Article

Wednesday, February 08, 2012

Allen & Heath iLive For Revitalized Islington Assembly Hall In London

The reopened Islington Assembly Hall in London has selected an Allen & Heath iLive digital mixing system, comprising iDR-48 and iDR-32 MixRacks with two iLive-T112 Control Surfaces, to manage both front of house and monitors.

Originally opened in 1930, the fully refurbished 800-capacity Hall was reopened in 2010 having fallen into disuse for nearly 30 years. The venue now hosts a busy rota of live music, as well as a vast range of events from conferences and parties to vintage fashion events.

“We chose iLive for several reasons,” explains Dan Turner, events operations manager at Islington Assembly Hall. “First, it sounds great and much better than many other digital desks out there. Allen & Heath clearly spent a lot of time getting the preamps right.

“It’s also an incredibly versatile system in the way it works, and pretty much any input or output can be configured to do what you need it to do. I have mainly used analog desks in the past, and iLive almost feels like it is one as the design is more hands-on than menu driven. This helps you get to grips with the beast a lot more quickly than with other digital desks where features can be hidden in layered menus and thus almost useless in the live environment.”

The venue refurbishment plans did not include the installation of a mixing desk, and as such, no multi-core was installed.

“The beauty of this digital system is that to increase the number of available channels all we need to do is use a larger mix rack, there’s no need to dig up the floor to install a cable. Having the mix rack on stage and the control surface at the rear of the hall connected by a single Cat-5 cable was a big selling point,” continues Turner. “We needed an elegant solution to stage monitoring using as few cables and as little space as possible. Using two iLive systems connected together we have achieved this. All signals are routed to where they need to go without the need for active splits on the floor saving valuable space”

The venue also installed Electro-Voice XLE181 line arrays with additional front fills and subs, and Martin Audio monitors on stage.

“Overall, the system sounds fantastic and will enable us to meet all the challenges that our varied events calendar will throw at us!,” concludes Turner.

Allen & Heath

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Virtual Sound Checks Without A High-End Digital Console

Here are a few ways to get it done

 
Here are some thoughts on doing virtual sound check if you don’t have a DiGiCo or Avid digital console at your disposal.

Disclaimer: This is not going to be exhaustive. There are hundreds of hardware/software combinations that will get you the same result. These are some ideas only.

Also, it should be noted that “cheap” is a relative term. All of these solutions are going to cost money, real money.

However, if you church is serious about raising the level of audio technician performance, it’s money well spent. On we go…

First, let’s define “virtual sound check.” It is simply the ability to record the band with each channel on it’s own track and then being able to play that recording back, in place through the same channels on your console.

To illustrate with a very primitive example, let’s say your “band” is a worship leader with an acoustic guitar. To facilitate virtual sound check, you would need a way to record the vocals and guitar on separate tracks, and you want those sources to come off the board before any EQ or dynamics.

Typically, you’re using direct outputs or the insert outputs. When you get ready to practice, you do a little patching (in software or hardware) and play back that recording through the same channels you use if the worship leader and his guitar were live in the room.

One thing should be immediately apparent here; the bigger your band (and the more sources you have), the more elaborate the system you’re going to need for virtual sound check. If you are running 30-40 inputs every weekend, this post is really not for you as that system is not going to be cheap.

Rather, I’m focusing on those who run fewer than 24 channels per weekend (a number that is not arbitrary, as you’ll see in a minute) and using an analog board. Here are a few ways to get it done.

Audio Interface(s)
The simplest way of doing this job is with a USB or more likely a FireWire interface such as the M-Audio ProFire 2626, a Focusrite Saffire Pro 40 or similar interface with 8 analog inputs and 8 analog outputs.

The first thing you’ll notice when shopping for an interface is that manufacturers get very creative in the way they count I/O. For example, the ProFire 2626 is listed as having 26 inputs and 26 outputs, which it does. But only 8 of them are analog.

And if you’re using an analog console, that’s all you care about. If you have a digital console with ADAT I/O, you gain you an additional set of 8 useable channels.

Now, the catch here is that there aren’t any interfaces with more than 8 channels of analog I/O (at least I can’t find any). So that means if you’re running 12 channels of audio, 4 get left behind. Unless you get creative. You might ask why you can’t just connect two 8-channel interfaces to your computer and send those inputs to your recording software.

The issue is that most DAW software won’t support multiple I/O devices simultaneously. If your DAW of choice doesn’t support multiple I/O devices, there is a workaround, at least on the Mac.

In Audio/MIDI settings, you can create what’s called an Aggregate Device, which allows you to create a virtual device that is made up of two or more actual devices. You then chose the Aggregate Device as your I/O source in your DAW, and all the inputs and outputs on all devices that make up the Aggregate Device are available to the DAW.

So an example system might be made up of two Focusrite Saffire Pro 40 interfaces combined into an aggregate device and recorded using Reaper on a Mac Mini. That would give you 16 channels of recording and playback for around $1500, give or take. That seems pretty reasonable; at least until you consider the next option.

Hard Disk-Based Recorders
There exist on the market a couple of hard drive-based recorders, most notably the Alesis HD24. This little 3-rack-space wonder is capable of recording or playing back 24 tracks of 48 hHz, 24-bit audio.

The HD24 has 24 channels of analog I/O (plus 24 channels of ADAT I/O) and costs about $1600. Really, this is the way to go. It requires no computer, is simple to set up and operate and is rock-solid reliable. Add 24 channels of TRS patch cables and you’re done.

