “a picture is worth a thousand words” and nowhere is this more applicable than when trying to teach complex concepts.

A graphical depiction can often convey an idea better, and quicker, than a whole bunch of words. This is because our brains are mainly image processors, not word processors; the part of our brain that processes words is actually very small in comparison to the part that processes visual information.

Therefore visual cues help us to better store and retrieve complex information.

Bearing this in mind I’ve been exploring various ways of representing key audio concepts and terminology visually.

This invariably involves a certain degree of simplification but I think the results are a useful weapon in the battle against incomprehension.

Let’s start by looking at a simple way to represent the frequency content of a single sound, such as a kick drum, shown here against a vertical axis denoting frequency (Figure 1).


On the left is a representation of a kick drum that has been miked in a standard way, with a single microphone poking through the hole in the front skin. Here we can see there is slightly more energy in the bottom and top of the sound (i.e., the thud and the click) than there is in the middle – quite common when close miking a kick drum.

The right side of the image represents the same signal after a little EQ has been applied, in this instance the bottom end has been enhanced while the lower and upper mid range frequencies have been reduced slightly to give that classic kick drum sound.

EQ isn’t the only way we affect the frequency content of sounds so let’s take a look at some other methods (Figure 2).


On the left is a representation of a snare drum that has been miked up in standard manner – a single mic above the top skin. In the middle is the same snare with a high-pass filter applied, as indicated by the fade (which denotes the gradual reduction in the lower frequency content). On the right is the same snare after compression has been applied. In this instance the compressor is limiting not just the dynamic range of the snare (which is difficult to depict in a static image) but also it’s frequency content, resulting in a tighter and punchier sound at the possible expense of some of the finer detail.

Now that we’ve established a simple way to visually represent the different sounds, and the ways in which we can affect them, let’s take a look at a full drum kit. The kit as a whole has the widest frequency range of just about any instrument (with the possible exception of the pipe organ), from the low thud of the kick drum to the fizzy sparkle of the cymbals.

It comprises multiple elements that all need to be miked up in a way that enables us to treat each individual sound in relative isolation such that when they are combined, they complement each and work together as a whole. If we take a standard four-piece drum kit, miked up in a standard way (i.e., a single mic on each drum with a pair of overheard mics), and just bring up all the faders, it might “look” something like Figure 3.


I’ve now added panning information to the horizontal axis to denote the positioning of the sounds within the stereo field (at the moment everything is panned centrally). The one thing that this depiction makes obvious is the clutter that occurs where the sounds overlap each other, particularly in the mid range where the kick, snare and toms all produce sound energy.

This is a common cause of “muddiness” in the drum mix – something which can quite easily be addressed with a little EQ and panning (Figure 4).


Here we can see that a bit of EQ has been used to bring the bottom end of the kick drum out, the kick has also been compressed and a low pass filter has been applied. The snare has also been compressed and a high pass filter has been used to tame the lower mid energy. The toms have been EQ’d and panned to create some space for the snare and mimic their physical placement in the kit. The cymbals (or overheads) have been high passed and panned wide to give the top of the kit a nice sense of width.

Overall you can quite clearly see that the processing, while quite subtle, has created space for each individual element of the kit so they can be clearly heard but also so they work together and complement each other. This avoids the muddiness that can so easily bog down the drum sound and gives the kit the clarity and definition that will help it to sound good, even in the busiest of mixes.

Speaking of which, let’s take a look at other common mix elements in the form of a simple three guitar set-up, i.e., bass, rhythm and lead guitars (Figure 5).


This depiction shows the potential for a messy sound in the lower mid range where all three instruments produce energy. We can also clearly see the masking that commonly occurs when the fundamental frequencies of the guitar overlap and obscure the harmonics of the bass. Guitars also tend to have a pronounced low end as a result of the use of directional microphones in close proximity to speaker cabinets which exacerbates the proximity effect.

Thankfully this can all easily be fixed with high-pass filters and a little panning (Figure 6).


First, the bass has been compressed, which helps tighten it up and enables it to rumble away at the bottom of the mix without jumping out or dipping down. The guitars have been high passed and panned, which not only creates more room for the bass but also helps them to come across much more clearly. This may result in the guitars sounding slightly thin when listened to in isolation, but when combined with the bass,  both instruments will come across much better while complimenting each other nicely.

Now that we have the tools to depict key elements and processing lets take a look at the mix as a whole.

Any musical performance that features more than one melodic or rhythmic element, be it live or recorded, requires these elements to be mixed.

Traditionally music was performed live and the mix was achieved by intelligently positioning the individual mix elements and augmenting where necessary. (You want the violins to be louder? Get more violins!)

In the early days of recording, where the performance was captured completely live with one or two microphones, a mix could be achieved by moving the musicians relative to the microphone(s), often during the performance to create dynamic variation.

In live performance we’ve evolved methods whereby each key element is miked or taken direct individually, so that we can treat them individually before combining them together in the mix.

So what happens if we combine our drum kit with the guitars and throw a vocal on top? (Figure 7)


It’s starting to look a little messy now but a mix is a complex interaction of multiple elements, so that’s quite normal. The vocals have been compressed to narrow their dynamic range and help them sit on top of the mix, but there’s still a risk of them being swamped by the other instruments that have energy in the same frequency range (such as guitars). Being as the vast majority of music we mix at gigs is song based, ensuring the vocals can be heard is a key challenge.

So what can we do to make them stand out a bit more? (Figure 8)


The answer, of course, is reverb, depicted here using an outer glow. They key is to use a reverb that’s small enough to ensure the vocals don’t sound too distant while exploiting the spectral smearing and stereo widening that can help to make the vocal sound bigger. A plate with a reverb time between 1.0 and 1.6 seconds and a pre delay below 10 ms usually does the job. I sometimes find that rolling off the top end (or adjusting the high ratio) of the reverb helps to make it sound more subtle – less like a digital reverb and more like a natural acoustic space around the vocal.

But that’s not the only way to make a vocal stand out in a busy mix (Figure 9).


Another trick I like to use is to apply a slap-back delay to the vocal, here depicted by the black drop shadow. A slap-back delay can be anything between 50 and 300 ms, but I find a setting of between 100 and 180 ms works particularly well on vocals in mid-tempo songs.

The feedback gain should be set so there is only one audible repeat – typically achieved by setting it to 10 percent. This creates a single echo very close to the vocal which has the effect of doubling it and helping it stand out against the background noise. Again rolling off the top end of the effect can help make it less obvious and more subtle.

As mix engineers we’re always striving to build better mixes, so hopefully these pictures have been worth a few thousand words and have provided a unique insight into the process. We might think we mix with our ears, but our brains are doing all of the hard work, so anything that can help us visualize such abstract concepts will enable us to better understand the nature of the mix and produce consistently high-quality results.

Andy Coules (andycoules.co.uk) is a sound engineer and audio educator who has toured the world with a diverse array of acts in a wide range of genres.

