Wednesday, November 19, 2014
Engineer Zoe Martin Tours & Teaches With Soundcraft Vi Series Consoles, Realtime Rack
Analog background made Soundcraft the winner when it came to her preferred mixing console
UK-based live mix engineer Zoe Martin utilizes Harman’s Soundcraft Vi6 consoles and the new Realtime Rack while touring with international artists and bands such as Maverick Sabre, Bonobo and The Radiophonic Workshop.
Having acquired substantial experience over the past decade, Martin also passes her mixing skills onto university students at the Brighton branch of the British and Irish Modern Music Institute with a Soundcraft Vi1 console.
Martin’s analog background made Soundcraft the winner when it came to her preferred mixing console, as the design and layout of the Vi Series are geared towards ease of use and instant response.
To this end, the Soundcraft Realtime Rack was also a logical addition to her toolkit, bringing UAD plug-ins to Soundcraft consoles and adding full remote snapshot triggering and storing. With four SHARC DSPs and a library of UAD powered plug-ins, as well as UAD direct developer plug-ins, it offers stable DSP levels and premium sound quality in a scalable, low-latency processing environment.
“I’ve always liked the Vi consoles, because they make the most ergonomic sense to me,” Martin says. “The Soundcraft consoles are the most analog-like digital desks in terms of layout and give me exactly what I need.
:The Realtime Rack is also a nice feature,” she continues. “It smoothes out each of my mixes where I have the precision bus compressor on. The compressors on the Vi6 are very nice to begin with, but the Realtime Rack definitely gives the whole mix a nice friendly squeeze.”
“Whether I’m teaching or touring, Vi consoles allow me to adapt quickly to situations because they’re so user-friendly. For example, if a mix needs more of a specific sound at the last minute, I find that with these consoles you can make on-the-fly changes instantly. I’m a very visual operator, so with features like Vistonics and FaderGlow, I’m able to see all the settings without much effort and can keep my eye on the band.”
Eventide Announces Strategic Partnership With Yamaha
Developing signature effects for new RIVAGE PM10 flagship console
Eventide announced the formation of a strategic partnership with Yamaha to develop signature effects for a future update to its new flagship live sound console, the RIVAGE PM10. (Read more about this new console here.)
The Eventide H3000 Live plug-in will include some of the most widely used signature Eventide effects used in the live sound industry.
“We have the utmost respect for Yamaha’s deep roots in the music equipment industry,” states Ray Maxwell, Eventide vice president of sales and marketing.” It’s an honor to work with the company who is the largest manufacturer of musical instruments in the world.
“Their pioneering work developing some of the earliest digital consoles continues to be refined, incorporating the evolving demands for integrated workflow necessary for today’s demanding live sound productions,” he continues. “The marriage of classic Eventide effects with the state-of-the-art Yamaha RIVAGE PM10 console is a match made in heaven.”
“Our new flagship RIVAGE PM10 system culminates more than 40 years of PM series history while heralding a new era of extraordinary sound, operation, and reliability.” commented Chihaya “Chick” Hirai, PA department manager of Yamaha Corporation. “Processing quality has always been a major strength of Yamaha digital consoles. We are excited about the new collaboration with Eventide, world renown for the legendary Harmonizer effects processors.”
Yamaha Pro Audio
New Master Fader v3.0 For Mackie DL Series Mixers
Significant update ready for new DL32R, also provides benefits to DL1608, DL806 users
Mackie has announced the availability of new Master Fader v3.0, providing significant upgrades for the new DL32R wireless digital mixer with iPad control as well as DL1608 and DL806 mixers. It’s available for free download from the Apple App Store here.
“Master Fader is designed from the ground up for wireless mixing. It’s the most intuitive and easy-to-use live sound control app out there,” states Ben Olswang, Mackie senior product manager. “Now, with a fresh user interface that delivers faster workflow and a whole new toolset designed to meet the rigorous demands of professional live sound applications, Master Fader is better and more powerful than ever.”
New features include the addition of four subgroups and four VCAs. These new grouping options deliver mix management tools critical for professional applications. Subgroups can be stereo-linked and feature dedicated processing. VCAs offer flexible control over groups of channels. Users can dial in the mix and get single-fader control over groups like drums, guitars and more.
Also new is the overview screen, which delivers at-a-glance information for all input and output channels, allowing simple, fast navigation. In addition, digital trim has been added to each DL1608/DL806 input.
“With digital trim, you get more mobile control than ever before,” Olswang notes. “Just get the preamp gain in the ballpark and adjust as needed from anywhere in the venue. It truly frees you to mix from anywhere.”
In addition, there’s also now the ability to patch any output to the DL1608/DL806 record path so that users can select exactly which outputs they want to record. This is useful for creating a completely separate recording mix from what is in the left/right mix.
With Master Fader v3.0, users can now choose a color for each channel, providing immediate visual grouping for easy identification. Also added are enhanced headphone controls, including level and delay, for seamless remote listening. Plus, users can now export presets, shows and even the entire system via email, Dropbox and iTunes.
“All of these new features mean serious workflow improvements for any DL Series user,” says Olswang. “Master Fader has been updated more often, with more new features added, than any other live sound control app out there. We are happy to provide more and more functionality to our DL community.”
Constant updates to the Master Fader user interface deliver increased workflow flexibility, improved speed and ease-of-use. “We’re dedicated to continuously improving Master Fader’s workflow through simple App Store updates,” Olswang says. “Each of these user interface updates delivers real-world results that make wireless mixing more intuitive and easy to teach and learn.”
Master Fader v3.0 is available for free download from the Apple App Store here.
Master Fader 3.0 is compatible with Mackie My Fader v3.0 (available soon). It is not compatible with My Fader v2.0. Do not update to Master Fader 3.0 at this time if you still need My Fader 2.0 control.
Additional specific information about the new DL32R wireless digital mixer is here, and a range of videos about numerous facets of the DL32R are here.
Tuesday, November 18, 2014
Yamaha Launches New Flagship RIVAGE PM10 Digital Console
Comprised of CS-R10 control surface, DSP-R10 DSP engine, RPio622 I/O rack, three types of RY cards, and two types of HY cards
Yamaha Corporation Japan has launched the RIVAGE PM10 digital mixing console, comprised of the CS-R10 control surface, DSP-R10 DSP engine, RPio622 I/O rack, three types of RY cards, and two types of HY cards to provide the flexibility to configure and match the scale and functional requirements of any application.
More than 10 years have passed since the Yamaha PM1D and PM5D helped define the digital console market, with the new RIVAGE PM10 significantly increasing the quality and versatility necessary in modern live sound environments.
“Our new flagship RIVAGE PM10 system is the result of more than 40 years of PM Series history and reinforces our passion to provide industry-leading commercial audio products,” states Laurence Italia, vice president/general manager, Yamaha Commercial Audio. “RIVAGE PM10 will ignite the market for a new era of extraordinary sound, operation, and reliability. Needless to say, RIVAGE PM10 would not be possible without the dedication of the Yamaha Japan R&D team as well as feedback from our customers.”
Under the fundamental philosophy of Yamaha live sound consoles, Hybrid Microphone Preamplifiers have been newly-developed, as has the analog section both in terms of components and overall design, achieving a pure, natural sound.
Precise models of the Rupert Neve Designs (RND) transformer circuitry and SILK processing have been created utilizing Yamaha’s VCM (Virtual Circuitry Modeling) technology. SILK processing includes selectable “RED” and “BLUE” characteristics, as well as a continuously variable “TEXTURE” knob that facilitates sonic shaping at the input stage, adding sparkle or power as required.