Other options include the Tascam X-48, which is a full-blown 24 channel workstation (and almost $5,000) and the excellent, but somewhat pricey JoeCo BlackBox, which will set you back almost $3,000 by the time you add a drive.

Caveats
There are a few caveats with any of these solutions. First, if your board has direct outputs, it’s a fairly simple matter to patch those direct outs to the inputs of whatever recording solution you use.

Getting back in, however, will require some re-patching. You’ll want to pull your mic inputs, and patch the outputs from the recorder or interface(s) into the Line Inputs on your console.

If you don’t have direct outs, you’ll need to use the inserts. One cool thing about the JoeCo BlackBox is that the inputs are normaled back out to the outputs during every operation except playback.

That means that for recording (or just sitting there), the insert signal is returned and you can continue to use the board normally. When you hit “Play,” it opens the normal and sends the recorded signal back to the return on the board. From a user interface standpoint, that’s really nice. However, it will cost you twice what an HD24 costs…

When using the inserts, you will likely need to push the cables into the console until the first click. An insert jack is a TRS (tip, ring, sleeve) connector, so it has 3 contact points. Most consoles use the ring as the send, so if you push a TS cable in to the first click, you get the equivalent of a direct out (albeit an unbalanced one). Pushing it in all the way will interrupt the signal, so you’ll only do that on playback.

Using inserts is going to mean a fair amount of patching and some experimenting, so don’t decide to try this out at 8:50 on Sunday morning.

Once you get the system up and running like you want, start recording your services in all their multi-track glory. Then during the week, you can practice and experiment just like the band is there, only they aren’t.

Keep in mind, you won’t have any acoustic energy coming from the stage, so things like drums and vocals will be a little different. But this is still a great tool for training and experimenting with various processor settings.

Like I said, this isn’t exhaustive; I only intended to give a few examples. Hopefully though, it will get you thinking about how you can implement a virtual sound check system in your church.

 

Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.

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Tannoy Unveils VLS Series Passive Column Array Loudspeakers

Tannoy has introduced the VLS Series passive column array loudspeakers offering a balance of performance and cost, when active beam-steering may neither be required nor affordable.

The VLS Series is the first Tannoy product to incorporate FAST (Focused Asymmetrical Shaping Technology), which delivers unique acoustic performance benefits. Central to this is its asymmetrical vertical dispersion, gently shaping the acoustic coverage towards the lower quadrant of the vertical axis. By the nature of a typical application, an “ideal” column loudspeaker should be biased in the vertical plane, toward the audience and away from reflective surfaces above (like ceilings) which are detrimental to intelligibility.

FAST also facilitates quicker, easier installation with less need for tilting or specific concern for optimal mounting height. Mounting is handled via supplied wall brackets.

Tannoy has packaged this performance in a slender and narrow profile, aesthetically refined, powder-coated aluminum chassis with curvilinear aluminum grille. Each model is available in either black or white as standard, with custom RAL finishes available at additional cost and lead-time.

Three models are available – VLS 7 (7 × 3.5-inch LF) designed for speech-only applications, VLS 15 (7 × 3.5-inch LF with 8 × 1-inch HF) and VLS 30 (14 × 3.5-inch LF and 16 × 1-inch HF), both of which are designed for more demanding full-range applications as well as speech.

All are IP64 rated for dust and water ingress and are salt spray and UV resistant as well as subject to rigorous high/low operational temperature and humidity testing.

Specification is aided by the addition of an exclusive Tannoy edition of EASE Focus v2.0 software, allowing systems to be designed with predictable results, along with the ability to specify VLS Series in conjunction with Tannoy’s existing column loudspeakers – including I Series and QFlex.

Tannoy

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How To Archive Multitrack DAW Recordings

The archived recordings must be prepared to weather obsolescence

 
Multitrack DAW recordings are dependent on a complex system of primary and secondary technologies.

As discussed in An Introduction to Archiving Music Recordings, each of these technologies represents an obstacle to the long-term viability of a multitrack archive.

Simply put, if the various software and hardware products you’re using today aren’t going to be around in their current versions for the useful life of the sound recordings you’re creating (i.e. the copyright term), the archived recording must be prepared to weather that obsolescence.

The goal of preparing multitrack DAW data for archive is to minimize the layers of technology necessary to completely reconstruct the master recording in the future.

This article will introduce some basic techniques for creating both Consolidated and Flat Multitracks for archival purposes.

What Is A Consolidated Multitrack?

A Consolidated Multitrack is a digital audio fileset that completely expresses the EDL (Edit Decision List) information from a multitrack master recording. Specifically:

—Each DAW track is expressed as a single, continuous Broadcast Wave file (BWF);
—All of the consolidated audio files share the same start times and durations;
—All of the consolidated audio files share the same digital audio precisions, i.e. sample rate and bit depth;
—All of the consolidated audio files share the same descriptive naming convention, e.g. trackname_songtitle_artistname.wav.

If all of the above specifications are met, a folder containing the consolidated audio files could be used to perfectly reconstruct the multitrack recording as far into the future as the Broadcast Wave file format remains viable.

Since the Broadcast Wave file is a widely accepted standard file format for media producers, its long-term viability (and eventual uniform migration) is virtually guaranteed.