Posted by Keith Clark on 09/01 at 11:14 AM
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Oxygen College Outfits New Studio With Audient ASP8024 Console

The Geelong West music facility will use the analog console predominantly by the Advanced Diploma of Sound Production students.

Increasing numbers of students at the audio engineering branch of Oxygen College have resulted in the commission of a brand new, larger studio where a 36 channel Audient ASP8024 mixing console with Dual Layer Control (DLC) and patchbay now takes center stage.

Newly opened in May, this latest addition to the Geelong West music facility is in the care of head of audio Tom Isaac, who explains, “It is used predominantly by our Advanced Diploma of Sound Production students.”

He continues, “The students and staff are very excited and cannot wait to get in there to record. We have other analog studios here at the college but none as large as this.”

Keen to develop the students’ understanding of signal flow, the fact that the studio is analog is definitely advantageous, reckons Isaac. “The Audient desk fits in with our needs for a training platform which teaches students ‘out of the box’ mixing as well as being a great sounding desk.”

“In this very digital age it’s important that students still understand analog workflows, routing, gain structure, TT patchbays etc. It’s amazing to see their reactions when we say: ‘OK, lets mix this, but you cannot use ANY plug-ins. Only the Audient and the outboard in the rack. It really gets them thinking about their mixing craft and how records have been mixed in the past.

We have some great outboard units from TubeTech, Universal Audio, Manley, API, Drawmer, Lexicon and Eventide to work with giving students all they need to pull together a great sounding mix.’

“Flexible routing is one of the key strengths, having so many inputs available on mix down is the main reason the ASP8024 got the job. Our students are very excited about this feature.” Keeping the students’ minds open to analog has certainly had an interesting effect, as he explains: “Whilst they appreciate our studios based on a control surface, ProTools and ‘in the box’ mixing, a lot of students are very passionate about classic analog outboard and an analogue summing environment. They may not be able to pinpoint specifically what exactly they like better about the sound of an analogue summed project versus an ‘in the box’ summed project, but as their critical listening skills develop, it’s been a great conversation starter.”

As the largest studio at Oxygen College with a large live tracking room, plus vocal and isolation booths, it helps to teach the finer points of running a commercial facility: they have simulated budgets, worked out a cost per hour of hire and have strict deadlines for recordings. They also work in tandem with Advanced Diploma of Music students, who release EPs as part of their course.

“The Sound Production Diploma students record and mix these with guidance from their trainers. Some go on to be released and enjoy radio play,” says Isaac.

The Audient console’s patchbay is getting a thorough workout as well. “Students are very much looking forward to combining their favourite plug-ins with some high-end outboard dynamics that they can insert via the ASP8024’s patch-bay.”

Oxygen College has been a creative arts training hub for five years now, providing state-of-the-art, industry-standard and ‘real-world’ education, for a range of arts including music, photography, painting and drawing. The college’s commitment to a low student-to-facility ratio, ensures students have excellent access to studios, equipment and training staff.


Posted by House Editor on 09/01 at 09:25 AM

Friday, August 28, 2015

Shenandoah University’s Conservatory Steps Up To DiGiCo SD9

Ohrstrom-Bryant Theatre’s new desk selected for the combination of tech and customer care

When a new piece of gear is specified for a theatre install, a variety of technical issues and requirements naturally influence the purchase decision. However, sometimes the final deciding factor ultimately comes down to a manufacturer’s people.

That was very much the case when New Market, Maryland-based Acme Professional recently installed a DiGiCo SD9 at Shenandoah University’s Ohrstrom-Bryant Theatre in Winchester, Virginia.

While the university has a noteworthy traditional music and recording program and the venue is used for everything from concerts and recitals to dance performances, Ohrstrom-Bryant is really all about one thing—musical theatre.

The venue hosts numerous productions each year spanning both long-running chestnuts and productions of more recent Broadway hits.

The recently ended 2015 summer season included performances of A Funny Thing happened On the Way To the Forum and The Music Man, as well as the musical version of The Addams Family.

Acme Professional’s Pete Cosmos reports that he initially specified another manufacturer’s console for Ohrstrom-Bryant, but that changed after members of the teaching staff attended last fall’s AES Convention in Los Angeles.

Golder O’Neill heads up the school’s recording program and, as the resident “gear guy” along with colleague Adam Olson, was tasked with checking out consoles at the audio trade show with an eye on replacing the analog desk that had been installed when the theatre opened nearly 20 years earlier.

“We got there pretty late,” O’Neill recalls. “Most of the people on the show floor were already packing up their booths and we had a hard time getting anyone to talk to us.” Until they got to the DiGiCo booth, that is, and US sales manager Matt Larson stopped the load-out activities to power up the SD9 again and demo it to the late arrivals.

“Matt showed us a number of things that were going to help us,” he continues. “Even something as simple as the ability to copy the settings and routing of one channel strip and copy it to another has saved time and just made for a better audio experience. But the most convincing argument was not about features. It was really educational.”

Like many school theatre programs, the audio staff running shows at Ohrstrom-Bryant primarily consists of students. “These students are learning how to use technology that they will be expected to know how to use when they leave school and get out into the ‘real world’. Matt, quite rightly, pointed out that a majority of shows both on Broadway and in other musical theatre centers around the world are mixed on DiGiCo consoles.”

Shenandoah has its roots as a conservatory of music; musical training and the conservatory came first and the designation as a university came later. The school has always been about training students to make a living in music. Those roots made the decision to go with the DiGiCo SD9 an easy one once it was understood that this would be the same tool that students would come to use as working professionals.

The venue’s heavy schedule—especially in the summer season—also made the move from analog to digital crucial.

“The summer musical theater season is the busiest and most important period of the year for us; we do four productions in an eight-week period,” O’Neill reports. Each show is in tech rehearsals for a week before performances begin. And with the short period allowed for the season, there is no break. The next show in the schedule starts tech rehearsals during the second week of performances for the show before it.

“Moving to the SD9 has been about more than just feature sets or overall audio quality, although those are great,” O’Neill says. “But the ability to be developing one show while all of the settings for the current show are saved and recallable all on the same console has made things much easier and smoother.”

As with any educational institution, what the staff wants can be secondary to other concerns—especially financial ones. And even after O’Neill knew what he wanted, he needed to sell that idea to the administration.

“Technology is changing so fast,” he says. “And if we are going to really educate the students in the theatre audio program and prepare them for being able to work after they leave school, then we have to have state-of-the art tools for them to learn on and, even more importantly, that they can get actual mixing time on. That was the argument that sold the administration.

“For me, there was no real light-bulb moment—no specific feature that helped me know that the SD9 was right for us. But Matt’s personal approach really clinched the deal. If he was willing to stop all of the packing activity at their booth to talk to us and ask us about what we do and what is important to us—and then thoroughly demo the SD9 while everyone else was leaving—then I knew that this was a company we could count on to stand behind its product.”