Josh Thomas, general manager of Rupert Neve Designs, states, “We’ve done a number of listening sessions to compare our hardware to the transformer and Silk Texture emulations that Dr. K of Yamaha Japan and the development team achieved. Rupert and I are both very impressed by how closely the emulations get to the original analog designs. We hope that you all enjoy the rich Rupert Neve sound.”
The channel EQ and dynamics have been upgraded with three newly developed algorithms plus the classic Yamaha “Legacy” algorithm. A new “Precise” algorithm includes Q parameters on the low and high shelving filters that allow filter overshoot to be flexibly controlled for added musical impact. An “Aggressive” algorithm has been designed for the most musical response, and a “Smooth” algorithm offers smooth, natural control while retaining the fundamental character of the source.
Two dynamics stages are provided, each functioning as a gate, compressor, ducker, or de-esser as required. Two compressor types are included: “Legacy Comp” features standard Yamaha digital console compressor characteristics; and “Comp260,” a VCM model of a popular analog comp/limiter from the mid-70s.
A total of 45 plug-ins are available for creative processing, with substantially increased processing power. New and noteworthy are plug-ins created in collaboration with Rupert Neve Designs, TC Electronic, and Eventide.
Collaboration with Rupert Neve Designs has resulted in the “Rupert EQ 773,” “Rupert Comp 754,” “Rupert EQ 810,” and “Rupert Comp 830,” all VCM models of Neve designed outboard devices from the 70s and 80s. All four models deliver the musical vintage outboard sound still favored by many engineers.
The alliance with TC Electronic has resulted in the inclusion of two new reverb plug-ins: the “VSS4HD” room simulation reverb that offers a multitude of reflection settings with musical-sounding reverb processing capabilities; and the “NON LIN2” plug-in that can function as an envelope-filtered gate reverb without requiring a trigger.
“I wish to congratulate Yamaha with raising the bar for live console design significantly,” says Thomas Lund, CTO of TC Electronic. By directly integrating the world’s finest EQ, dynamics and reverbs, this mixing system is truly outstanding. Therefore, a big congratulations to the users also.”
Cooperation with Eventide has produced the “H3000-LIVE” harmonizer, an accurate reproduction of the H3000 Ultra Harmonizer algorithm with an interface streamlined for live sound use. “The marriage of classic Eventide effects with the state-of-the-art Yamaha RIVAGE PM10 mixing system is a match made in heaven,” notes Ray Maxwell, Eventide vice president of sales and marketing.
Console operation has been given the ample attention as well, with the Yamaha-familiar Selected Channel section implemented in full, allowing all parameters of the selected channel to be directly and intuitively controlled. There are also two large LCD touch screens that provide logical continuity with the faders, offering the same operability as the Centralogic interface in 12-channel groups.
In addition to this dual interface hybrid operation style, horseshoe-ring encoder position indicators, refined panel layout, and several other details contribute to significantly improved overall operability.
New features extend the usefulness of the scene memory: “Isolate” enables the engineer to specify entire channel modules including EQ and dynamics libraries, etc., that will be protected from recall operations without altering the Recall Safe settings, which is significant for situations where scene memory is heavily used.
The Overlay Filter can be “overlaid” on a current mix to apply temporary offsets to fader and mix/matrix send levels independently from Scene Recall, valuable in situations that require sudden changes and the ability to easily revert to the original settings.
The CS-R10 control surface and DSP-R10 DSP engine are connected via a dedicated Cat-5e ring “Console Network.” The DSP engine connects to the RPio622 I/O racks via the newly developed TWINLANe ring network using multi-mode optical fiber. The fact that both networks use simple ring connections facilitates redundancy and reliability.
TWINLANe is an original Yamaha protocol that can handle up to 400 96 kHz audio channels over distances of up to 300 meters with low latency and redundancy for high reliability. Up to two DSP-R10 DSP engines and eight RPio622 I/O racks can be connected in a single ring.
Connection to a computer for multi-track recording, connectivity to the NUAGE advanced production system, Yamaha CL and QL Series consoles, as well as other external equipment is possible via a Dante network. Dante works with TWINLANe to create an exceptionally flexible, reliable system.
The new RIVAGE PM10 console has an expected delivery during 2015.
Click to enlarge
Yamaha Commercial Audio
Community R Series Delivers 360-Degree Coverage At Białystok City Stadium In Poland
More than 100 loudspeakers deployed to cover grandstand, promenades and public areas around building
Białystok City Stadium, located in the largest city in Northeastern Poland and the capital of the Podlaskie Voivodeship, has undergone major redevelopment, with a new 360-degree grandstand hosting up to 22,400 who are served by a new sound reinforcement system with Community Professional loudspeakers.
The system was designed by Tommex Żebrowscy Sp. J., with installation completed by Zeto S.A. in time for the opening of the modernized venue earlier this year, under the direction of project manager Zbigniew Kaczmarczyk.
The main loudspeaker set comprises 48 Community R2-474s and three R2-52s. The combination provides the dispersion patterns required for complete coverage of the stadium’s 360-degree grandstand. In addition, 38 R.25-94s and 41 R.5-99s deliver audio for the stadium’s internal promenades and public areas around the building.
The main system components are housed in the commentary room and linked to four separate amplifier rooms. The audio signal is transmitted over the stadium’s network using the Dante protocol and is triple redundant, with two fiber rings and a turbo function ring. It’s the largest system based on Dante operating in any stadium in Poland.
The system uses a Dynacord CMS 2200-3 audio mixer, 16 x 16 DSP equipped digital control matrix and a DPM 8016 digital matrix manager. The audio matrix is 32 × 32 and handled by Dynacord’s P 64 digital audio matrix units. By using the IRIS-Net network, each element of the system is continuously monitored and all operations are controlled via a Dell all-in-one touch-screen PC.
Twenty Dynacord DSA 8410 4-channel amplifiers provide a total of 80 kW to power the Community loudspeakers. A DPC 8015 microphone, located on the Events Security Commander desk, has full priority over other audio signals, enabling the effective communication of voice messages in the event of non-standard events.
“Visitors to Białystok City Stadium will not only enjoy their sports with great audio quality, but they can also be assured that their safety is our primary concern with reliability being a fundamental consideration in the design and every component of the system,” says Marcin Zimny of Tommex Żebrowscy.
Roland & Mobeon Partner For Streaming Media West 2014 Conference
Largest streaming and live production conference on the West Coast
Roland and Mobeon have teamed up to be the official streaming partners of Streaming Media and Streaming Live West conference currently taking place (November 17–19, 2014) at the Hyatt Huntington Beach.
The conference is the largest streaming and live production conference on the West Coast, offering the opportunity to learn about the latest online video technologies and new business strategies.
Mobeon, an interactive digital is responsible for producing and streaming three conference rooms plus the keynote address, while Roland has supplied a VR-50HD live AV production and streaming switcher for each seminar room.
The VR-50HD outputs up to1080p over USB 3.0, SDI and HMDI, which simplifies setup and increases the live production workflow for 47 seminars and discussion panels. This output will be fed to a computer running Wirecast as the encoder and then sent to USTREAM for streaming.
Mark Alamares, principal of Mobeon, selected the VR-50HD not only for its streaming capabilities but also the ability to record the live production simply by connecting it via USB 3.0 to a computer with capture software. Other reasons he cites include portability, ease-of-use, and quick setup.