Creating a Consolidated Multitrack:

1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Consolidated Multitrack.
2. Hide or delete any auxiliary signal path to simplify the working environment.
3. If additional Takes or Playlists are to be included in the Consolidated Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. Using session boundaries, location markers, or some other timeline tool, establish a repeatable global timeline selection that includes all audio from the earliest drop-in to beyond the longest running audio file.
5. Once your global selection is made, use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track.
6. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_ohbabybaby_jimmysingsalot.wav

Once the above steps have been followed, a choice has to be made about how to present these consolidated audio files as a discrete multitrack recording for archive.

Minimally, a folder that follows the same naming convention as the consolidated audio files should be created to contain all of the associated audio files and metadata (like screen shots, rtf files containing session notes, credits, etc.). This method works fine, but will always require the multitrack to be reconstructed in a DAW for playback.

Alternately, a facility like Pro Tools’ ‘Save Session Copy’ could be used to create a new, independent playback session for only the archival material.

Using this method one would need to be careful to remove any non-archival audio and metadata from the source session before saving the copy.

This approach would facilitate more convenient short-term use of the archive, but doesn’t actually provide any additional content.

What Is A Flat Multitrack?

A Flat Multitrack is a digital audio fileset that completely expresses the EDL information from a multitrack master recording, but also expresses some subset of DAW metadata. What metadata is ‘flattened’ into the archive is up to you, your client, or contractual obligations, but it could include:

—Plug-in processing like amp simulation, ‘printed’ effects from auxiliary channels, or automated processing;
—Automation data, like the fader rides on a lead vocal track;
—Bounced submixes that would otherwise require reconstructing both complex routing and plugin processing.

It is critically important to note that a Flat Multitrack should never be archived instead of a Consolidated Multitrack, but only in addition. The Consolidated Multitrack is the master recording; the Flat Multitrack (when applicable) is an extension of that master.

Once a Consolidated Multitrack has been created, a Flat Multitrack can be created by repeating the process with a few additional steps:

1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Flat Multitrack.
2. Hide or delete all auxiliary signal path and metadata that is not going to be flattened.
3. If additional Takes or Playlists are to be included in the Flat Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. To flatten real-time processes like automation, time-based effects, or submixing, bounce/re-record the appropriate track outputs to new tracks, and remove the source tracks from the session. Note what metadata has been flattened.
5. Flatten additional metadata by processing audio files with offline versions of real-time plug-ins. Note what metadata has been flattened.
6. Make a global timeline selection, and use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track (including whatever metadata has been flattened into them).
7. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_flatcompression_ohbabybaby_jimmysingsalot.wav

Since it would be unlikely that every track within a DAW project would have metadata worth flattening, there will likely be some tracks that remain in their consolidated form. I would caution that it would be both redundant and confusing to include these audio files in a Flat Multitrack archive.

Preferably, an additional folder of flattened audio files can be clearly labeled, and organized with the Consolidated Multitrack data. Future users can then reconstruct the Consolidated archive, and opt-in to any of the available, clearly labeled, flat content.

Contents Versus Carrier

It should be noted that this tutorial only addresses the form of the contents of a multitrack archive. The question of how to effectively store this information is an entirely additional- though related- matter.

Anybody who is serious about the subject should examine the Producer and Engineers Wings’ “Recommendation for Delivery of Recorded Music Projects” (pdf). It contains an example of a widely-adopted approach to redundant archival storage.

Rob Schlette is chief mastering engineer and owner of Anthem Mastering (anthemmastering.com) in St. Louis, MO, which provides trusted specialized mastering services to music clients across North America.

Be sure to visit the Pro Audio Files for more great recording content.

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Hosa Technology Mogan Elite Omni Earset Microphone Now Shipping

Hosa Technology has announced that the new Mogan Elite omni earset microphone, the newest addition to the Mogan brand of subminiature microphones, is now shipping.

The Mogan Elite earset mic is outfitted with a moisture-resistant, 2.5 mm omni-directional capsule with -45 dB nominal sensitivity that is designed to be positioned farther from the user’s mouth.

Delivering full-frequency audio performance (20 Hz – 20 kHz) and high gain before feedback, this microphone provides a natural, resonant sound quality.

The new mic also offers an innovative earpiece designed to be worn comfortably for extended periods. With a fleshy ear cushion concealing its fully adjustable, sprung-steel (stainless) mechanism, the mic feels natural when worn over one’s left or right ear.

An interchangeable cable system allows connection of the mic to most popular wireless transmitters, including models from Shure, AKG, Sennheiser, and Audio Technica. Each mic ships with a detachable, Kevlar-reinforced cable with a hardwired connector.

The new Mogan Elite Omni Earset Microphone is available in either beige or black to blend with a variety of skin tones. Additionally, each unit includes a foam windscreen and a single mic clip. The entire package ships in an impact-resistant, compression-molded neoprene zippered case.

Jonathan Pusey, Hosa Technology director of sales and marketing, states, “The Mogan Elite earset microphone delivers impressive performance, enabling this microphone to be right at home in a number of high-end applications, including broadcast and theater, in which audio performance is critical. In addition to world-class audio quality, the Elite earset mic is very comfortable to wear and may be worn without distraction for hours on end.

“The earpiece is highly adjustable and the boom mechanism facilitates precise positioning of the microphone’s capsule. I am quite confident the Elite earset will be right at home in a number of demanding audio environments.”