Shenandoah University

Posted by House Editor on 08/28 at 02:36 PM
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Creative Console Strategies

PSW Top 20 presented by Renkus-Heinz

The first “big” mixing consoles I owned were a 12-channel Kelsey and a 16-channel Yamaha PM1000. The Kelsey saw the most use because the PM1000 weighed in at 110 pounds, and that was without the wooden case I built for it.

With a limited number of channels, buses and features available, I learned to be quite frugal when deciding what to mike onstage. For larger shows, the Kelsey sometimes served as a sub mixer for the drums and bass, feeding the Yamaha.

One day a buddy asked me to mix on his rig at a large outdoor jazz festival. It sported a 32-channel PM1000, and I was in heaven for two reasons. First, he didn’t ask me to help lift or move it, and second, I didn’t have to pick and choose what to put in the PA. With 32 inputs I could mike up everything onstage and still have empty channels.

Lacking, however, were monitor buses, but it was a problem easily solved back then by routing inputs to one of the four mix matrix buses and using those to feed stage wedges. Not as ideal as having individual aux sends on every channel, but musicians were aware of technology limitations and were happy to get more than one mix in those days.

Fast forward to today. One of my small digital consoles offers 66 processing channels and up to 14 mono buses in a rack-mount form factor. With onboard GEQs, FX units, comps and gates, there’s no need to carry outboard gear, and it can be picked up effortlessly.

But as full-featured as these smaller boards are, bigger is often still better because clients always seem to need another feed or send somewhere, and there’s almost always extra inputs that show up at the last minute.

An 8-channel version of the author’s 12-channel Kelsey console, circa the mid-1970s.

Regardless of the size of the console, sometimes we have to be a little “creative” to get the desired results. Here are some things I do.

Doubling Up
When running both front of house and monitors from the same console, it means that the monitors either share the same channel EQ dialed in for the mains (post EQ sends) or they do not get any EQ at all (pre EQ sends). This might not cut it for a picky performer or an acoustic instrument.

What I do is use a simple splitter box to send the microphone or DI to two channels instead of one. The first channel is for the house mix and the second (usually adjacent to the first) can be “dialed in” with an acceptable EQ for the monitors.

On smaller acoustic shows, I might place every input into two channels, effectively providing separate house and monitor consoles. If there aren’t enough splitter boxes handy, I can use a channel’s direct output to feed the second channel. On a digital console that offers channel patching, simply patch an input to more than one channel in the menu.

I’ve also used a second channel for singers who want a significant amount of effects in their monitors but don’t want to hear the effects when they’re talking in between songs. Sure, I could mute the FX masters, but on most of my consoles they’re on a different layer by default. Using the second channel for effects to the monitors, I can simply press the convenient mute button to stop the effects as needed.

It’s also easier to dial in a good mix of “dry” verses “wet” vocals in the monitors because I can simply send dry effects to the monitors from one channel, and then wet it up as needed with the second channel.

Another use for second channels is making a killer board recording. Many of us make recordings of live shows, and there are a lot of ways to do it. “Down and dirty” board tapes can be had by taking a copy of the main L+R outputs and sending them to a stereo recording device.

Newer digital consoles may offer the option of recording a stereo feed to a USB drive, but the mix and some instruments may not sound “right” because they were equalized and balanced to be heard through the PA rather than a recording.

Multi-track recordings can be attained by sending the channel direct outputs to a recorder or splitting off the inputs with a splitter snake or grabbing the inputs off a digital network – but this involves using a stand-alone multi-track recorder and possibly a lot more extra gear.

Sometimes all that’s needed is a good stereo board tape. Sure, you can set up a mix using an aux send, but this raises the problem of sharing the EQ with the house PA. Using second channels on instruments or vocals that have been “overly adjusted” to sound good in the house can result in a better recording because you can have control over difficult stage sounds as well as EQ directly for the recording.

Matrix Mixing
For years I carried around a line-level distribution amplifier in my rack because I was always running out of outputs on the corporate-type shows that make up the majority of my work. I might only need a few inputs but dozens of outputs for the main loudspeakers, delay and fill loudspeakers, as well as feeds to the venue for underbalcony fills, lobby systems, overflow rooms, onstage and backstage monitors, video and safety recordings, intercoms and dressing rooms, etc.

This is why I gravitate to consoles that have extensive matrix sections. In its most simple form, a matrix takes a selection of inputs (usually derived from the group and main output buses) and allows routing of those signals, complete with level control, to a series of outputs. Complex matrix systems offer the ability to choose from a variety of inputs, including specific channels or external sources, and may supply processing that includes EQ, compression, limiting and even signal delay. 

Even when you don’t have this many channels and options, a matrix section can considerably expand flexibility.

A matrix adds a ton of flexibility to a console and gives the user a lot of easy solutions to routing problems, like adding a support act console.

While there are many ways to tie two or more consoles into a single PA, more than a few times I’ve simply patched the support act console into the external matrix input on the main console and fed the PA from both consoles through the matrix out.

Another good use for a matrix is creating mix-minus feeds. This refers to a program feed that has been remixed to exclude one or more input components. Sometimes the video people might want the program audio minus the playback audio they’re sending to front of house, or teleprompter operators want to hear the program feed but with less music. I can whip up a quick mix-minus by routing the various parts of the program through subgroups and into the matrix. Levels of each feed can then simply be adjusted as needed. 

A trend I’ve been seeing of late is providing audio feeds for remote meetings. I need the audio from the remote site in the PA, but don’t need to send it back to them. so I’ll create a mix minus the remote audio by using the matrix, and then add processing like leveling and compression before sending to the remote site.

More than a few times I’ve been mixing monitors from a smaller front of house board and have run out of aux sends. Using a matrix, I’ve set up side fill mixes as well as individual performer mixes.

Additional Pursuits
One great feature about larger consoles is, of course, more channels to use. I can employ back-up mics or run back-up lines without having to re-patch.

Ever wonder why there are two mics on the podium at high-level events? One is normally not on, used as a spare that’s already in place in case there’s a problem with the main podium mic. Simply un-muting the spare keeps the show rolling. It’s the same reason we place two lavalier mics on important presenters at corporate shows or on lead actors in theatrical performances. 

Note: sometimes two different pattern podium mics are used, like a cardioid and supercardioid, with the mix engineer choosing between the two, depending on the person speaking.

Larger channel counts allow me to do some things normally not pursued when I’m running out of inputs. For example, instead of choosing between two overheads, or a single overhead and a ride cymbal mic, I’ll probably use two overheads and a ride cymbal mic on the kit. Same with snare drum. With enough channels, I often opt for a bottom snare mic to pick up the snap, capturing a better overall drum sound.

Even smaller-format consoles like this Yamaha QL1 afford capabilities not imagined a relatively short time ago.