The VR-50HD can handle multiple inputs with scaling, analog or digital, computer or video resolutions, plus mix audio. Alamares says, “It’s an all-inclusive streaming and live production system in one package.”
“This will be our third Streaming Media Conference in a row but the first time using the Roland VR-50HD. We’re excited to leverage the power and technology in such a compact box to produce the event this year,” he adds.
“The Streaming Media Conferences are great events to learn about and discuss streaming solutions. We are excited to be the official streaming partner along with Mobeon and to showcase the Roland VR-50HD in a live setting to streaming professionals”, says Rob Read, Roland marketing communications manager.
Streaming Media West
In The Studio: Analog Tape Recording Basics And Getting That Sound
If you’re looking for a prime example of what Toffler wrote about in Future Shock, look no further than analog tape.
In little more than a decade, the two-inch multitrack tape machine has gone from studio staple to relic rarity. And while many audio veterans wax nostalgic for that warm analog sound, few will admit to missing the work that went with it.
These days, owning an analog tape machine is somewhat akin to driving a classic car, with ongoing maintenance, scarcity of parts, and exotic fuel (analog tape) that’s expensive and hard to find.
So while a handful of top studios still offer those classic spinning reels (and the engineers to maintain them), the good news for the rest of us is that there are now more convenient ways to achieve that classic magnetic sound.
A Bit of History
Analog recording, of course, predates tape — with everything from wax cylinders to wire being used to capture a performance. But when American audio engineer Jack Mullin discovered a pair of German Magnetophon machines during World War II, he knew right away he was on to something big.
The format offered two major advantages over the acetate disks of the day: a recording time of more than 30 minutes, and the ability for recordings to be edited. It was the first time audio could be manipulated.
Mullin brought the two Magnetophons back home after the war and demonstrated them for Bing Crosby at MGM Studios in 1947. Crosby immediately saw the potential for prerecording his radio shows, and invested a small fortune of $50,000 in a local electronics company called Ampex to develop a production model.
Ampex and Mullen soon followed with commercial grade recorders. One of the first Ampex Model 200 recorders was given to guitarist Les Paul, who took the concept of audio manipulation to a higher level. Paul had already been experimenting with overdubbed recording on disks and, quickly realizing the potential for adding more channels and additional recording and playback heads, came to Ampex with the idea for the first multi-track tape recorders.
The format evolved from two tracks to three and four, and although Ampex built some of the first eight-track machines in in the late 1950s, most commercially available machines were limited to four tracks until 1966, when Abbey Road recording engineers Geoff Emerick and Ken Townshend began experimenting with multiple machines during the recording of Sgt. Pepper’s Lonely Hearts Club Band.
The Studer A800 Multichannel Tape Recorder records up to 24 tracks of audio.
Ampex responded to the demand the following year, introducing the revolutionary MM-1000, which recorded eight tracks on one-inch tape. Scully also introduced a 12-track one-inch design that year, but it was quickly overshadowed by a 16-track version of the MM-1000, using two-inch tape. MCI followed in 1968 with 24 tracks on two-inch tape, and the two-inch 24-track became the most common format in professional recording studios throughout most of the 1970s and 1980s.
With the prevalence of home and project studios and digital technology in the late 1980s and 1990s, a number of other tape formats emerged, including various multitrack on-reel and cassette configurations as well as multiple digital tape formats. But for the sake of this article, we’ll focus mainly on multitrack analog tape, the most sonically revered recording medium of all time.
How A Tape Machine Works
In the simplest of terms, magnetic tape consists of a thin layer of Mylar or similar material coated with iron oxide. The tape machine head exerts a charge on the oxide, which polarizes the oxide particles and effectively “captures” the signal. It’s a process that creates some interesting byproducts, many of which directly influence the sound of the recording.
Probably the most commonly cited chracteristic of analog recording is its “warmth.” Tape warmth adds a level of color to the sound, primarily softening the attacks of musical notes, and thickening up the low frequency range. Recording at slightly hot levels to analog tape can also produce a nice distortion that works well with certain types of music such as rock, soul, and blues.
As multitrack recording evolved, a number of different manufacturers began to emerge. By the early 1980s, Ampex was no longer the dominant multitrack manufacturer, facing stiff competition from MCI, Studer, 3M, and Otari. Although a handful of smaller manufacturers, including Stephens, Aces, and a few others also entered the fray, Ampex, Studer, 3M, MCI (later owned by Sony), and Otari became the dominant brands.
Each of these manufacturers’ different models became loved (or despised) for their mechanical attributes and characteristic sound. In the day, a recording studio’s model of multitrack tape recorder was considered as intrinsic to its sound as their acoustics, console or microphone collection.
The Subtle Differences
A multitude of factors influence each machine’s characteristic sound, beginning with the tape heads, amplifiers, and other electronics.
Beyond that, other factors have a bearing on the sound of an analog recording, some of which are unique to each particular machine. Variations in the machine’s speed stability (wow and flutter), alignment of the tape heads and the angle of the tape, condition of the heads (cleanliness, magnetization, etc), tape tension, and other physical factors are just a few things that can affects the sound of a recording.
Besides the machine itself, other factors can affect the particular sound of an analog recording, including the brand of tape used. Back in the heyday of analog, the major brands of tape each had their supporters and detractors. Ampex tape was one of the leading brands, with their 456 formula being the most prominent.
The brand of tape used subtly affect the tonal color of a recording.
Other popular brands included AGFA and 3M. Each tape formulation imparted its own subtle sound to a recording, and each machine had to be realigned each time a brand was changed. Some studios made a policy of sticking to one brand of tape, but it was not uncommon for variations to occur even within different batches of the same brand of tape.
Tape speed is another major factor. Faster tape speeds tend to deliver cleaner sound quality, since the signal is spread over a larger area and the signal-to-noise ratio is increased. The most commonly used speeds with two-inch tape are 15 and 30 IPS (inches per second). Although 30 IPS delivers better overall sound quality, most pros agree that lower frequencies sound better at 15 IPS. Indeed, in the modern era, when tape is most often being used for its sonic effect, slower speeds prevail.
Getting That Analog Tape Sound
Although owning a classic two-inch Studer or Ampex tape machine certainly earns bragging rights, in today’s DAW-oriented world, the fact is that fewer of us would opt to record to analog tape, even if we could. Space considerations, cost considerations, and the scarcity of tape and parts are only the beginning.
The fact is, tape’s destructive editing can be a slow and tedious process in a world where time is truly money. And even the medium itself is no longer cost effective. A single reel of two-inch tape averages around $200. At 15 IPS, that tape holds around 30 minutes of 24-track recording time (half that at 30 IPS). Compare that to a 2TB hard disk, currently selling for just over $100, that can hold many hours of multitrack audio, and you can see why running tape for every project is not an option for most of us.
Screenshot of the Studer A800 plug-in from UA.
Fortunately there are a number of great sounding plug-in processors for your DAW that can bring some of that analog tape warmth and “glue” to your tracks. For example, Universal Audio’s popular Studer A800 Multichannel Tape Recorder is a very faithful emulation of the original machine’s sound, having been painstakingly developed over a one year period with input from the original manufacturer. In fact, the sonic differences between the A800 plug-in and the original A800 hardware are so minute, that many of the world’s top engineers opt to use the plug-in for their day-to-day work.