MSRP for the Mogan Elite omni earset mic is $400.

Hosa Technology

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Engineer/Producer Matthew Noble Utilizes Metric Halo ChannelStrip On Recent Projects

For more than three decades, Matthew Noble has been at the forefront of pop music as a session guitarist, programmer, songwriter, engineer, and producer, with an engineering client list that includes Rihanna, Diana King, Southside Johnny, and Rod Stewart, among many.

These days, he performs most of his work out of the Loft Studios in Bronxville, NY and in the newly renovated Riverworks Recording in Dobbs Ferry, NY. Recent work with the musical Big River and gospel artist Rell Holland & Experience have put Noble’s new favorite plug-in, the Metric Halo ChannelStrip, through its paces.

“I tried Metric Halo’s ChannelStrip because some other people that I respect were using it,” explains Noble. “My friend Keith Brown, who is a well-known Nashville songwriter, was working on a project with Billie Decker, who is one of the hottest mix engineers in country music. Keith’s enthusiasm for the plug-in, together with his revelation that Billie uses it ‘all over the place,’ was enough to motivate me to check it out.”

Riverworks Recording boasts a huge, luscious acoustical space, which has changed the way both Noble and the producers and artists he works with approach the recording process.

“So much of my work there has involved tracking live instruments, as opposed to the ‘virtual players’ that live inside our modern computers,” he says. “While it’s been a refreshing change, it has also brought with it challenges. For example, getting a great drum sound and a great overall mix with the new expectations for how long things take these days is not easy.

“ChannelStrip has been very helpful because all the functions that I need to access quickly are all in one plug-in. These include the less ‘sexy’ functions, such as phase reverse and multiple trims, in addition to full-blown and flexible dynamics and equalization. Having everything in one plug-in has greatly improved my workflow.”

Noble often puts Metric Halo’s well-crafted presets to use: “The ChannelStrip presets are a great starting point. They’re especially useful in a time crunch, when the client is breathing down your neck. The acoustic guitar and drum presets are often spot on, right out of the gate. When I tweak, the informative GUI lets me know exactly what I’m doing.”

Of course, the best GUI in the world is useless if the algorithms behind it don’t cut the mustard. It’s here that Noble finds it really shines. “ChannelStrip has a great sound,” he said. “Like an SSL, it can be very aggressive and not at all subtle. Despite all its flexibility and sonic muscle, it has remarkably low CPU drain, which means I can use it whenever I need it.”

Metric Halo

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Enter The PSW Sweepstakes To Win An Audio-Technica Microphone Or Headphones

Enter to win an Audio-Technica microphone or headphones in the first PSW Sweepstakes of 2012.

ProSoundWeb is giving away three Audio-Technica 50th Anniversary Limited Edition products each month in January, February and March.

Specifically, for each drawing, we’re giving away:

1st prize - AT4050/LE Multi-Pattern Condenser Microphone
—Special 50th anniversary edition in silver-colored metallic finish with etched-on serial number and carefully crafted wooden carrying case
—Transparent uppers/mids balanced by rich low-end qualities combine with advanced acoustic engineering for extensive performance capabilities and highest quality
—Dual-diaphragm capsule design maintains precise polar pattern definition across the full frequency range of the microphone
—The 2-micron-thick, vapor-deposited gold diaphragms undergo a five-step aging process so that the optimum characteristics achieved remain constant over years of use
—Three switchable polar patterns: omni, cardioid, figure-of-eight
—Transformerless circuitry virtually eliminates low-frequency distortion and provides superior correlation of high-speed transients
—State-of-the-art surface-mount electronics ensure compliance with A-T’s stringent consistency and reliability standards
—Switchable 80 Hz hi-pass filter and 10 dB pad
—Custom shock mount provides superior isolation
—Valued at $995.

2nd prize - ATM25/LE Hypercardioid Dynamic Instrument Microphone
—Exclusive 50th anniversary edition in silver-colored metallic finish with serial number etched on the surface
—Ideal for kick drum, toms, and other highly dynamic instruments
—Handles very high SPL at close range
—Big, warm low-frequency response with excellent presence
—Multi-level grille and rugged construction
—Offers very full sound on close-up vocals and dialogue
—Corrosion-resistant contacts from gold-plated XLRM-type connector
—Rugged, all-metal design and construction for years of trouble-free use
—Valued at $489

3rd prize - ATH-M50s/LE Professional Studio Monitor Headphones
—Special 50th anniversary edition in silver-colored metallic finish
—Exceptional audio quality for professional monitoring and mixing
—Collapsible design ideal for easy portability and convenient storage
—Proprietary 45 mm large-aperture drivers with neodymium magnet systems
—Closed-back cushioned earcup design creates an outstanding seal for maximum isolation
—Adjustable padded headband for comfort during long mixing/recording sessions
—Single-sided straight cable terminates to gold-plated mini-plug with screw-on 1/4-inch adapter
—Valued at $209

Go here to enter the latest PSW Sweepstakes. Note that entrants are asked to register to receive the ProSoundWeb Daily e-newsletter.

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Tuesday, February 07, 2012

Rat Sound Deploys L-Acoustics KARAi At Orange County’s Galaxy Theatre

Orange County’s 40-year-old Galaxy Theatre changed ownership back in August and has now undergone a significant metamorphosis into two separate live performance spaces: the intimate 350-capacity Constellation Room and much larger Observatory, which can accommodate an audience of over 1,000.