Extra channels can also be turned into an ad hoc intercom system. I place a mic at front of house, plug it into a spare channel and send it to a powered loudspeaker placed backstage via an aux send. A mic placed backstage is routed to a second spare channel, and by pressing that channel’s PFL button, I can hear the person backstage on my headphones. Not perfect, but when the intercom power supply loses its magic smoke 15 minutes before a cue-heavy show, you do what you have to do.

One more use for extra channels is “phantom mixing.” Ever get to the point where you’re satisfied with the mix and then a person walks up to FOH and tells you that they can’t hear their girlfriend, boyfriend, wife, child, niece, etc.? A quick twist of an unused channel knob and a sincere “Is that better?” usually gets them out of your hair…

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

Posted by Keith Clark on 08/28 at 10:56 AM
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Allen & Heath iLive Out With The Summerland Tour

This year’s tour comprises Everclear, Toadies, Fuel, and American Hi-Fi, travelling across North America to over 30 different venues.

The Summerland Tour, previously hailed by Rolling Stone magazine as one of the “10 Hottest Summer Package Tours”, returned this year with Allen & Heath iLive digital mixing systems managing front of house and monitor positions.

Now in its fourth year, the annual Summerland tour is devised by Art Alexakis, front man of 90s alternative rock band, Everclear. This year’s tour comprises Everclear, Toadies, Fuel, and American Hi-Fi, travelling across North America to over 30 different venues.

Alexakis gives credit to the iLive systems – comprising iLive-T112 Control Surfaces and iDR-32 MixRacks - for playing a huge role in the success.

“We have used Allen & Heath for the last couple of years and the sound has just been phenomenal. I am so glad to be able to use them on every tour,” says Alexakis.

According to Derek Steinman, production manager/front of house of Everclear/Summerland Tour, iLive provides superiority in sound that surpasses the competition.

“We carry two Allen & Heath iLive T-112 surfaces and two iDR-32 mix racks for front of house and monitors. The board sells itself, and I can just hit the ground running. This console allows me to be connected to the band and get a focused mix at the same time,” said Steinman.

John Riley, monitor engineer/stage manager, strongly believes Allen & Heath created the best console on the market.

“We only wanted this brand. It’s simple and I personally favour the side chain filters and side chain compressions. I recommend Allen & Heath to all my engineering friends.” says Riley.

Allen & Heath

Posted by House Editor on 08/28 at 09:02 AM
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Thursday, August 27, 2015

All Mobile Video Adds Studer Vista X For New Remote Truck

AMV equips 10th mobile broadcast truck with new digital console for high-profile events.

All Mobile Video (AMV) provides end-to-end video and audio solutions for broadcast, entertainment, programming and events by adopting the latest technologies for its Manhattan and Los Angeles studios and nationwide fleet of mobile production trucks.

In outfitting its new, yet-to-be-named remote truck, AMV’s previous positive experience with Harman’s Studer led them to Studer’s flagship Vista X large-scale mixing console for broadcast applications.

The new remote truck will be operational by the end of 2015 and will service the type of high-profile entertainment programs AMV excels in producing—their previous credits include the MTV Video Music Awards, the Tournament of Roses parade, the CMT Music Awards, the NCAA Final Four and a “who’s who” of corporate clients.

The Vista X features Studer’s latest Infinity Processing Engine DSP core, which uses CPU-based processors to provide control of 800 or more audio channels and 12 A-Link high-capacity fiber digital audio interfaces to provide more than 5,000 inputs and outputs.

The Vista X’s many features include the company’s patented Vistonics color touchscreen interface, VistaMix automated mic mixing, Quad Star technology that uses four redundant processors to achieve aviation-standard levels of redundancy, Dante connectivity and much more.

Ian Vysick, audio development specialist at AMV, says, “We’ve had a very close relationship with Studer for a long time and their consoles have always performed as promised. From a sonic standpoint, I have yet to find a better-sounding digital console. They’re responsive and easy to use, our operators love them and in fact the people using the consoles ask for them specifically.” On many occasions customers have found and hired AMV because they learned that the company uses Studer consoles.

“This console has everything,” Vysick stated. The Vista X is AMV’s 10th Studer console, and Vysick notes that the company’s mixing engineers are very comfortable with Studer’s onboard dynamics processing and EQ and the way the consoles operate in general, making the step up to the Vista X a logical move. He also points out that because of the expanded internal processing the amount of onboard gear needed gets smaller every time AMV buys a new console – a distinct advantage in the cramped confines of a mobile truck.

“Because the system has the two Lexicon PCM96 reverb and effects processors integrated into the console, we only need a few pieces of outboard gear to complete what we believe is a really nice package.”

The new truck will provide connectivity for projects of all size and scale. “There’s a ton of MADI in this truck – 11 MADI cards with redundant streams to the video router and to the outside world, connection to multiple ProTools systems, and also a MADI card going straight into the RTS intercom and more,” Vysick noted.

For Vysick, another major advantage of the Vista X is its Dante connectivity. “We’ve integrated Dante into the vehicle alongside the MADI system, so the vehicle is already ahead of the game and will be ready for all future developments.”

All of AMV’s remote trucks are outfitted with Studer Remote Stagebox units and the new vehicle will be equipped with three Remote Stageboxes, all of which will be linked to the Vista X console. “Fiber is now so reliable that the fears in using it have gone away,” Vysick pointed out. “Now we can have everything in house and it’s the cleanest signal possible to the vehicle.”

One of the Stageboxes is equipped with a Dante card. “An advantage of this is I’ll be able to work alongside other vendors in adopting Dante,” Vysick added. “I can talk to PA vendors and intercom vendors and let them know that now that I have a Dante stream inside the venue, they can take my Dante setup and use it as a further tool to integrate the vehicle into the overall system.”

“With the incorporation of Dante, this truck with the Studer Vista X is state-of-the-art and will be ahead the game for years to come,” Vysick concluded.


Posted by House Editor on 08/27 at 11:33 AM

Sound Devices To Introduce New Products At IBC 2015

Show will feature the new 688 12-track portable mixer/recorder with the new SL-6 powering and wireless system.

Sound Devices will feature its 688 12-track portable mixer/recorder with the new SL-6 powering and wireless system at IBC 2015 (Hall 8, Stand B59).

By pairing these products together, users can streamline and de-clutter any audio bag, creating the ability to control and monitor an entire audio system in one wireless location.

Launched earlier this year, the 688 is the newest member of the 6-Series family of mixer/recorders and incorporates a multi-channel mixer, recorder, and MixAssist auto-mixing technology.

To simplify interconnection between the 688 and multiple channels of wireless, the new SL-6 powering and wireless system was developed using Sound Devices’ SuperSlot technology.

SuperSlot is a non-proprietary, open wireless control and interfacing standard. Combined with SL-6 compatible dual-channel wireless receivers, production sound mixers now have a product option that offers cable-free powering, fully integrated receiver control and monitoring, audio interconnection, and antenna distribution.