Aside from the obvious convenience factor, one of the biggest advantages of tape emulation plug-ins are their flexibility. You can choose to process only certain tracks, rather than the whole mix — imparting the warmth, low-end bump, and cohesive properties of tape to the drums and bass tracks, for example, without adding any tape color to guitars and vocals. Or you can add just a hint of tape compression to the mix, without oversaturating things the way an actual two-track machine might.
Regardless of how you choose to implement it, the sound of analog tape can be a great addition to your digital mix. Take a UAD tape emulation plug-in like the Studer A800 and test it on a few tracks. You might find it’s just the thing to add a bit of nice, understated warmth, cohesiveness and punch to your mix. Tape lives on.
Daniel Keller is a musician, engineer and producer. Since 2002 he has been president and CEO of Get It In Writing, a public relations and marketing firm focused on audio and multimedia professionals and their toys. Despite being immersed in professional audio his entire adult life, he still refuses to grow up. This article is courtesy of Universal Audio.
Monday, November 17, 2014
Twin DiGiCo SD7 Desks Reinforce The “Stars Under The Stars” At Hollywood Bowl
Consoles enabled production staff to upgrade the operation of the installed PA system to 96 kHz resolution
The Hollywood Bowl incorporated a pair of DiGiCo SD7 digital consoles – one for front-of-house, one for monitors – for its busy 2014 summer series of concerts, which came to a close in late September, enabling production staff to upgrade the operation of the installed PA system to 96 kHz resolution.
This year’s 15-week summer series, anchored by the Los Angeles Philharmonic, featured performances by major artists from all genres of music, including Gloria Estefan, Yo-Yo Ma, Elvis Costello, Herbie Hancock, Robyn, Steve Martin and a host of others.
The iconic 18,000-seat Hollywood Bowl may have been named Pollstar’s best major outdoor concert venue for an unprecedented tenth consecutive year in early 2014, but the team behind its audio technology is not content to rest on its laurels.
According to Paul Geller, production director at the Hollywood Bowl for more than three decades, the PA system sounded great at 48 kHz. “But when you hear it at 96k, there are certain subtle differences that you pick up, such as the dynamic range of the highs and lows,” he says. “It also produces a very natural acoustic sound. It’s a great marriage between the DiGiCo console and the L-Acoustics loudspeaker system.”
The Hollywood Bowl’s core audio team includes Fred Vogler, principal sound designer and mixer for the Los Angeles Philharmonic, which, along with the Hollywood Bowl Orchestra, makes its summer home at the venue; Michael “Shep” Sheppard, head of audio/video; and Kevin Wapner, assistant audio/video and monitor mixer.
Vogler reports that the DiGiCo desks can comfortably handle even the largest productions. “We’re at 96k with 160-something inputs and it still sounds as pristine as with two channels. I think that a big part of the resolution we’re getting is the front end of that console. It’s got a lot of high quality mic pres, signal paths and processing. It’s wonderful to have that kind of front end and feed it out of an L-Acoustics speaker system on the back end.”
Vogler addresses the zoned loudspeaker system via the SD7’s matrix outputs, separately feeding the main left/right hangs, including flown subwoofers, down fill and front fill speakers, subs and mid-high boxes on the deck, plus various audience fill speakers. On the input side, says Vogler, he typically creates groups. “I’ll have a string group, woodwinds, percussion, horns or brass, lead vocals or solo instrument group, a band group and an effects group. I like the conductor or the notes on the page to do the balancing. It doesn’t mean that my mix is static, but I don’t get too heavy-handed with moving things up or down.”
The booth at front of house is well equipped with several different outboard high-end reverb systems, but Vogler also makes use of the many DiGiCo effects processers. “I use the tap delays and some of the onboard reverbs,” he reports. The Hollywood Bowl shell interacts with instruments on the stage, so for orchestral performances, “only a sparing amount of reverb is required,” he says. “I’ll pick one based on the piece of music or the need of a particular instrument.”
There is one DiGiCo feature that Vogler favors on vocals, the DiGiTuBe. “I really like the tube emulation; I like the warmth that it adds, especially if you have a vocalist that’s a little on the harder sounding side, or someone who tends to scream more,” he says. “It definitely rounds out that impact and makes it a little warmer and more engaging.”
He continues, “I also love the Dynamic EQ, where you can take a frequency band and put it into a compressor and apply it to every channel. You don’t have to take all the frequencies; you can just compress a band of frequencies. That’s a very handy feature.”
The two SD7s are on a single Optocore loop, with all of the DiGiCo DSP located on stage. “If I can’t see a conductor well and I need specific cues, I put the conductor cam on the console’s meterbridge screen. If I’m really bored I’ll put the monitor mixer on there,” laughs Vogler.
The venue’s summer series is non-stop, featuring almost daily concerts by the LA Phil, the Hollywood Bowl Orchestra, numerous guest performers and lease events – that is, concerts organized by third party promoters. “Because we’re doing one-offs we get one rehearsal, so there’s not a lot of time to fool around,” says Vogler. “We have several console templates that Shep has organized: orchestra, orchestra loud, jazz and rock. We have the basic EQs and onboard as well as external processing figured out for each particular type of presentation or program. Mike puts those together, labels the boards and organizes the session templates.”
Vogler is still coming to grips with every facet of the operation of the SD7 consoles. “It’s a very sophisticated board. I have a lot of flexibility in the way I can assign things or move things. I always chuckle when people ask me if I know what all the knobs and buttons do. I have to tell, them, no, but I’m working on it. I like the fact that I don’t know what every knob and button does. I’m still learning new things to do with it. It keeps me on my toes; I’m not bored using it. We’ve been very happy with it – and I’m eager to learn more.”
Posted by Keith Clark on 11/17 at 08:47 AM
Ashly Audio Gear Serves As Cost-Effective, Efficient Backbone For New System At Michigan Church
Ashly gear came in at a price point that allowed the church to meet its needs reliably without breaking the budget
New Hope Baptist Church in Grand Rapids, MI, recently underwent a renovation of its 750-seat sanctuary and associated overflow and ancillary rooms, with a new sound system centered on Ashly Audio processing, amplification, and user control as a centerpiece of the project.
The Ashly gear came in at a price point that allowed the church to meet its needs reliably without breaking the budget. Pro Audio Inc. of nearby Grandville, MI designed and installed the new system.
“The performance of Ashly gear is top-notch and yet its cost is very reasonable,” says Ken Reinecke, owner of Pro Audio Inc. “That’s a rare combination. In terms of sound quality, Ashly’s processing and amplification is transparent. What goes in is what comes out, plus whatever dynamics or equalization I’ve dialed in.
“And the gear can run at 96 kHz for super critical situations,” he continues. “Programming Ashly’s Protea DSP platform is straightforward. I just drag and drop whatever processing I need, and drag appropriate inputs to appropriate outputs. Linking groups is also handy. Ashly gear is also tremendously reliable – it basically never breaks or fails.”
Because space at New Hope Baptist Church was at a premium, it also helped that Reinecke could fit all of the system’s processing and amplification in just six rack spaces.
A new Soundcraft Si Performer 3 console serves as the user interface in the sanctuary itself. From there, an Ashly ne8800 8x8 Network DSP Protea processor handles all of the loudspeaker settings for the sanctuary’s new JBL VRX928LA line arrays, joined by JBL 915S subwoofers. The stereo line arrays are powered by a single Ashly nXe 3.04 network four-channel amplifier rated at 3,000 watts per channel, providing ample power to the main sanctuary system.
