Friday, December 14, 2012

Argosy Console Introduces New Mirage Workstation For Yamaha/Steinberg NUAGE

Custom designed for new integrated audio production system

Argosy Consoles has announced the premiere of the new Mirage For NUAGE, a new workstation custom-designed specifically for Yamaha and Steinberg’s recently introduced NUAGE integrated audio production system.

NUAGE offers Yamaha control and interface hardware as well as Steinberg Nuendo DAW software working together to offer productivity flexibility and audio quality for post-production applications.

The system’s control surface and audio interface hardware are modular, with all units communicating with each other and the central computer via network, making it easy to create custom configurations to match a variety of applications.

The Argosy Mirage workstation for NUAGE is specifically built to create a landscape to enhance this production creativity and flexibility.

The workstation is available in two models, the Mirage NUAGE-32LS-B-B in standard black and the Mirage NUAGE-32LS-B-M with mahogany hardwood trim.

Both models feature a sturdy tubular steel frame and powder-coated steel legs and body panels, concealed cable trunking and padded armrests in front of each of the racks.

Both Mirage Models include two integrated rack modules with space for six RU in each.

Optional accessories including monitor arms, speaker platforms and blank panels are also available to further customize Mirage for NUAGE.

“It has been very exciting for Argosy to work so closely with a company such as Yamaha, and early in the development cycle of NUAGE, so that we could build a customized Mirage workstation to fully complement this exciting new system,” says Argosy vice president Tim Thompson. “We are thrilled that Argosy can play a part in this offering of a total post-production system solutions package.”

Both Mirage for NUAGE models are available now for immediate shipping.

Argosy Consoles

Posted by Keith Clark on 12/14 at 02:57 PM
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Thursday, December 13, 2012

In The Studio: What I Learned About Mixing In 2012

A year full of learning experiences, both professionally and personally
This article is provided by Home Studio Corner.


2012 has been full of learning experiences for me, both personally and professionally.

Usually when someone uses the phrase “learning experience,” they actually mean “painful experience.”

It’s so true, right?

Any time I learn a powerful lesson, there’s almost always some sort of discomfort that comes along with it.

Take mixing, for example. I love to mix, but there are days where I would rather swim with hungry alligators than work on a mix.

Sometimes mixing comes easy. Sometimes it’s painful. But if you can persevere, there’s always something to learn, something that will make your future mixes better.

Here are some of the things I learned.

Give Your Mix a “Time Out”

No matter how awesome you think your mix is, resist the urge to share it with the world until you’ve given it a time out.

Time-outs work like gangbusters in the Gilder house. Our two-year-old Owen can completely lose it some days. Sticking him in a “time-out” in the corner for two minutes usually does the trick.

When it comes to mixing, one of the best things you can do for your mix is to simply step away. After a couple hours of mixing, your ears have accustomed themselves to what they’re hearing. If your mix has some blatantly huge issues, you might not even notice them.

THAT’S why you need to give your mix a time-out. It’s misbehaving, and you don’t even know it, because you’re too close to the situation.

Come back to it the next day, or maybe take a lunch break. When you come back with fresh ears, you’ll instantly hear those two or three issues in the mix that need to be addressed.

Now you can address them before you send the mix off to be heard by others.

Trust me on this one. Your ears are tricky little creatures. Make ‘em be honest with you before you do your final bounce.

Level vs EQ
This one’s so easy I regularly forget about it.

If you’re fighting with a particular track in a mix, and you just can’t seem to make it sound right, what’s your first instinct?

I can tell you what mine is. I instantly reach for an EQ knob and start twisting. If it’s too harsh or too boomy, then I obviously need to cut some frequencies in the low mids or upper mids, right?


But what I’ve found is that when I find myself using very aggressive EQ, chances are there’s another more obvious problem at hand.

What problem?

The track is simply at the wrong volume.

Almost every time I find myself really wrestling with a particular track, using crazy EQ cuts, sooner or later I realize that the track is simply too loud or too quiet.