One of the primary improvements made to the venue in the process was the installation of L-Acoustics KARAi line source arrays provided by Certified Provider Rat Sound Systems of Camarillo, California.

Jon Reiser, along with business partner Courtney Michaelis and a third silent partner, are the team behind transforming the Galaxy from a lackluster aging venue into an edgy hipster hangout that consistently attracts some of the brightest up-and-coming acts like Foster the People, Crystal Castles, The Naked and Famous, Young the Giant, Tyga and Warpaint. Reiser is no stranger to SoCal’s indie music scene having previously spent five years as a partner and talent buyer for Costa Mesa’s Detroit Bar.

Shortly after purchasing the Galaxy, Reiser brought in L.A.‘s Foster the People to play the newly dubbed Observatory. Knowing that the club’s pre-existing PA system wouldn’t be adequate to cover the crowd for such a popular band, he turned to Rat Sound to provide a temporary dV-DOSC rig while simultaneously initiating the design process to permanently integrate a new KARAi system.

Today, the Observatory features left and right arrays each comprised of six KARAi elements flown adjacent to two SB18i subs. Four SB28 subs are also located down on the floor—two housed in bunkers inside the stage plus one on each side of the stage.

A custom enclosure built across the front of the stage houses four coaxial 8XT front-fill speakers, while a single 115XT HiQ flown in the center of the house two feet downstage of the stage lip serves as a downfill. All systems are powered and processed by LA8 amplified controllers housed in racks at the monitor mix position.

Rat Sound provided not only the installation of the L-Acoustics arrays, but the rest of the venue’s sound and lighting systems and acoustic treatments as a full turnkey package.

Rat Sound director of installations David Myers notes that the rental dV-DOSC system helped minimize the venue’s downtime to only two nights despite the significant amount of remodeling and equipment upgrades.

“With the new acoustic treatments and KARAi system in place, the Observatory sounds phenomenal,” says Myers. “KARAi is extremely coherent; it’s like having the artist right in front of you in a studio environment. And the bands, engineers, promoters and audiences are all loving the sound. The Naked and Famous recently played the room and their front-of-house engineer told us after the show that we had totally spoiled him for the rest of the dates on their tour.”

Reiser adds, “The Naked and Famous immediately booked another show with us for March and I know that a big part of that was because of how impressed they were with the system and production level. We’ve had a lot of really great shows in here lately—including Scott Weiland, two nights with Young the Giant, and an epic New Year’s weekend with ATB and Tiesto—and everyone’s walked away at the end of the night being very happy with the house sound.”

L-Acoustics

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Revolabs Enhances HD Control Panel For Entire HD Line Of Wireless Microphone Systems

Revolabs has announced that the company’s Windows-based HD Control Panel software has been enhanced to support the entire HD line of wireless microphone systems, bringing the monitoring and configuration tools found on the Executive HD to the HD Single/Dual Channel and the HD Venue systems.

In addition, based upon customer feedback, Revolabs has created several new features for the HD line, including a DIP switch display, mute groups for Executive HD systems, and an expanded control system API.

With the HD Control Panel, users can monitor and control networked HD wireless microphone systems from a single PC software program with an intuitive graphical user interface.

The HD Control Panel allows users to control the mute status and gain of each microphone, and to lock out presenters from using the mute button.

The software also provides the ability to monitor each microphone closely for its real-time status, such as battery level.

The monitor tab of the HD Control Panel has been enhanced to provide the DIP switch status for each system, eliminating the need to look on the back of the system to see which switches are active.

Revolabs has also added several commands to the HD systems’ API, allowing A/V control systems to send global commands, turn off microphones, and even initiate pairing, all from the convenience of a room’s touch panel.

Finally, Revolabs has bolstered the Executive HD with the ability to assign systems to mute groups. This allows all systems in a building to be bussed together without muting each other, unless they are assigned to the same group.

“We are pleased to bring the capabilities of the HD Control Panel to users of our HD Single/Dual Channel and HD Venue systems, in addition to offering powerful new features across our entire HD line,” says JP Carney, CEO of Revolabs. “We take pride in listening to our customers as we continually strive to meet their evolving needs. New features, such as those released today, are a direct result of customer feedback.”

The enhanced HD Control Panel and new features are available through a firmware update (version 2.6.1) to both the base station and microphones. The update is available now at www.revolabs.com/downloads.

New feature enhancements require a Gold unlock code provided as part of a Revolabs service plan. Any system that has previously been unlocked will automatically receive the new features upon completion of the firmware upgrade.. 

Revolabs

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In The Studio: Three Mid-Side Processing Tricks (Includes Audio Samples)

A form of processing on stereo sources for practical or creative effects

 
In this article I’ll explain how I use mid-side (MS) processing on stereo sources for practical or creative effects.

Mid-Side?
Two channels of audio can be combined in a way that gives us control over what is the same in each signal, the middle, and what is different, the sides.

The middle is where the kick drum, snare, bass, vocals and a lot of other instruments are, the sides have any hard-panned instruments and spatial effects like reverb.

It can be pretty interesting to listen to music like this, there can be a lot hidden in the side channel.

MS is also a stereo microphone technique using a cardioid microphone facing the source and a bidirectional mic turned 90 degrees away just picking up ambience.