“We are thrilled to bring the most powerful portable mixer/recorder available in the 688, together with our innovative SL-6 accessory to IBC,” says Matt Anderson, CEO of Sound Devices.

“A mixing bag comprised of a 688 with SL-6 plus a SuperSlot-compatible wireless receiver offers audio mixing, recording and wireless receiver control all from the mixer, ultimately simplifying power distribution and interconnection. IBC attendees can stop by our stand during the show to get a glimpse into our next-generation audio recording and mixing capabilities.”

Posted by House Editor on 08/27 at 11:23 AM

Wednesday, August 26, 2015

Saint Saëns Records Organ Symphony With Mackie DL32R (Video)

Recording and live sound engineer Dan Kury finds a portable, digital recording solution with impressive results.

Recording and live sound engineer Dan Kury refers to himself as an old school engineer, having logged more than 40 years mixing and recording a wide variety of musical genres in all sorts of venues while hauling around big consoles, outboard racks and huge audio snakes.

When he discovered the Mackie DL1608 16-channel digital mixer, he was a happy man.

“The DL1608 is the first piece of equipment that changed my life,” Kury begins.

“The equipment, the labor, the time setting up and tearing down, the lids for the cases stacked under the table-all eliminated or greatly reduced.” Besides, he says, “You don’t need a console when you can grab an iPad and go up to the balcony or mix from anywhere in the venue.”

Still, for some gigs, he wanted more than 16 input channels. So when Mackie released the DL32R 32-channel rack-mount digital mixer, Kury jumped on it.

“Now I have a DL32R in a four-space rack, along with two wireless systems,” he explains. “It sits on the side of the stage, so no need for a big, heavy snake. Setup time and teardown time is half of what it used to be with a traditional console. Not only that, now I have DCAs. What an incredible luxury.”

Kury is also a big fan of Mackie’s Master Fader control app. “It’s easy to get around,” he explains. “The DL32R does a lot of things, so you’d think the app would be complicated, but it’s so well-designed and efficient to use.”

Although he was confident that the DL32R preamps would compare well with the high-priced preamp/interface he typically uses for recording shows, Kury wanted to be absolutely sure. His opportunity to compare the preamps came when he mixed the Metropolitan Orchestra of St. Louis performance of Saint Saëns’ Symphony No 3 at the First Presbyterian Church of Kirkwood, Missouri.

Saint Saëns’ Symphony No 3 is known as the “Organ Symphony,” and First Presbyterian Church of Kirkwood has a beautiful new Casavant Frerespipe organ, designed in the style of the great Parisian symphonic organs of the late 19th century.

“It’s a magnificent instrument,” says Kury. “If you play the lowest C with the pedal, and you pull the 32-foot stop, the 32-foot pipe produces a root pitch of 16 Hz.” This recording would indeed be a great preamp test.

“For my comparison test, I made two separate recordings,” Kury recalls. “I sent the mics to a splitter, and I recorded one feed through my high-end interface and preamps and the other feed through the DL32R.” Kury then sent the signals via FireWire to a Mac computer running a DAW.

The results were as Kury predicted. “The Mackie preamps absolutely held their own against the high-end preamps,” he says. “In a blind test, I could not tell the difference. There is no question that you can use a Mackie iPad-controlled mixer to do a multi-track symphonic recording. If I do more multi-track symphonic recordings, I will happily use the DL32R preamps.”

“The DL32R is absolutely remarkable,” he concludes. “It blew my mind.”

Dan Kury

Posted by House Editor on 08/26 at 08:09 AM

Tuesday, August 25, 2015

BSS Audio Soundweb Contrio Wall Controller For The SoHo Event Space

Aurora Sound & Light Design selects EC-4BV Ethernet wall controller and a BLU-50 signal processor for the Manhattan venue.

The SoHo is one of Manhattan’s newest upscale event spaces, an authentic penthouse SoHo loft that provides a full range of catering, DJ and other services for all types of affairs.

Celebrating its grand opening last July, The SoHo features an in-house music system operated by a BSS Audio Soundweb Contrio wall controller that makes music selection easy for The SoHo’s staff.

Ian Hoffer, president of New Jersey systems integrator Aurora Sound & Light Design chose the Contrio EC-4BV Ethernet wall controller and a BSS Audio BLU-50 signal processor with BLU link connectivity to serve as the operational heart of The SoHo’s music system.

“We needed a controller that would be absolutely simple to use per the owner’s wishes and discreetly blend into the space’s beautifully appointed décor,” said Hoffer. The Contrio EC-4BV was perfect for the application.”

The BSS Audio Soundweb Contrio EC-4BV features a push/rotary encoder, an encoder ring, four buttons and a color LCD screen that can display custom text and graphics. Available in black or white, the EC-4BV connects to a system via Ethernet and can be custom-configured using the Harman HiQnet Audio Architect system configuration and control software.

Music for The SoHo events is furnished via an iPad or a visiting DJ, and a mic is also provided. “At first the owner was going to go to a music store and buy a mixer, but since the space is rented to all kinds of people for all kinds of events, I advised him that a mixer would be impractical. People needed something that would be totally simple to use, and the BSS Contrio turned out to be ideal.”

Hoffer configured the EC-4BV to be able to switch between an iPad, DJ and mic and also mix the inputs together if required. A fourth button is programmed to select the mic only – a user only has to tap the “Mic Solo” button on the controller. The Contrio is connected via an Ethernet cable to a PoE (Power over Ethernet) power injector, which feeds into a switch and the BSS Audio BLU-50.

“This was the first time I have ever installed a BSS Audio Contrio product and I have to say I really like the fact that although it’s simple to use, the LCD screen and configurable buttons enable it to be customized to a high degree. A cool feature is the illuminated ring around the volume knob that acts as an input meter so you can see what your signal is doing.”

BSS Audio
Harman Professional

Posted by House Editor on 08/25 at 02:53 PM
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Church Sound: Two Big Tricks To Mixing Monitors

This article is provided by DerekSoundGuy.com.

I hate to admit this, but the thing I dislike most about audio production is mixing monitors. There, I’ve said it.

Sorry to all the musicians out there. In truth, musicians need to learn what they need to hear as that’s where their frustration lies.

It starts with a lack of focus. When the musician can’t hear something, they ask for more of it. Then the monitor mix gets louder. We’ve gone through changes that have taken the pain out of mixing monitors and created mixes the musicians liked.

As practice progressed from our Tuesday night rehearsal through to the Sunday morning run-through, the monitor mixes always got louder. To be fair, I’m not blaming the musician here, or the sound person, it’s just what has happened.

But what happens when seven monitors are on stage, feeding back into mics, and also bleeding into the seats because of their immense volume? It affects the mix, and what happens when the mix gets affected? It’s a downward spiral from there.