Additionally, output from the console feeds an Ashly Pema 8250 combination 8 x 8 Matrix Protea processor and 8-channel 250-watt amplifier. The Pema 8250 takes additional inputs, such as wall jacks for microphones or music players that are located in the narthex and other out rooms. The 8-channel Pema 8250 handles all of the audio routing, signal processing, and amplification for New Hope Baptist’s two nurseries, nursing room, bathrooms, and narthex.
Four Ashly WR-5 programmable wall-mounted remote controls located strategically around the out rooms provide customized, intuitive control of those inputs, as well as of the sound from the sanctuary.
“The nice thing about that arrangement is that folks can make changes in all those rooms without bothering the sound tech,” says Reinecke. “While the main system is controlled entirely from the Soundcraft console, the church staff can use the mic inputs and the Ashly WR-5 remote controls to play music or have a stand-alone PA for meetings and small events in those out rooms.”
Guerrilla Recording: Adding More Dynamic Range To Mixes
Even with today’s inexpensive recording systems, it’s possible to achieve a dynamic range of over 90 dB—in other words, the loudest sounds you record can be over 90 decibels louder than the background noise.
Managing all of the “area” between these two extremes, for each and every sound you record, is a skill that’s critical to making a good-sounding recording.
Fortunately, there are expanders, compressors, and limiters (collectively called dynamics processors) that help in this task. We’ll start here with expanders, and in a subsequent article, move along to the others.
Using An Expander
Of the three types of dynamics processors, compressors are probably the most familiar—but let’s begin with expanders, because they’re a bit easier to understand. A lot of people record without using an expander in the signal chain, and I think that’s a shame, because effective use of an expander can do an awful lot to clean up the tracks that you record.
Expansion refers to the process of increasing a signal’s dynamic range—making it bigger. Isn’t 90 dB a large enough dynamic range, you ask? Certainly—but that number refers to a system’s potential dynamic range, not necessarily the range you’ll get if you plug in a mic and start recording.
An expander is important in optimizing the actual dynamic range you get out of a system. An expander operates at the low end of the dynamic range, where signals are at their quietest, or perhaps nonexistent. In other words, when audio is coming through the signal chain, the expander may be doing nothing at all.
But when that audio stops coming through, the expander goes to work by lowering the signal further, expanding the background noise floor downward so that there’s a larger dynamic range overall (Figure 1).
Figure 1: Without expansion (left), the noise level in the signal chain can far exceed a stage’s background noise. With expansion (right), the noise is brought down to the noise floor, without affecting the signal itself.
Not surprisingly, this is called downward expansion. There is such thing as upward expansion, but you don’t really need to know about it. To understand this better, consider what happens when you plug a microphone into a mixing board and crank up the gain. If you talk or sing into the mic, you’ll hear yourself coming through the headphones loudly. (Careful—you might also get a feedback shriek if it’s too loud.)
But if you stop singing, odds are you won’t hear silence—especially in a bedroom or den Guerrilla studio. You’ll hear the heating or air-conditioning system, planes going overhead, street traffic, or your kid brother’s video game down the hall. This is all stuff that doesn’t belong on your recording!
Sure, domestic sounds are charming—if it’s 1970 and you’re Paul McCartney recording your first solo album. But we Guerrilla recordists are going after a slick, clean sound, and part of “clean” means not having anything on your tracks that you don’t want there.
Here’s where expansion comes in. You may have encountered a device called a noise gate, which is a crude form of an expander. In a noise gate, once the signal falls below a certain threshold, an electronic gate closes and no sound is allowed to pass through (noise or otherwise). However, when the signal begins to rise above that same threshold again, the gate opens up, allowing the signal to pass through once more.
Naturally, this also allows unwanted noise to pass through along with the signal, but the idea is that noise is less troublesome when signal is present to mask it. Like faint starlight in the night sky, noise is most noticeable when it’s by itself. Mix in a little signal (or sunlight in this analogy) and you’re less likely to notice the faint background stuff.
Figure 2: As the sound of a crash cymbal decays, it eventually falls below the noise floor and becomes inaudible.
An expander works on the same principle as a noise gate, but an expander is a bit more subtle: it’s not as obvious to the ear when it’s doing its thing. Here are the parameters that you’re likely to find on an expander, or the expander component of a compressor/expander:
Threshold: This control sets the level at which the expansion effect begins to set in. Imagine a cymbal crash that begins at 0 dB (the top of the dynamic range) and slowly decays to –_dB.
At a certain point in its decay, the sound of the cymbal will get so quiet that you’ll hear background noise mixed in with the cymbal, and at a still-later point you’ll hear only background noise, as the noise masks what’s left of the crash (Figure 2).
Figure 3: When you run the crash-cymbal sound through an expander, the threshold level determines how much of the cymbal’s decay makes it through before the expander closes down the noise.
If you were miking this cymbal by itself (perhaps to sample it for a collection of drum sounds), you might want an expander to kick in toward the tail end of the decay in order to take the background noise out of the sonic picture (Figure 3). The threshold control determines when this happens.
If you were to set the expander’s threshold to –30 dB, the expander would begin to shut down the signal when the cymbal decayed 30 dB below its initial peak. In this case you could get away with a lot more noise happening in or outside your studio without worrying about these sounds making it onto your cymbal sample.
But if you wanted to make a long, realistic sample of the cymbal and capture a lot of its decay, you’d probably want to set the control lower—perhaps –60 dB—and record it at a time when your studio is at its quietest, such as late at night. (Bummer for your sleeping housemates!)
Since the expander is set to a low threshold, the signal chain will be more susceptible to noise coming into the mic or created by the mic preamp.
To learn how to set the threshold control, here’s an exercise. Pretend you’re about to record a fairly loud electric guitar part using a miked amp. Set up your signal chain, with your mic in front of the amp, and gain-stage the chain so you’re exploiting the full dynamic range of all the stages without unwanted distortion.
Next, put the expander into the signal chain, ideally by way of your mixer channel’s insert jack. If the expander has compressor or limiter sections, bypass them by pressing the appropriate bypass switches or turning those sections’ threshold controls all the way up.
Turn the expander’s ratio knob (which I’ll discuss in a moment) all the way up, turn the threshold knob all the way down, and let your guitar sit on a stand with the amp running and the mic picking up the amp’s background noise.
Now put on the headphones, slowly turn up the expander’s threshold knob, and listen to what happens. At a certain point in the knob’s travel, the sound of the idling guitar amp will cut out—this is the point you’re looking for. Set the threshold slightly above this point.
Now, if you so much as touch the guitar’s strings, you should hear the gate open up, with the amp sound (and perhaps some string noise) coming through.
That’s what you want—the expander is gating out the noise, unless some signal is present as well (you touching the strings), at which point the gate opens to let both signal and noise pass through. The expander’s threshold is properly set, at least for now.
Figure 4: Below the expansion threshold, the level coming out of an expander’s output is much less than it would be if the expander weren’t in the circuit.
Ratio: In the exercise above, you probably noticed that when the expander’s gate closes, no sound is let through—the gate closes completely. That’s okay, but it isn’t ideal.
Setting an expander’s ratio control properly allows the circuit to close more gradually as a sound decays, and it allows the expander to stay slightly open after the sound has decayed below the noise floor.
It’s a bit like leaving a bedroom door open a little when you sleep: doing this lets in some of the light from the hall (similar to the background noise in this analogy) so you aren’t in complete darkness (total silence).