In this situation the two signals would need to be decoded into stereo. The side mic signal is duplicated, polarity inverted and the two side signals are then panned hard left and right. This is not a true stereo mic technique but can sound very nice. The balance of mid and side signals can be adjusted as needed by changing the level of the three tracks.

You can manually encode and decode stereo files to MS and use mono plugins to process mid or side individually. A lot more plugins have an MS mode now. Many of the modules in the T-Racks suite allow mid side processing, as does Ozone, a few compressors and equalizers and a distortion also come to mind.

You can do this for subtle or crazy effects, its a fun way to experiment with plugins and get some unique sounds.

Loud & Wide
For a recent mastering job I used a Fairchild compressor plugin in MS mode (Lat/Vert) to compress the middle and increase the level of the sides. I did this in parallel so I could blend the effect in easily. I was also using this to get a lot of extra loudness. You can call this parallel MS compression.

Compare -

The master without parallel MS compression: listen

With parallel MS compression: listen

With parallel compression soloed: listen

No More Messy Verb
Someone asked ma about clearing up the middle of a mix when using a lot of reverb. Using MS compression on the reverb return can work well. Compress the middle more than the sides and increase the side volume if you want more width.

Here is an example of that on some drums - Steven Slate playing in KONTAKT. The whole kit is sent into Valhalla Room. With the Fairchild after the reverb I’m lowering the middle by 2 dB and raising the sides by 2 dB.

Here is this effect with lots of reverb on the drums: listen

And now with MS compression on just the reverb bus: listen

There is NO compression on the drums themselves, I’m only compressing the reverb return and widening it.

Wacky Effects
Here is an example of what you can do with a stereo loop and any plug-in. This is a little more complicated, and only works if there are hard panned sounds.

The loop I started with had a hi-hat that wasn’t panned very hard -  I copied it to a new track, filtered out all the lows, boosted some highs and then panned it hard left. Then I recorded the combined original and panned track to a new file.

Here is what I’m starting with: listen

Now that I had something on the sides I could mess around with MS processing.

The first thing you have to do is convert left-right to mid and side. I use the free +matrix MS decoder from SoundHack.com. After that I used a delay plugin to add some filtered echoes just to the middle by disabling the right side input.

In the next insert I used a distortion on just the right side. This brought out a lot more of the reverb than was heard in the original loop. Lastly, second MS decoder was used to bring it back to stereo.

Here is how the loop sounds now with delay in the middle and distortion on the sides: listen

Pretty cool right!? I hope you have found these tricks useful.

Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.

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Fishman Unveils Triple Play Wireless Guitar Controller

The new Fishman Triple Play Wireless Guitar Controller combines guitar with any virtual instrument or hardware synthesizer to access a wide range of instruments, samples and sounds on stage to expand the depth and impact of live performances.

Triple Play comes with a wireless controller, hexaphonic pickup, and wireless USB receiver. The controller and included software works with industry standard DAWs and vitual instruments and installs quickly on any electric guitar. The system can be easily removed from the guitar because it doesn’t require any permanent installation.

The Triple Play system features several “hold” functions such as sustain, looping, and arpeggiators, along with string or fret splits for multiple instruments.

Also included are menu navigation controls for the included software and a guitar synthesizer volume control. A guitar, mix, synth switch is easily accessible during performances.

A low profile design (less than .5-inch) allows the controller to be left on the guitar and still fit in the case. It operates with a rechargeable Lithium Ion battery (included).

Triple Play’s powered USB wireless receiver interfaces with computers or iOS devices. The system comes with a comprehensive Windows, OSX and iOS software bundle to get users started.

A Triple Play Wireless Guitar Expander option provides additional connectivity for interfacing wireless MIDI signals to computers or iOS devices. It adds a full function USB audio interface with guitar input, bypass and headphone output, MIDI hardware IN and OUT and support for footswitches to extend Triple Play’s capabilities for recording, performing or composing music.

The new Triple Play Wireless Guitar Controller is scheduled for release in June 2012.

Fishman

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Church Sound: How To Transition From Analog To Digital Mixing

A digital mixer is a whole new way of doing the same old things

 
I’m in the process of helping one of my churches transition from an analog mixer to a digital mixer.

They were in need of more channels than their Allen & Heath 16-channel MixWiz with some outboard gear (front of house EQ, couple of compressors, effects unit) could provide.

Based on the maximum number of channels that they anticipated needing over the next five years, I recommended the PreSonus StudioLive 24.4, one of the least expensive 24-channel digital mixers on the market.

The church has two audio volunteers that are pretty much average in their knowledge of sound and sound systems so this would be a typical transition for a lot of churches in the 100-400 person attendance range. Volunteers selected more for their willingness to serve than their knowledge of audio. I know that nothing has been touched with the front of house EQ, compressors and FX since I helped them set it up about a year ago.

Some things that you need to consider in this transition is how uncomfortable the volunteers are going to be until they make the paradigm switch from the analog WYSIWYG (what you see is what you get) to the digital layers.

Depending on the digital board, layers control everything from different grouping of faders (1-8, 9-16, etc) to control over the aux sends, FX, etc. Outboard gear usually goes away and everything is now handled with the digital mixer. It’s a big transition and you shouldn’t minimize it, but treat it with care and planning and the transition will go smoothly.

Getting Started
What I recommend is that the digital mixer not be put into service immediately but be brought into a two-to-four-week training duty cycle. It requires some mics and cables as well as a couple of speakers for monitors and front of house stand-ins. If you have instruments that you can plug in that helps as well. Keep the existing analog system going as the production system until everyone has been trained and is comfortable with the digital board.