My worship pastor and I had a discussion about this and subsequently talked with the band. We needed to reign things back in and to do it, we had to educate musicians and vocalists on what is actually needed in the monitor mixes. 

The band was informed monitors aren’t meant to provide a nice, clean full band mix. If that was the case, they’d get the main mix and everything would be good. But no, the monitor is there for them to hear what they need to hear, in order to play as part of the band. So what does this mean for everyone?

First, it means swallowing a little bit of pride, and second, it means thinking critically for what is really needed.

The musicians were asked to consider what’s in their monitor mix and what really needs to be there. For instance, not all four singers need to be in everyone’s monitor. Just the person leading the song needs to be in any of the musician’s monitors. Everyone wants to hear the beautiful harmonies of the backing vocalists, but other than the vocalists themselves, it isn’t needed.

The bass player needs to hear the kick drum and the whole acoustic drum kit as well, but on most stages with live floor monitors, the drums will come through without any additional monitor needs. 

The major difference between live monitors and in-ear monitors (IEMs) and drum mixing is that with IEMs, more of the drums need to be in monitors for the sake of keeping time. Every mic on the drum kit does not need to be there though. I put the kick and overheads in the IEMs, sometimes the snare; it depends how well it is coming through the overhead channels. But the tom mics don’t need to be there as the overheads will pick those up.

The keyboards are important for the vocalists, often for finding the note and staying in key for a song.  The acoustic guitar can also help with this and keep track of the feel of the song.

What about the electric guitar and the bass guitar? Again, this depends on the song, but they aren’t really needed much in most monitors other than the electric guitarist and the bassist. And if you have live amps on stage (which we do) then you might not need them at all.

Basically, everyone doesn’t need to be heard in everyone’s monitor mix, whether live wedges or IEMs. We don’t want to deafen our musicians, and we want to keep the sound as clean as possible for the congregation on Sunday morning.

The next step was training sound operators and musicians what to do when something can’t be heard.The problem isn’t necessarily solved by turning something up louder, it might be that something else needs to come down. If the guitar players say they can’t hear the lead vocalist, maybe instead of boosting the lead vocalist, try turning down the other instruments in their monitors. If that doesn’t work, consider boosting the vocal.

Using the above processes, we significantly lowered our stage volume while it was still loud enough that all of the musicians and vocalists could hear themselves just fine. All it took was some communication and a little education.

During the next practice, musicians said with the monitors being quieter that they really noticed the room more. The vocalists felt more comfortable as the sound wasn’t blasting their ears, but was loud enough to hear themselves, and it showed in their comfort level onstage. 

Comfortable musicians play and sing better than if they’re worrying about their monitor mix. And those comfortable musicians along with a quieter stage means a clearer mix for the congregation.

Derek Sexsmith is the director of technical services at Heritage Park Alliance Church in Windsor, Ontario, Canada. His blog, dereksoundguy.com, chronicles his experience working on the technical aspects of a church. Also follow him on twitter @dereksoundguy.


Posted by Keith Clark on 08/25 at 08:32 AM
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Thursday, August 20, 2015

Masque Sound Supports The Williamstown Theatre Festival

Custom audio package and services helps Massachusetts festival celebrate 61st season with support from Masque for 12 years in a row.

When some of the theatre world’s biggest and brightest stars descended upon Northwest Massachusetts for the 61st season of the Williamstown Theatre Festival, Masque Sound once again returned to the venue with a custom audio equipment package to support sound supervisor Ben Truppin-Brown’s vision.

This marks the 12th year that Masque Sound has been providing audio services and support to the festival.

Since 1955, the Williamstown Theatre Festival has brought actors, directors, designers and playwrights to the Berkshires, engaging an audience of both residents and summer visitors.

Each season is designed to present unique opportunities for artists and audiences alike, revisiting classic plays with innovative productions, developing and nurturing bold new plays and musicals, and offering a rich array of cultural events including Free Theatre, Late-Night Cabarets, readings, workshops and educational programs.

The Williamstown Theatre Festival’s 2015 season schedule brings a number of famous faces to the Berkshire stage with stars such as Kyra Sedgwick, Cynthia Nixon, Audra McDonald and Will Swenson appearing in a variety of plays on the Main and Nikos Stages. 

Truppin-Brown’s goal in designing the sound for the 2015 season was twofold. He wanted to create a simple, comprehensive repertory sound system that addressed the basic needs of each space, while maintaining the flexibility to accommodate the specific needs of each individual production. It was also important that the systems be different enough so that the staff and interns are exposed to a wide range of equipment to cater to the varying experience-level of the team.

One of the biggest challenges Truppin-Brown faced in designing the sound for Williamstown was the short 24-hour turnover time between productions.

“The schedule at Williamstown is insanely variable, and terribly complex to navigate,” says Truppin-Brown. “It changes quickly and often, and we have to remain flexible enough to accommodate the various, sudden shifts throughout the day. The work we do with Masque Sound, both before the season begins and during the summer, allows us to respond quickly and efficiently to those changes as they occur.”

This summer, the Main Stage’s system, provided by Masque Sound, was comprised of a Yamaha CL1 console, a Yamaha DME64N for signal processing and for a delay matrix, a Yamaha Rio 3224-D for additional I/O, and a Meyer Sound ULTRA Series main PA. Meyer Sound UPQ-1P’s drive the bulk of the system, with a mix of UPJ-1P and UPM-2P fills.

The low frequency is provided by a pair of 600-HP’s on the ground and a pair of UMS-1p’s in the air. Under-balcony delays are EAW JF60’s. Several DPA 4021s are used in various configurations for area reinforcement and an 8-channel Sennheiser EM 1046-RX diversity receiver is used for various wireless needs, with SK 5012 body pack transmitters and several SKM 300 handheld transmitters.

Signal distribution is all handled over a redundant Dante network. This allowed Truppin-Brown to phase out a good amount of cable that was so heavily relied upon in the past. A spare stock of Meyer Sound and EAW speakers complements this system for show-specific needs.

The Nikos Stage system includes a Yamaha CL5 console, a Rio 3224-D for the main patch, two Meyer Sound Galileo 616 processors, four d&b audiotechnik D12 amplifiers, four d&b D6 amplifiers, several d&b E-PAC amplifiers, and several Lab Grüppen FP2400Q amplifiers. The main PA is made up of d&b audiotechnik E12s left and right with E8s on center, and a pair of B2-subs underneath the deck. Surrounds and rears are L-Acoustics MTD108a’s.

Signal is once again distributed over Dante, making additions simple and efficient. This summer, Truppin-Brown was also able to incorporate d&b Audiotechnik’s R1 amplifier control software into the Nikos system. It is an extremely powerful tool, allowing convenient and customizable access to the DSP built into the stages’ d&b amplifiers.