Figure 4 is a graph showing how an expander reduces the level at the output when the input level is below the threshold.
Figure 5: Recording a crash cymbal through an expander set to a high threshold and high ratio (left) can cut off the cymbal’s natural decay. Lowering the ratio and threshold (right) can result in a more natural sound.
Depending on the recording situation, setting up an expander to work like a noise gate—where it slams shut, resulting in sudden silence—can sound unnatural. This is particularly true with a gently decaying sound such as a crash cymbal, which would be abruptly cut off by noise-gate-like expander action (Figure 5).
A hard-closing gate can mess with the sound in even worse ways, for instance chopping off consonants at the ends of vocal phrases. We need to set the ratio control to avoid these problems, while still allowing the expander to clean up the sound.
With the guitar from the previous exercise still on its stand and the threshold control properly set, start turning down the ratio control and listen to what happens.
At a certain point, you’ll start to hear the sound of the idling guitar amp coming through—that’s the gate opening up slightly. When the ratio control is all the way down, the amp noise should be exactly as loud as if the expander weren’t in the signal chain at all; in other words, the gate is all the way open.
When you record a track, look for a happy medium between these points. When the signal chain is idling, the gate should be closed enough to quiet the track signifi cantly, but not closed so much that passages with no playing sound unnaturally silent next to played passages (unless, of course, that’s the effect you’re going for).
You also shouldn’t be able to hear the gate noticeably opening or closing when you start or stop playing. As much as possible, it should simply sound like your system is a lot quieter.
The trick to using an expander effectively is to find suitable threshold and ratio settings based on the sound you’re about to record, as well as the song you’re recording.
You want the expander to be responsive to any sound you make during the performance—in other words, to anything that you actually want recorded on your track—but not necessarily anything else.
Play lightly and let some notes or chords decay. Think about the performance you’re about to record: Will you be playing full-out through the whole track? Is there a point where you’ll need to hold a chord for several seconds? Will you be playing any passages very quietly?
Test out any such critical performance moments and listen to how the expander reacts. If the expander seems to be too sensitive to what you’re doing, turn up the threshold control a little.
Adjust the controls one at a time until the expander is doing its job cleaning up your signal chain, without calling attention to itself. You may need to compromise—one pair of settings may be good for one part of the song while another is good for a different passage.
Try to find settings that work as well as possible across the whole performance. If necessary, you can always punch in certain sections that require very different expander settings.
Attack & Decay: Most (if not all) expanders have these controls. You’ll recognize these terms if you have experience programming synthesizers: attack specifies how fast something rises, and decay specifies how fast it falls afterward.
In the case of an expander, attack determines how fast the gate opens when its threshold is suddenly exceeded, and decay determines how fast it closes again when the signal suddenly goes away. You can usually set these knobs and forget them.
Normally you want a very quick attack (so as not to cut off the beginnings of sounds) and a medium decay—perhaps around 200 milliseconds—to make sure the ends of sounds don’t get truncated.
The two sections of an integrated compressor/expander unit may have only one set of attack and decay controls but separate ratio and threshold; that’s okay. Having the same settings for both sides usually works fine.
Indicator LED: This is a handy visual element that you can use in conjunction with your ears. One LED, or a series of LEDs indicating a range of levels, may light to show that the device is actively expanding the noise downward.
When the gate begins opening, the LED may go dark, or a series of LEDs may progressively turn off as the gate opens wider. Indicator LEDs aren’t really that necessary on the expander side—they’re much more useful in compression—but they’re nice to have anyway.
Editor’s Note: This article is excerpted from Karl Coryat’s Guerrilla Home Recording—2nd Edition, available from Hal Leonard here.
Cal State University Bakersfield Upgrades Theater With Soundcraft Si Performer 3
New console used for live mixing of shows at the 485-seat Doré Theatre
California State University Bakersfield recently added a Harman’s Soundcraft Si Performer 3 digital console to solidify the technology supporting its growing music and theater programs.
The console is used for live mixing of shows at the 485-seat Doré Theatre, which include jazz and other music productions, theatrical productions, musicals, and opera. The board was purchased through Full Compass Systems.
To accommodate the 14 to 20 wireless microphones for the Music and Theatre Department productions, Performing Arts technician Frank Robinson chose the Soundcraft Si Performer 3 with 32 channels for sufficient capacity and room for future expansion. The sound booth that houses the console also allows for a simplified wiring network, and will permit elimination of much of the internal house wiring at a later date.
“We chose Soundcraft because we wanted a brand-new system within our budget that works consistently and reliably,” says Robinson. “From my decades of tour sound experience, I know that Soundcraft makes solid consoles, and the Si Performer is a good combination of intuitive design and great sound for the money.”
Situated in an educational environment, the theater features continental seating that is both wide and shallow, creating good acoustics for theatrical and musical applications. Robinson capitalized on these favorable acoustic conditions by specifying a console that is great not only in sound quality but also workflow.
“At our venue, not a lot of reproduction is needed, but we still wanted to get the perfect mix, and this board delivers on that,” he notes. “Since installation, events for which we used the Si Performer went smoothly, partly thanks to how quickly the students learned how to operate the board. Compared to some of the more expensive Yamaha consoles, this Soundcraft console sounds great and makes mixing easier, especially when it comes to patching.”
Full Compass Systems
Vision Seven Media Group In Germany Becomes A “Flagship” Audient Studio
New ASP8024 console joined by its compact little brother, an ASP4816 console, as well as several other Audient components
Vision Seven Media Group (VSM) in Neusaess, Germany is finishing up a recent upgrade that includes a range of Audient components, including a 48-channel ASP8024 console with Dual Layer Control (DLC).
The brainchild of U.S.-born producer/musician Joe Webb La Fontaine, the broad-reaching studio and multi-media complex also has an educational division, VSM Institute of Arts, which is set to develop the technical skills of young people, preparing them for work in the music industry.
In the upgrade, the new ASP8024 console is joined by its compact little brother, an ASP4816 console, as well as an ASP510 surround sound monitor controller, an ASP880 8-channel mic pre and ADC, and two iD22 USB audio interfaces. These components arrived at the same time as the facility’s new Avid Pro Tools 11 HDX and HD Native systems.
But as far as La Fontaine is concerned, analog is where it’s at. “The digital world is great and the possibilities are endless,” he notes, “but at the end of the day, we’re just trying to make it sound like analog.
“I’ve always been a user of English-type mixers,” he continues. “The EQ and filters in the ASP8024 and ASP4816 have that authentic sound: special and warm. Just a true pleasure to work with—and let‘s face it, it all comes down to the sound. I also like having my hardware effects on the board. It‘s old school and the way so many experienced engineers like to work. Plug-ins are great, and of course I also have a ton of them, but there is something special about putting your hand on a real knob and turning it, and hearing something change.”
VSM offers a comprehensive list of services ranging from recording, mixing, and mastering through artist development, video production, graphic design and promotion to touring packages, calling on a large network of professional engineers and producers to augment the VSM core team. With so many people passing through the studios, La Fontaine needed to be sure about the gear.
“We need a system that is flexible in its routing, runs 20 hours a day, seven days a week, and has the exact feature set needed - no more, no less,” he explains. “It’s got to make my job as senior engineer fun and serve as a professional tool-set for our in-house engineers and clients and for teaching and educational purposes. That’s Audient - and that’s why I choose to fill my rooms with these professional products.”