Before you start with the digital mixer, make sure everyone has reviewed the user manual. A digital board is a computer with knobs and faders and is significantly more complex than an analog mixer. While they are pretty robust, you can still mess them up and repairs can be costly.

An Investment of Protection
One thing to invest in if you haven’t is a top-line power conditioner like those from Furman. I also recommend a computer UPS (battery backup) from a company like APC or Tripp Lite. Get a decent capacity one. The reason is that because a digital mixer is a computer, when power is interrupted you can’t just switch it back on like an analog mixer. You need to boot it up and, depending on the mixer, that could take anywhere from a minute to several minutes.

Having a UPS unit, the mixer will stay powered on, so even if the rest of the system is knocked offline by the power interruption, when the power comes back on, the mixer will still be up.

Unboxing The Mixer
Once you get the mixer unboxed, check for any damage. If everything looks good bring all faders down to minimum and turn on the mixer. I like to let the mixer “burn in” for about four hours with nothing going on or plugged in just to let all the electronics warm up to full operating temperature. This will check to ensure that nothing is shorting out. Be aware of any burning electrical smell or smoke. If you detect either one shut the mixer down immediately and unplug it. Contact the vendor.

Preparing For Training
The StudioLive is close to an analog board in that all the channel faders are on one surface as opposed to layers. This makes the transition somewhat easier. All effects, aux send levels are controlled through the center “Fat Channel.” That will be where most of the confusion is going to come in so be prepared to spend a lot of time going through this area.

The StudioLive is set up pretty easy so I was able to figure 85% of the board out without looking at the manual. There are also a ton of video tutorials on the PreSonus site and YouTube that can help with anything to do with the board. But for volunteer sound techs it will be a bit of a challenge.

Building A Mini-System
Hook up a mic to channel 1 on the mixer and hook up a speaker to aux send 1 and to front of house. This will be the basic training setup.

Once you get it hooked up, bring up the gain to an appropriate level. A digital board is less forgiving about exceeding the 0 level than an analog board before going into clipping so run the level less than needed for training until you get comfortable with the way the board handles signals.

Don’t worry about EQ settings or FX yet. All you want to do is to learn the signal flow from the channel to the aux send and FOH.

Once you’ve figured out how to adjust the aux send levels for the channel and you can adjust FOH level you’ve gotten over the initial hump.

Using EQ
The next thing you’ll want to learn is how to adjust EQ’s for each channel. Depending on the digital mixer you’ll either have a screen that will have a parametric equalizer, or in the case of the StudioLive, you’ll have the knob adjustments for high, high mid, low mid and low bands. As with all digital mixers you are able to set the frequency points for all these bands as well as the Q, which is the width of the frequency adjustment. This is a lot more adjustability than what an analog mixer has and is worth spending some time practicing.

After the channel EQs get figured out you’ll want to adjust the front of house EQ. On the StudioLive it’s set the same way that the individual channel EQs are set. One nice advantage about digital mixers is that most of them have a library of preset EQs that you can start with. The StudioLive has built in a nice set of professional quality EQ presets that are good enough to leave alone and assign to each channel.

The other nice feature of digital boards is the ability to save all your settings to a scene. So you are able to set up multiple scenes for different worship teams or different instruments and recall them just by dialing up the scene and pressing the load button. So no more needing to reserve channels based on who’s playing that day.

Enter Effects
The power of digital mixers means that you can assign FX to each and every channel, both to auxes and to front of house, so you’ve got a lot of flexibility. Just remember that just because you can doesn’t mean you should. Less is more, at least in the beginning. Some boards give you more FX capabilities than others. The StudioLive offers two channels of FX, others more.

Multi-track Recording
Another advantage that digital mixers have is that they usually provide some form of multi-track recording capability. In the case of the StudioLive, it’s provided by a FireWire port into the provided Studio One software. This means you can record each channel separately into your computer, as long as it has a Firewire port.

One very cool reason for doing this for the worship team is the ability to do what’s called a Virtual Sound Check. What that means is that you don’t need the worship team there to set up the board. You can play back the individual tracks back into their respective board channels and use those tracks as the sound check.

Then, once the band gets in, sound check is very minimal. It’s also a great way for the sound team to train on the board and allows them to massage settings without needing the musicians.

Saving Scenes
Once you get everything set the way you want it remember to save your settings to a scene. I usually recommend naming the scene with the church name and 1. That way you can always recover your baseline settings.

Sound techs should create their own “sandbox” scene, which allows them to manipulate settings and save it to their own scene without affecting the master scene. Make sure that no one other than the lead sound tech saves to the master scene.

Once you’ve got the master scene saved it won’t matter what changes people make to the board during the week. Bringing back the master scene will only require a quick push of a button, and in the case of the StudioLive, resetting the gain and adjusting the faders. In other digital boards, gain settings and fader positions are saved within the scene.

Making The Switch
Once the sound techs are comfortable with the digital board then it’s time to switch out the old analog board with the new digital one. Check all your settings. Be sure any settings you change are saved to the master scene once you’re happy with how everything sounds.

Finally, when you shut things down, do NOT shut things down by just turning off the power conditioner. This WILL damage the digital mixer. Follow the shutdown procedure in the manual. It can be anything from just powering off the mixer with the mixer’s power switch to a shut-down procedure on the screen.