For Unknown Soldier, the newest musical on the Nikos stage, Masque Sound provided 20 channels of wireless via Sennheiser EM3532 with SK 5012 wireless transmitters, a large and comprehensive microphone package, two additional Rio 3224-D’s in remote locations added to the existing Dante network, and a new system of loudspeakers.

Leon Rothenberg, the sound designer, was happy to take advantage of the existing d&b audiotechnik amplification in the room. He specified d&b Q10’s as mains, a compact, three-box d&b Q1 array at center, six d&b E0 front fills, four d&b E0 box fills, two d&b E3 wide delays, and a single d&b T10 center delay. Onstage foldback was handled by d&b E3’s and E0’s, and upstage effects were driven by d&b E8’s.

“From the first, casual conversation with Scott Kalata at Masque Sound about our system needs, to the daily interactions with John Sullivan (our sales rep) and Tara Perez (our show key) in the Masque Sound shop, to the constant support received throughout the summer, Masque Sound has been extremely supportive and infinitely accessible,” adds Truppin-Brown.

“On location Stephen Dee, the Main Stage associate sound supervisor and Adrianna Brannon, the Nikos associate sound supervisor, have been absolutely invaluable, and have been working with me since March to design and implement these systems. The equipment at this year’s festival was selected with very specific goals in mind, and working with everyone there to accomplish those goals was a pleasure. Masque Sound is endlessly accommodating and the quality of support we receive is unparalleled.”

Masque Sound

Posted by House Editor on 08/20 at 07:38 AM
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Wednesday, August 19, 2015

Sound Devices Supports HBO’s Show Me A Hero

Production sound mixer Frank Stettner selects the Sound Devices 970, the company’s first-ever audio-only rack-mounted solution.

When production sound mixer Frank Stettner began wrapping up his work on the final season of HBO’s popular series Boardwalk Empire, he decided it was time to buy a new digital audio recorder.

He needed a product that offered the additional audio tracks necessary for his next project, HBO’s new mini-series, Show Me a Hero.

With the guidance of Gotham Sound, Stettner chose Sound Devices’ 970 audio recorder for its 64-track capabilities.

Show Me a Hero is based on the real-life story of former Yonkers Mayor Nick Wasicko, played by Oscar Isaac.

The mini-series covers Wasicko’s battle with the federal courts, which in 1987 ordered the construction of public housing in the predominantly white, middle-class section of Yonkers. The events divided the city, with several heated city council meetings disrupted by yelling and protests. Stettner knew that successfully capturing the audio of these recreations would require the recording of multiple ISO tracks.

To recreate the city council meetings, several cast members playing the council, the mayor and the clerk sat at desks with a microphone positioned in front of each actor. Stettner also needed to record four to five speaking parts from the actors depicting hecklers in the crowd. His second mixer captured the council’s audio, while he concentrated on the lines of the hecklers. Since there were three cameras shooting at the same time, the 970s were tasked with capturing the fast-paced dialogue from the whole scene—all angles, all the time.

“With the 970, I had access to 64-tracks, all of which were Dante-enabled, so I could route all the fixed mics via a LAN to the individual ISO tracks. I also sent them to a second mixing board with a second mixer where they were combined to a mono mix, which I incorporated into my mono dailies mix,” Stettner explained. “Gotham Sound was great in helping set this up for me, and even had one of their technicians on site so that I could correctly interface between the two boards quickly.” 

In addition to two Sound Devices 970s running redundantly, Stettner used a Yamaha QL5 Series digital console and Rio1608-D input/output box connected to 12 Shure SM57 microphones. Audio was then routed over Cat5 to the LAN. The mics were routed to two places, the Dante ISO inputs on the 970 and the channel inputs of the Yamaha QL5.

“By having access to all the stem elements, we were able to give editorial many ways to make sense of material recorded in a very hectic audio environment,” Stettner says. “The 970 made that possible—this is when its flexibility really opened my eyes to what a great tool it is.”

Stettner also had Sound Devices’ 633 on hand for over the shoulder locations.

After wrapping, Stettner praised not just the performance of the gear, but also the technical support that Sound Devices provided. “Anyone I ask says that Sound Devices, in terms of how they deal with people and how they develop products, is the place to be,” he says. “Any time we were stumped, there was a simple fix that they knew just how to implement.”

The Sound Devices 970 is the company’s first-ever audio-only rack-mounted solution. The half-rack, 2U device simplifies any application that requires high-quality, high-track-count audio recording. The 970 records 64 channels of monophonic or polyphonic 24-bit WAV files from any of its 144 available inputs. Available inputs include 64 channels of Ethernet-based Dante, 64 channels of optical or coaxial MADI, eight channels of line-level analog and eight channels of AES digital. Any input can be assigned to any track. It also supports recording of up to 32 tracks at 96 kHz.

Sound Devices

Posted by House Editor on 08/19 at 01:31 PM

Cut ’Em Off At The Pass: Effective Uses Of High-Pass Filtering

So there I was, system engineer at a county fair gig. The act of the day was a traveling ’60s reviews with three or four artists who were, shall we say, past their prime.

They weren’t carrying engineers, so we got the duty.

Soundcheck went fine. The artists cruised through their paces and the hired back-up band was surprisingly good. Nothing to do but hit catering and wait for the “white hair, blue hair and no hair” crowd to show up.

Show time. The band started the intro, everything was rocking in an old school sort of way and the emcee/star came out. He was much more animated than he had been at soundcheck – running around the stage, exhorting the crowd to put down their walkers and dance, generally getting them in the mood.

Suddenly I heard a phantom kick drum that was waaaay off the beat. I cued up my cans and began to solo channels.

The offending thump came and went, but I finally put my eyes and ears together and realized that the star, we’ll call him “Frankie” for the sake of this article, was running around clapping his hands while holding his SM58.

At first I tried riding the mute button on his microphone, but I was spending so much time on him I couldn’t mix the rest of the show.

So I reached for the variable high-pass filter knob and ran it up to 300 Hz. It thinned his voice out a bit but I doubt anyone noticed but me.

Problem solved.

Combat The Unwanted
High-pass filters are probably one of the most under-utilized features on the console. The most common use has traditionally been to combat unwanted proximity effect, which is the tendency of directional mics to increase their output at low frequencies as the sound source gets closer to the mic.

Cardioid and hypercardioid mics get their directional characteristics from ports in the mic capsule that allow sound to impinge on the rear of the diaphragm as well as the front. The added length of the ports creates a difference in path length between sounds hitting the front of the diaphragm and the rear.

Pressure differences between the front and rear of the diaphragm are what make it move. These different path lengths cause a difference
in pressure because of two factors: phase and amplitude.

The phase component is dominant at higher frequencies. A 20 kHz wave is slightly more than a half-inch long. The path length difference from the front of the diaphragm to the rear is large as a percentage of the wavelength, so almost complete cancellation can occur.

This is one reason why microphone directivity breaks down as frequency decreases, and it is also why the diaphragms of cardioid mics are damped at about 6 dB per octave as the frequency rises. Remember: more pressure difference equals more diaphragm movement.