I’ll remove all the EQ crazyness and move the fader up or down a few dB.

The result? It sounds instantly better.

Don’t forget how powerful a simple fader move can be.

The Lousy Speaker

This is one of those tips the music stores don’t want you to think about. :>)

Yes, it’s important to have good, accurate studio monitors and/or headphones. However, I’ve found (especially this past year) that my crappy little 3-inch loudspeakers are insanely helpful mixing tools.

People always ask me, “What crappy speakers do you recommend?”

That’s a hilarious question. Just use something crappy. Something with a small speaker on it. I’ve got some old Roland speakers with a 3-inch woofer. They’re not full range, and they’re not good for making really detailed decisions.

However, they INSTANTLY show me major flaws in my mix. Like if the vocal is too loud, or if the bass it too muddy. Or maybe the guitars are too aggressive or the snare drum disappears.

As an added bonus, I only use one of these speakers, and I send a mono signal to it. So now it’s my crap speaker AND my mono speaker.

I’ve found that if I can get the mix to sound killer on this speaker, it will sound killer in my car, and it will still sound great on my nice monitors, too.

It’s not always fun to hear your mix on a crap speaker, but it can be insanely useful.

Dare to Compare

Comparisons can be painful.

You might think your mix is amazing, but then you hear it next to a professional mix, and you’re instantly sad. The pro mix sounds so much better than yours.

Yes, it hurts (and yes, I’ve been there many a time). But it’s good for you to take these long honest “looks” at your mixes. If your mix doesn’t stand up to a professional mix, then you probably have some more work to do.

It’s difficult and frustrating, but it makes you better.

Compare your mixes to professional ones, and slowly but surely your mixes will start sounding more and more professional.

The Mix is Slave to the Recording
This is probably the most important thing you could learn. (And it’s been something I’ve run into over and over again this year.)

Your mix is slave to the recording. If the recording sounds horrible, your mix will sound horrible.

It’s so easy to have the “I’ll just fix it in the mix” mentality, but you simply CAN’T do it. You can enhance a recording. You can even make it sound better. But your mixes won’t be amazing if you’re not putting as much focus and effort on the recording side as you do the mixing side.

You must…you simply MUST…put in huge amounts of effort to capture great-sounding performances. This means rescheduling a session for another day if the performer isn’t “feeling it.” It means re-recording tracks that simply don’t sound good. It means spending lots of time working out the arrangement before you lay down a bunch of tracks. It means taking the time to audition a few mic placements before you start recording.

The better your recordings sound, the better your mixes will sound. Period.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.

Posted by Keith Clark on 12/13 at 05:11 PM
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Apogee Releases MiC Carrying Case

Nylon case features precision interior cutouts for a snug, perfect fit for MiC, tripod, cables and more

Apogee Electronics announces the MiC Carrying Case which is designed to precisely store and protect the MiC and its accessories. 

Utilizing durable materials, this 7.25”x4.25”x2” case offers ultimate protection with a sleek and elegant design.  With this portable storage case, on-the-go musicians, voice-over artists, and podcasters always have access to their MiC no matter when or where inspiration strikes.

“We wanted to offer our consumers a customized storage option for the MiC,” says Apogee CEO and Co-Founder Betty Bennett.  “The MiC is very important in the daily lives of many of our consumers, and it’s critical that the device is protected well.  We carefully designed this case to offer MiC users peace of mind when traveling.”

A hard nylon exterior and a padded interior combine structure and safety for the MiC.  On one side, precision cutouts snugly protect the MiC, tripod stand, and MiC stand adapter.  This ensures that the delicate pieces will not rub or bump against each other in transit.  Additionally, the soft all-purpose protective foam will not leave behind any scratches on the metal.  The other side of the MiC carrying case is a mesh pocket perfect for storing cables or other small items.