Summary
A digital mixer is a whole new way of doing the same old things. It’s exciting as well as terrifying for volunteers, so go slow. Take it one step at a time and ensure they are comfortable with the new system before putting it into production. You’ll achieve a seamless transition and have fun doing it!

Brian Gowing has helped over 30 churches meet their technology requirements. Brian works towards shepherding the church, analyzing their technical requirements, sourcing the equipment, installing the equipment and training the volunteer personnel.  As he likes to say, “equipping the saints with technology to help spread the Good News.” Contact Brian here.

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Joe Peavey And Steve Spittle Join QSC Audio

QSC Audio Products has announced the addition of two new members to its professional team, with the appointments of Joe Peavey to the position of product manager, software and Steve Spittle to the position of business development manager.

Peavey will be working with the Q-Sys team to identify and define improvements and additions to Q-Sys software functionality as well as providing high-level technical support. He has a lengthy background in the installed sound market by his work with the family business, Peavey Electronics, specifically working in manufacturing, tech support and finally product manager of the MediaMatrix line of DSP products.

Since leaving Peavey Electronics in 2006, Peavey has focused on creating hardware and software solutions for various audio manufacturers and consulting services for integrators in the U.S. and Canada.

“In the many months since my first interactions with the company, QSC continually amazes me with their attention to the market, their workforce and quality,” says Peavey. “I am proud to join forces with an organization of their caliber and reputation on a product at the top of its game.”

Spittle, in his new role at business development manager, will focus on expanding opportunities for growth in the company’s integrated systems business. He was previously western U.S. sales manager at Avid, and a vice president/owner at Millar Electronics, a manufacturers’ rep firm located in the southeastern U.S.

“QSC makes great products and cares about its customers,” he says. “I’m looking forward to working with this dynamic team to continue to build on this foundation for growth.”

Spittle is located in QSC’s Costa Mesa headquarters, while Peavey is located in the company’s satellite offices in Boulder, CO.

QSC Audio

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Mojave Audio Debuts MA-101SP Matched Pair Cardioid Condenser Microphones

Mojave Audio has introduced the new MA-101SP, a matched pair of MA-101fet cardioid condenser microphones for use in a range of stereo recording and live sound reinforcement tasks, with instruments such as drums and guitar amplifiers, as well as capturing room ambience and general stereo recording.

Each MA-101fet in the matched pair provides warm, full-bodied reproductions of instruments without the shrillness and high frequency artifacts so often encountered with modern condenser microphones.

The microphone’s warm FET circuitry and externally polarized capacitor mic element combine to deliver low noise and high quality performance.

The MA-101fet features both omni and cardioid polar patterns by way of interchangeable capsules and is outfitted with a 3-micron thick, .8-inch diameter gold sputtered diaphragm.

As one would expect from a David Royer designed microphone, each MA-101fet in the MA-101SP matched pair offer performance specifications that are impressive. Frequency response is 20 Hz - 20 kHz (+/- 3 dB), sensitivity rating is -40 dB (1 volt per pascal), and the distortion rating is less than 1 percent @ 120 dB SPL (-15 db pad off) and less than 1 percent @ 135 dB SPL (-15 dB pad on). The microphones operate on standard 48-volt Phantom power.

Mojave Audio president Dusty Wakeman states, “The new MA-101SP matched pair of microphones is the result of countless requests from the audio community. Drawing upon the strengths of the MA-101fet, these mics are a terrific choice for a wide range of stereo recording tasks where imaging is critical. 

“Engaging the 15 dB pad allows one to take advantage of the fast transient response on instruments such as snares, toms and loud guitar amps. The MA-101SP is a remarkably versatile general purpose recording and sound reinforcement tool that, I’m confident, will find a home in a wide variety of environments.”

The new Mojave Audio MA-101SP ships in a single carrying case that includes a stereo bar. MSRP is $1,195, and availability is Q1, 2012.

Mojave Audio

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Monday, February 06, 2012

Meyer Sound Promotes Miguel Lourtie To European Technical Services Manager

Meyer Sound has announced Miguel Lourtie as its new European technical services manager, where he will supervise the company’s technical support team in Europe and assume primary responsibility for sales support and design services in the region.

“Customer support is paramount at Meyer Sound,” says John Monitto, Meyer Sound’s director of technical support worldwide. “Our customers expect an extremely high level of technical expertise and customer service. With his outstanding technical skills, customer rapport, experience in the field, and fluency in several languages, Miguel is a great fit to lead our technical group in Europe.”

Lourtie joined Meyer Sound European technical services in 2007, and has played a vital role in supporting a number of major Meyer Sound projects across the continent, including the Mantziusgården Culture Center, Montreux Jazz Festival, and the Grimaldi Forum. He also serves as a seminar instructor as part of Meyer Sound’s extensive education program.

Prior to joining Meyer Sound, Lourtie founded Lourisom, an audio consulting and distribution business in Portugal and previously a Meyer Sound distributor.

“To ensure a seamless show, high-quality audio tools and the person driving the system are equally crucial,” says Lourtie. “The Meyer Sound tech support network has some of the best sound engineers in the industry, and I look forward to working even more closely with them to help our customers get the best out of their Meyer Sound equipment.”

Lourtie will continue to be based in Lisbon, Portugal.

Meyer Sound

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