But the key to proximity effect is the amplitude disparity. The inverse square law tells us that every time we double the distance from the source to the diaphragm, we lose 6 dB. This is very powerful at short distances; for example, the difference between a singer being a quarter-inch from the mic and a half-inch from the mic is 6 dB.

It also means that the difference in path length from the front of the diaphragm to the rear becomes more and more significant as the source gets closer.

Since phase cancellations are a fixed percentage of amplitude at any given frequency the amplitude factor becomes much more dominant at close distances than the phase factor. The phase part of the equation has less and less effect at longer wavelengths while the amplitude part holds true at all frequencies.

Hence proximity effect.

Proximity effect can go as high in frequency as 500 Hz depending on the mic, although 200-300 Hz is more common. The amplitude gain can be as much as 16 dB! This is probably why high-pass filters were put on mics and into consoles in the first place.

Another Benefit
But sweepable high-pass filters can also be used to help you clean up your overall mix.

One of the things we learn from audiology is that lower frequency sounds obscure higher frequency sounds, but not the other way around. This is one of the principles that makes sound masking work.

It’s useful in sound masking systems, but in a live performance situation, not so much.

Many live mixers react to this unconsciously when they reach for the house graphic and hack away at 125 Hz and 160 Hz. True, many rooms react poorly in that frequency range, but the room is only one part of the problem.

Let’s think about the physics of low frequency sound waves.

A 100 Hz wavelength is 11.3 feet long (at sea level, at 72 degrees Fahrenheit etc., etc.). This is typically above the crossover point for subwoofers, so it’s probably being reproduced by the main arrays.

In order to provide good directivity at any frequency, the array must be larger than the wavelength. If the array is not larger than the frequency of interest, the sound waves wrap around the array and it behaves as an omnidirectional source.

Even if the line array is fairly long, you only get the directivity benefits in the vertical axis. Chances are, the array is four feet wide (or less), which means that in the horizontal plane, pattern control starts to break down at around 250-300 Hz.

What is in close proximity to the array on the horizontal axis? The stage. And the mics on the stage.

Even if the subs are being run from an aux send on the console (which I highly recommend), there is still energy from the sources being routed to the subs that finds its way back into the stage mics.

Because the same laws of physics hold true for stage sources as for main arrays, the mics are picking up the desired musical content in these frequency ranges – plus the adjacent instruments and floor wedges, plus the room resonances, plus the wraparound from the main system in the longer wavelength frequencies below about 300 Hz.

This is happening even if we don’t consider the artist clapping with a mic in his hands or tapping his foot on the mic stand base. And to compound the problem, the cardioid pattern of the mics breaks down in the lower frequencies as well.

The inverse square law (minus channel compression) is your only friend at this point!

Incremental Steps
So, what’s a poor sound engineer to do? Directional cardioid subs and cardioid sub arrays can help enormously with the least directional part of the mains, which is often closest to the stage.

We’ve already made gains in cleaning up the stage sound (at least in some cases) with tools like in-ear monitoring and instrument amplifiers located off stage in isolation cabinets.

While these techniques are incrementally helpful, there’s another tool at our disposal: the console channel’s variable high-pass filter.

The earlier we can deal with these issues in the signal chain, the better, which is why high-pass filters are found on many outboard mic preamps as well. If your mics have a shelving filter, try that first. If it doesn’t degrade the instrument sound, leave it switched in.

Next, at soundcheck, start your equalization process for each mic by sweeping the high-pass filter up until you hear it affect the sound. Obviously, there are some inputs that might be left out of this process like the kick drum, bass guitar and a low piano mic. DIs and other direct feeds don’t count because they aren’t picking up ambient sound.

A sharper knee and a steeper slope will allow you to set the filter to a higher frequency without degrading the natural tone of the source up to a point. Too steep of a slope can cause a filter to “ring.” Filters have resonances too.

Then, during the show, solo each mic with headphones that provide good low frequency isolation and response (I like beyerdynamic DT770s), and you may find you can cheat your high-pass filters upin frequency a little higher.

Oh and by the way, have the monitor engineer try this too, only he/she can be quite a bit more aggressive with it. The performers on stage don’t have high-pass filters on their IEMs, and many ear molds don’t do a great job of isolating lower frequencies.

The Payoff
Using this approach should lead to cutting less in the 125-200 Hz range on the system EQ because you are solving the problem at the source.

You’ll also be surprised at the increased clarity in your overall mix. The system will have more headroom as well since the frequency ranges we’re dealing with are real energy hogs.

Remember, garbage in garbage out. Why deal with it in your mix when you can cut it off at the pass?

The high pass, that is.

Bruce Main has been a systems engineer and FOH mixer on and off for more than 30 years. He has also built, owned and operated recording studios and designed and installed sound systems.

Posted by Keith Clark on 08/19 at 12:14 PM
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Professional Wireless Systems Supports America’s Night Of Hope In San Francisco

Joel and Victoria Osteen kick off 7th annual event at AT&T Park with wireless coordination by PWS.

America’s Night of Hope, with Joel and Victoria Osteen, kicked off its 7th annual event this year in San Francisco’s AT&T Park.

Out front a packed stadium greeted the mega-church pastor and his family with thunderous applause, while behind the scenes Professional Wireless Systems (PWS) was on site providing frequency coordination, backstage communication and event support.

PWS successfully integrated Joel Osteen’s wireless equipment together for coordination within one of the nation’s most congested RF cities.

Night of Hope, is an inspirational experience meant to bring people together in an environment of encouragement, belief and worship.

The two-hour event delivers heartfelt words of encouragement from Victoria Osteen, a special testimony from Dodie Osteen, and concludes with prayer and an inspirational message from Joel Osteen.

This year, PWS managed 70 frequencies site-wide. The Radio Active Designs (RAD) UV-1G system was on hand this year and was a valuable asset in clearing the spectrum of communications to make room for mics and in-ears. San Francisco is one of the nation’s most congested RF areas with an abundance of DTV stations in close proximity to each other; creating the potential for spectrum lockdown PWS’ frequency coordination was paramount at the AT&T Park stadium to ensure an interference-free event.

Along with coordination, PWS also supported all wireless gear provided by Joel Osteen’s audio production team. The equipment, from Shure, consisted of 24 microphones and 14 in-ears. John Garrido, the lead RF coordinator during the event, organized all gear in such a manner that the Osteen’s had complete freedom of movement during their services and the ability to convey their messages without delay.

“Working in a major metropolitan city like San Francisco, it is important that we are able to have clear communication for the onstage talent,” says Jim Van Winkle, general manager of PWS. “John Garrido did an excellent job of coordinating the frequencies, securing a pathway to ensure that there was no interference or signal problems. The evening was a resounding success.”

Professional Wireless Systems

Posted by House Editor on 08/19 at 08:52 AM
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