The Apogee MiC, released in February 2012, is the first studio-quality microphone to make a direct digital connection to iPad, iPhone and Mac.  Roughly the size of an iPhone, MiC makes it easy to record any voice or acoustic instrument anytime, anywhere.  It features PureDIGITAL technology which provides a no-noise signal allowing the user to capture the true tone of the voice or instrument.  The MiC and carrying case are sold separately.

The MiC Carrying Case is available in black.  It retails for $19.95.

Apogee also offer the MiC Pro Accessory Kit which includes the Carrying Case, Microphone Stand Adapter, 3-meter Mac cable, and a 3-meter iPad/iPhone Cable.  It retails for $49.95.

Apogee Electronics

Posted by Keith Clark on 12/13 at 01:53 PM

Synchro Arts Releases Updated Revoice Pro (“Pro2”) Editing Program

New version adds manual editing of time and pitch, as well as fostering faster work flows

Synchro Arts, creators of VocALign audio alignment software, has announced the release of Revoice Pro 2, the first major update of the original audio processing program.

Revoice Pro has been designed for professional audio editors and provides:

• Easy-to-use tools for manipulating audio features (timing, pitch, vibrato, inflection and level) with precision and quality.
• Unique automation of audio feature editing tasks to save time and enhance results.

Specifically, this update includes:

• Manual Time, Pitch and Level editing of the APT output
• VST 3 and Audio Unit Revoice Pro Link plug-ins
• Improved quality of alignment and audio processing
• Numerous time saving features and faster work flows

Revoice Pro is a purpose-built stand-alone program which includes two unique automated processes:

1. Audio Performance Transfer (APT) process, which automatically transfers selected timing, pitch, vibrato, inflection and/or loudness characteristics of a good “guide” audio signal to one or more target audio signals (“dub”). APT is powered by and includes an advanced version of VocALign.

2. Realistic Doubler, which instantly creates natural-sounding mono or stereo double tracks from one input signal. It can also provide creative time and pitch modulation effects.

ReVoice Pro also provides adjustable control of the tightness of the time, pitch and loudness corrections, so editors can get the exact effect they want—whether getting the tightest ADR or vocal replacement—or thicker but clearer and punchier vocals.

Revoice Pro 2 has added the ability to make manual adjustments to the pitch, timing and level of APT putput signal with easy to use controls.

The program integrates in several ways with all professional Mac-based digital audio workstations and includes Audio Suite, VST3 and AU Link plug-ins plus a Rewire interface for efficient monitoring options and synchronized playback.

Applications include:

• Tightening the timing, pitch and vibrato of “stacked” lead and backing vocals or instrumental tracks.
• Creating one or more realistic double tracks from a single input tracks.
• Lip-syncing dialogue (ADR) and vocals by the same or different performers, even when there are noisy guide tracks.
• Changing the inflection in dialogue (ADR, voice-overs etc.) with the desired guide pattern provided by recording the director or dialogue editor.

Revoice Pro accomplishes these tasks automatically.

Main Features:

• APT that includes process-based audition, providing instant comparison of before and after processing of one or two signals, as well as protected regions, which avoids creating artifacts by not processing the time, pitch or both in small, user-defined areas of signals where guide and dub differ.
• Doubler that processes mono or stereo input and creates mono or stereo outputs.
• Stand-alone program that works with all professional Mac-based editors including Pro Tools, Cubase, Nuendo, Logic, etc.
• Multiple methods for efficient audio transfers – including drag and drop, plug-ins, and export tool, all able to place output at correct time.
• Includes rewire for extra monitoring options and in-place testing.
• Multiple view windows, multiple open sessions.
• High-quality multitrack scrubbing.
• Manual waveform and pitch editing options.

Added Features in Revoice Pro 2:

• Manually adjustable timing, pitch, vibrato, and level with intuitive controls. (Useful for preparing Harmony tracks to be guide tracks without needing external editor.)
• New APT process controls can restrict audio from being moved too far, allowing better alignment of signals containing multiple short “bursts” of audio that match a guide.
• More manual pitch editing functions to group, split, join pitch groups and blocks. Smooth Pitch command for removing discontinuities and jumps in pitch trace.
• Improved quality of audio processing.