The teaching of young people is a vision that has become a reality with the development of the VSM Institute of Arts. “Basically we work as two organizations under one roof. It’s what I call “BAM” (Business As Missions/Community) concept. The commercial side is business for profit, and the missions/community side is a non-profit youth organization that has been running since 1999 here in Europe,” he says.
So with multiple projects running concurrently, La Fontaine decided to splurge on the big desk. “I chose the 48-channel version because we have a schedule that needs tools that are flexible,” he says, adding that it would be “almost impossible without a mixer like the Audient ASP8024 DLC 48.
“I also have a fixed system with DAW stems going back into analogue channels on the board with my new and vintage outboard gear fixed on the inserts. This is all possible without the need for external summing equipment. Everything happens in the ASP8024. I still love working with an inline concept mixer. With 96 faders, eight subgroups, plus the eight DLC faders – and with all the Audient products – I have finally been able to find the ideal way to get through my daily workload. Since our major studio update we have had only good comments from our commercial clients, artists and students. And we as the VSM engineering team are also super-happy.”
Thursday, November 13, 2014
Empire PRO Hosting QSC, Shure, and Yamaha Commercial Audio Training Day In Early December
Will cover how line arrays, digital consoles, monitors, amplifiers, wireless and digital microphones, and in-ear monitors operate
QSC, Shure, and Yamaha Commercial Audio will present a day of live sound training at Empire PRO, a leading pro audio, video and lighting distribution partner located in Bell, CA, on Thursday, December 4 (2014).
The training session, titled “Modern Technology For Today’s Production,” will cover how line arrays, digital consoles, monitors, amplifiers, wireless and digital microphones, and in-ear monitors operate, along with how they can optimally work together for small- to mid-size productions.
Presenters will focus on system design, networking, and the RF landscape—frequency coordination, antenna setup and best practices, special tools, and tips. A professional production of the training will also be streamed live at www.mldistrict.com/empirepro, courtesy of DataVideo.
“Our goal is to provide the pro AV community with support and resources that empower them to grow their business,” explains Edmond Khanian, vice president of sales at Empire PRO. “We’re more than a great place to get the products you need, when you need them, and at the prices you expect; we are a partner whose top priority is the success of our dealers’ business. We are proud and grateful to have strong relationships with industry leaders like QSC, Shure, and Yamaha Commercial Audio, who share and support our vision.”
• 9:40 am—10 am: Introduction
• 10 am—11:15 am: Shure, RF spectrum update/incentive auction
• 11:15 am—12:30 pm: Shure – RF coordination using Shure gear (highlighting BLX, QLXD, ULXD, AXT600 and WWB6), and how to choose the right antenna
• 12:30 pm—1:30 pm: QSC – line array specification and deployment, and amplifier technology
• 1:30 pm—2:30 pm: Lunch
• 2:30 pm—3:30 pm: QSC – Touchmix (for production)
• 3:30 pm—4:45 pm: Yamaha – CL consoles (who, what, where, when, why and how)
• 4:45 pm—6 pm: Yamaha – QL consoles (who, what, where, when, why and how)
Empire PRO is located at 5675 Mansfield Way, Bell, CA. Space is limited, so attendees should RSVP as soon as possible at www.empirepro.com/qsc-yamaha-shuretraining.php. Or tune into the live stream of the session at www.mldistrict.com/empirepro.
Yamaha Commercial Audio
Edward Jones Dome Gets A Sonic Renovation With BSS Audio, Crown Audio & JBL Professional
Dome is now the largest installation of powered JBL VerTec loudspeakers in the world
Opened in 1995, the Edward Jones Dome in St. Louis is home to the St. Louis Rams of the NFL in addition to hosting a range of other events, recently undergoing a sound reinforcement upgrade utilizing Harman’s BSS Audio, Crown Audio and JBL Professional components.
Owned by the St. Louis Regional Sports Authority and operated by the St. Louis Convention & Visitors Commission, the facility is part of the city’s convention center complex and seats up to 68,000 people.
Following completion of the sonic renovation project, the dome is now the largest installation of powered JBL VerTec loudspeakers in the world. At the heart of it are 12 arrays that each contain 12 JBL VerTec VT4889A full size line array elements with DrivePack DPDA input modules.
In addition, there are two arrays of nine and two arrays of 12 JBL VerTec VT4887A compact line array loudspeakers with DPDA modules that cover the field. All arrays utilize a top and bottom short frame and are suspended by a custom Polar Focus rigging system that allows every array to be operable for service.
The installation also features four BSS Audio BLU-800 devices, two of which are used in the control room to create new networks. Of the remaining two, one feeds all bowl loudspeakers and the other all of the under deck and back of the house fill loudspeakers.
Rounding out the installation are 25 Crown amplifiers. The South amp room, which powers underdeck and back-of-house loudspeakers is equipped with 10 DriveCore Install (DCi) 4|1250N amplifiers. The North amp room, which powers underdeck and back-of-house loudspeakers as well as an overflow press area, has 11 DCi 4|1250N amplifiers.
There are also three DCi 2|300N amplifiers in the main control room which power various booth monitors and elevators, and one Crown DCi 4I1250N amplifier, also in the main control room, which is available as an auxiliary amp for various loudspeaker needs on and around the field.
“For this project we worked very closely with Harman to identify an audio solution that would address all of the goals the owners had to improve audio performance throughout the facility in an efficient, cost-effective manner,” notes Paul Murdick, who heads the technology solutions division of TSI Global, a St. Louis-area full-service systems integrator. “With this installation we wanted to take advantage of the acoustical upgrades, which we were confident would make the audio sound better than it did before. Also, with the Sports Authority’s plans to diversify the kinds of events it hosts, it was the perfect time to upgrade the entire system rather than do a quick fix.”
Lee Buckalew, systems designer for TSI Global, notes that the amplifiers being network-enabled is particularly beneficial. “It was our first time using them, and we found them to be amazingly stable,” he says. “Their networking and error-reporting capabilities are impressive.”
Throughout the project, TSI worked closely with consultants Ross & Baruzzini, as well as John Powell, vice president of sales for JBL, Doug Nelson, accounts manager with Network Sales and Marketing, and Aaron Kunz, Harman territory sales manager. Brad Ricks and George Georgallis from Harman assisted with system tuning and Emilian Wojtowycz, also of Harman, offered pre-tuning assistance. Polar Focus provided engineering input, creating a system that provided solutions for ease of service while simplifying the seismic restraint needs, all while increasing the seismic protection offered.
In order to expedite tuning of the massive PA the team utilized JBL HiQnet Performance Manager and the system is now remotely configured and managed on a day-to-day basis using Harman HiQnet Audio Architect. “We were able to minimize our time from start to finish on this project using both Harman software platforms,” adds Buckalew.
“From start to finish, Harman was overwhelmingly supportive,” concludes Murdick. “As a project that began more than two years ago when the owners spoke to us about fixing a center cluster of loudspeakers and getting more field coverage with the audio system, Harman really demonstrated its commitment to working with systems integrators over the course of a relatively long-term installation.”
Making Waves: Adaptive Concert Sound For Imelda May Stateside
Irish rockabilly artist Imelda May has been making waves in the U.S. ever since her memorable guest appearance with guitarist Jeff Beck at the 2010 Grammy Awards.
Since then she’s been touring hard, most recently playing a string of shows backed by her 4-piece band from Providence, RI to San Francisco and numerous points between.