• NEW AU (Audio Unit) and VST3 Revoice Pro Link plug-ins so Logic, Cubase, Nuendo and other (Mac) DAWs that use AU and VST3 plug ins can now transfer audio from one or more DAW tracks simultaneously to Revoice Pro.
• APT and Doubler Process control blocks can be instantly copied with all settings to other signals by drag and drop, saving the user set-up time.
• Tracks and waveforms can also be copied by drag and drop to creates copies (with auto incremented name).
• Individual channels of a multi-channel file can be displayed for detailed examination
• Can create default session templates
• Can drag and drop processed audio as a copy into DAWs or as “bounce” to Revoice Pro tracks.
• Can duplicate processes by dragging

A 14-day free trial license (iLok-based) for Revoice Pro can be obtained from, along with downloads of the Revoice Pro program, online manuals, demos and tutorial videos.

Full licenses (iLok-based) can be purchased from Synchro Arts’ dealers or on-line from

Recommended retail price of Revoice Pro:

US$799 for North America
£499 (ex VAT) for UK and the rest of the world
€590 (ex VAT) for Europe

Introductory Offer (Ends March 31, 2013):

• 25 percent discount for new Synchro Arts customers
• Further discounts on trade-ins for current VocALign pwners

(Note: Product pricing, features, specifications, system requirements, and availability are subject to change without notice.)

Synchro Arts


Posted by Keith Clark on 12/13 at 11:10 AM
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Tuesday, December 11, 2012

Radial Engineering Introduces The SixPack 500 Series Power Rack

Accommodates all 500 series modules, including older ones made by API

Radial Engineering has introduced the SixPack, the latest Workhorse power rack designed for the 500 series module format.

The SixPack is Radial’s sixth power rack in its growing 500 series range, joining 15 modules that are currently available and a clear indication of the company’s proactive stance with respect to building and supporting the 500 series standard.

According to Radial president Peter Janis, “Given today’s ever changing digital environment, recording studios must be agile for them to adapt and they must also bring a higher level of creativity and uniqueness to the recording process. We believe that the 500 series is perfectly suited for this new reality as it is compact and easily transportable.

“And when you consider the hundreds of modules that are now available plus the tremendous patching options that can be put to use, there has never been a more exciting time to be involved in music production,” he adds.

The SixPack is a 6-slot power rack designed to accommodate all 500 series modules, including older ones made by API.

It provides a hefty 1600 milliamps of current for more than 265 milliamps average power per slot. This allows the mixing and matching of solid state and tube modules without concern about powering.

The external supply also provides 48V phantom power for mic preamps. Protective circuitry in each slot safeguards the SixPack and other connected modules against malfunction or short circuits.

The design begins with 14 gauge steel construction for added durability and improved shielding against noise. 

Modules may be patched in series using the FEED switch and stereo mated using the LINK function. 

Extra connectivity includes 1/4-inch TRS connectors wired in parallel with the XLRs for cross-patching and parallel processing. These are also wired in parallel to a set of D-Sub connectors for easy patching to and from the workstation. 

Two front-panel XLR convenience jacks provide access to channels 7 and 8 on the D-Sub or may be assigned to channels 1 and 4 to create a stereo channel strip.

Separate ground lugs on the rear panel enable the system tech to incorporate star grounding or other schemes into the studio design. 

Optional mounting hardware fosters recessing the SixPack into the workstation or angle mount it for a more ergonomic set up. The same hardware works double duty to enable the SixPack to be mounted into a standard 19-inch rack.

The SixPack will start shipping January 2013.

Radial Engineering

Posted by Keith Clark on 12/11 at 03:29 PM
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Dominic Harter Returns To Harman To Head Up Soundcraft Global Sales

Originally joined Harman in 1998, primarily to manage the supply of Harman equipment to the Millennium Dome project

Dominic Harter has rejoined Harman Professional to head up Soundcraft’s global sales operation, following the promotion of Adrian Curtis to VP sales of the Harman Professional EMEA sales team.

Harter originally joined Harman in 1998, primarily to manage the systems supply of the vast quantity of Harman equipment to the Millennium Dome project, working with all brands on all aspects of the project.

With the completion of the project, he became sales manager with BSS Audio and C Audio, using his technical knowledge and sales skills to support the distributors and products.

Harter has spent the last 10 years with Turbosound, where he was director of R&D before becoming sales director.

“It’s great to be back at Harman with so many great people and old friends,” Harter reports. “Soundcraft was the first brand I ever worked for as a student, and I am relishing this opportunity to steer the brand through the next phases of the digital console market development, and work with all the regional Harman Pro sales teams around the world.

“It is a pleasure to be part of such a fantastic brand.”

Harman Professional

Posted by Keith Clark on 12/11 at 12:42 PM
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CAD Audio Introduces “Sessions” MH510 Professional Headphones

Offers extended lows, smooth mids and articulate, life-like highs

CAD Audio has introduced new Sessions MH510 professional headphones, designed for recording and live audio environments.

MH510 headphones produce a wide frequency response (10 Hz – 24 kHz) with extended lows, smooth mids and articulate, life-like highs for accurate and natural reproduction.

High SPL capability delivers ample volume while the design provides exceptional isolation ensuring a private listening experience that eliminates bleed through into the playback environment.

MH510 headphones are available in a distinct and modern cosmetic design with four colors––black, white/red, back chrome and black/orange to choose from.

Each headphone is supplied with two cables (coiled and straight) and two sets of earpads.

CAD executive director of business development Glenn Roop notes, “Given CAD Audio’s acclaimed studio heritage, developing the MH510 headphones was a match made in heaven. The Sessions project was a natural for our design team. We engineered for the highest level of performance with no compromises, and then added some flair and style into the mix.”

MSRP for the MH510 headphone is $159. They will be on display at the upcoming Winter NAMM 2013 show at the Anaheim (CA) Convention Center, booth 6632.

CAD Audio

Posted by Keith Clark on 12/11 at 12:01 PM
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In The Studio: Five Keys To Executing A Fast Mix

The better I’ve become at mixing quickly, the better I’ve become at mixing in general
This article is provided by the Pro Audio Files.


This topic is an “eat your vegetables” one.

Maybe not as fun as how to mix flangers and phasers to create alien noises, but I feel it’s an important topic for any professional or aspiring professional.

At some point in your career you’ll be expected to mix a record very fast. In all likelihood this will in fact become somewhat of a common occurrence.

So here are some tips to help you get good results fast.

Incidentally, I’ve found that the better I’ve become at mixing quickly, the better I’ve become at mixing in general.

1) Know your tools

People tend to get a little gear crazy. Having options and the right tools is important, but knowing how to use them is more important. It’s important to really learn the character and capabilities of every piece of hardware and software you use.

Having this relationship with your equipment allows you to access what you are trying to do without having to experiment or guess.

2) Take time to organize

In an effort to be quick, we may short change things like labeling audio channels or aux sends. In reality, spending a little extra time up front to make sure everything is color coordinated, labeled, and in an order and format that is easy for you to navigate makes the actual mix process much faster.

So even if you’re under the gun, take a little extra time upfront and save yourself time and frustration down the road.

3) Commit early

If you’re using hardware inserts, CPU heavy plugins, or just things that glitch or could change easily — reprint early on. If you need to tweak things down the road, you can always pop another EQ or compressor on there. Committing early keeps your head clear and keeps you moving forward. It also frees up your hardware and CPU.

I also like to do basic automation early on. If something is obvious, just give it a quick fader move.

4) Listen to the rough mix with a notepad ready

The basic trading of lead instruments, places for delay throws, the elements that are important and where — these ideas should pop out pretty early on. If you listen to the rough mix and take notes on these things, when it comes time to mix you won’t need to constantly reference the rough mix in order to get your sense of direction.