For May’s recent month-long U.S. tour, front of house engineer Trevor Gilligan and monitor engineer Richard McIntosh were happy to use whatever in-house gear was available at each venue.
“I’m having a lot of fun, to be honest,” says Gilligan, speaking the day after playing New York City’s Irving Plaza. “You get very complacent if you’ve taken your own desk. I’m going into these venues and some of the gear is old! It takes me back. I’ve been mixing at Irving Plaza on and off for the last 20 years. The first time I was there was with the Buzzcocks.”
Front of house engineer Trevor Gilligan at a Yamaha PM5D that served as his console on a West Coast show.
Dialing It Up
This time out, Gilligan, who’s also worked with Kasabian, The Hoosiers, and Beverly Knight among numerous other artists and has been at front of house for May since the beginning of 2011, is traveling extremely light. “I haven’t got anything apart from… actually, I didn’t even bring my headphones. I’ve been using the buds I use for my MP3 player,” he laughs.
Gilligan does have a full set of Sennheiser microphones that he’s been using for the past decade and would have normally brought them on this tour. “But with the extra security at the airports, well, you can’t lock the boxes, and Heathrow Airport is not the most secure place. They’ll steal the box or rip your case open. Though sometimes I get to a small gig and wish I had my own mics.”
Happily, the in-house systems encountered on the tour have been generally well maintained on this tour, he admits, “Deep down, you’re never fully happy unless you’ve got everything you want.” What Gilligan wants has varied over the years. He terms himself “a bit of a DiGiCo guy” in terms of console preference, but at the moment, he’s also been having a lot of success with Avid desks.
Imelda May and guitarist Darrel Higham (also her husband) performing on the current tour.
After initially balking at using the Avid VENUE when it was first introduced, instead preferring an analog console, he notes that, “Over the last couple of years I’ve noticed that every time we do a gig with an Avid, it’s always a really, really good gig.” But after finally getting to spend a few hours with the console before a show, he made some happy discoveries. “There’s one of the best compressors I’ve ever used, and it’s standard. You just need someone to point it out, or have the time to sit and mess about with it.
“They also have everything I need, and are small and compact. The consoles are so easy to use once you know them. And I’ve never had a problem loading a file. The days are gone when you have to stand behind a giant mixing desk.”
This stop also had monitor engineer Robert McIntosh on a Yamaha PM5D console.
Console options on this tour have varied, including models from Soundcraft, Yamaha PM5Ds for a couple of gigs, and McIntosh drawing a Midas PRO Series board at the previous show. For the show immediately following our conversation, Gilligan notes that McIntosh would be using a DiGiCo SD Series console and some “really great” d&b audiotechnik stage monitors.
“So it’s from one extreme to the other,” he says. “But you just go in and dial the stuff up. All of the venue guys have been great, and everything’s been going really smoothly.”
Basic & Effective
More likely to be found at front of house, McIntosh got the call from May’s camp three years ago and has worked monitors with her whenever she comes to the U.S.
“I kept telling them I’m not a monitor engineer, but they just won’t listen. I guess they just like me. It’s barrels of fun; they’re incredible,” says the veteran engineer who also has a long working relationship with Bobby Caldwell and has toured with John Tesh, Colin Hay, Ambrosia, and Jennifer Love Hewitt.
McIntosh prefers mixing on an Avid VENUE Profile or SC48 because he already has the show files saved and at the ready, easy to load and tweak. It’s a matter of keeping things simple. However, he adds that he’d never before had the opportunity to work on a DiGiCo board until the day we talked. “It’s a great console, and it may soon become my favorite if I keep getting them,” he notes.
May and her band, husband Darrel Higham on guitar, Dave Priseman on trumpet, flugelhorn, and guitar, Al Gare on electric and acoustic bass, and drummer Steve Rushton, are all on wedges. “They like stuff old school, even what they want in their mixes,” McIntosh explains. “There’s not much cross-mixing. They want to hear themselves, and the drummer is the only other one who gets Imelda. They’ve just got it down, working acoustically; so of course, I’m delighted.”
May is supplied with an inside and outside pair of wedges. In an approach devised by Gilligan, the inside set is delayed just a touch so that the output of all four reach her simultaneously. There’s no reverb or compression applied to the signal.
A single mic is applied to guitar and bass amps. Here it’s a Sennheiser e 609.
“It’s basic, but effective,” McIntosh says. “Imelda likes some power and sometimes that’s a little challenging, depending on which mic we’re using.” This time out the vocal mic choice was a Shure SM58, in place of the more usual Sennheiser e965.
“It’s much easier for me to dial up a 58,” he says. “I think Trevor likes the [e 965] because it’s a little more ‘hi-fi’ for him, but it’s a little more challenging for me.”
The number of lines from stage is typically around two dozen, plus or minus a few. “On this tour,” Gilligan points out, “Darrel’s not singing and he’s just got one mic in front of his amp. We haven’t got a slap pickup on the bass – all Al’s got is one line for his [acoustic bass] and one line for his electric, direct from the instrument.” The same goes for Higham’s guitar amp.
On this night, the mic set for Steve Rushton’s kit includes a Shure SM57 on snare, Sennheiser e 604s on toms, another SM57 as well as a Shure Beta 52A on kick, Shure SM137s under the cymbals, and dual Sennheiser MD 421s for overhead.
Horn player Priseman prefers a beyerdynamic M88 hypercardioid dynamic whenever one is available, preferring the punchy sound it delivers in his wedges. On the kit, Rushton’s cymbals are miked from below: “He plays the cymbals really nicely, so I mic him quite close, sometimes within eight to 12 inches of the cymbal,” Gilligan says. “If I’m really lucky and have a good digital desk, then I’ll put a bit of compression on them for when he whacks the hell out of them.”
He adds, “I’ve been getting a lot of success, believe it or not, putting an SM57 inside the kick drum. The 57 has a bit of extra bottom end; it’s really good for a little dynamic mic. Steve doesn’t have a big rock sound, so I put it reasonably close to the skin and then put the other mic, whether it be a 52 (Shure Beta 52A) or whatever, just inside the hole.
“If I’ve got a digital console I might put them both in place, and then on the out mic, start messing with the timing. You can get almost an out-of-phase sound. If I’m in these little venues, flipping the phase on one of them does the trick. You can get some nice results.”
The 4-piece band backing up Imelda May on the tour included Darrell Higham on guitar, Steve Rushton on drums, Dave Priseman on trumpet, and Al Gare on bass.
The acoustic bass proved a challenge, largely due to the original instrument getting crushed in an unfortunate forklift incident early in the tour. After that, the quality of instrument each time out proved to be hit and miss. “Some of them sounded like planks of wood,” he notes. “The worst ones were the pickups—you get more body noise than bass.”
Traveling light makes for sound economic judgment on a tour of this scale, even when touring Europe, and both engineers met the challenges adapting on a show-to-show basis.
“I always try and find a desk that’s fully compact without racks of stuff, just a plug-and-go system,” concludes Gilligan, who typically works with Capital Sound in England. “If we can get it down to one truck instead of two, it saves us a massive amount of money. I’ve managed to find certain systems—like the Martin Audio MLA Mini—that don’t take up a lot of space and sound fantastic.
“It just makes more sense nowadays, because of the way the business is going. Everything’s getting cut back at certain levels, unless you’re massively successful.”
Based in Los Angeles, Steve Harvey is a long-time pro audio journalist and photographer.