5) Tweak and move. Find your “anchors” and your “leads”

Set up your reverb and delay throws, and your aux routing, then get to moving. Generally there are rhythmic “anchors” in a song: drums, bass, whatever is pinning the groove down. Throw that up first. Do basic EQ or compression if necessary to get that element sounding good on its own.

Start filling in the rest of the rhythmic elements weighing them against your anchor(s). If you keep the word “groove” as your focus, you will put your rhythm section together in no time.

Then do your other instrumentation. Keep doing little tweaks as you go — nothing major unless it really really needs it. For lead elements, like vocals, guitar solos, whatever, do those last.

The tendency is for the last elements to come in the loudest — so if you put your leads in early, you might end up with an organ pad or string section up too loud. Which means you’ll then have to re-balance, so bring those leads in last. If you’ve left enough headroom, it shouldn’t be an issue to get them to sit just right.

Now, if you have more time to mix, you may want to bring in your lead first and build everything around it. But for something quick I find bringing the leads in last moves faster.

This one is more about personal process — works for me — might be different for you.

Matthew Weiss is the head engineer for Studio E, located in Philadelphia. Recent credits include Ronnie Spector, Uri Caine, Royce Da 5’9” and Philadelphia Slick.

Be sure to visit the Pro Audio Files for more great recording content. To comment or ask questions about this article go here.

Posted by Keith Clark on 12/11 at 11:40 AM
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DigiTech Now Shipping iStomp With 10 Downloadable E-Pedals & Updated Stomp Shop App

App allows auditioning of any pedal for free

DigiTech is now shipping its iStomp Programmable Pedal, pre-loaded with 10 of its most popular e-pedals and now features an updated Stomp Shop app where guitarists are able to try any pedal for free for up to five minutes.

“When we introduced the iStomp we led a new product category that changed the way guitarists and musicians thought about creating their tone,”  explains Scott Klimt, marketing manager for DigiTech. “With each new e-pedal introduction we have expanded their capabilities and brought a new level of utility and functionality never before available.

“The addition of eight more e-pedals included with the Stomp Shop update underscores our commitment to continuing to improve our offerings in the downloadable pedal category.”

The 10 e-pedals that will be included upon purchase now include:
—Redline Overdrive
—Total Recall Delay
—Blue Pearl Chorus
— Death Metal
— DOD FX25B Envelope Filter
—Continuum Reverb
— Jet Flanger
— Compressor
— Octaver
— Double Cross Delay

The updated Stomp Shop app provides faster image and sound clip downloading for demoing new e-pedals, new store front themes giving users the ability to customize their online interface, and new clearer settings that reset the LED color on the iStomp for each new e-pedal.

The iStomp pedal, which easily connects to an Apple iOS device using the DigiTech Smart Cable that comes with the iStomp, offers numerous options when testing unfamiliar sounds.

With four knobs to control effects parameters that change their function according to which effect is loaded, every musician can create his or her signature sound.

The DigiTech iStomp sells at a suggested retail of $229.95 and comes with an iOS authorization cable and power supply, with e-pedals available from $4.99 to $19.99.


Posted by Keith Clark on 12/11 at 10:43 AM
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Monday, December 10, 2012

In The Studio: A Tale Of A Project-Saving Monitoring Technique

A "wrong" hookup turns out to be invaluable in helping to overcome a singer's difficulties with hearing tracks via headphones

As guitarist Jeff Baxter once said to me during the “heat” of a Rod Stewart recording session: “We can do anything—the impossible just takes a little longer.” Along with talented musicians like Jeff, engineers, producers, and live mixers often are called upon to solve problems that are at first glance, flat impossible.

To a technically challenged client, all the flashy and complicated gear, computers and the alacrity at which a pro uses them to produce nearly instant, seemingly magical results (I think) hypnotizes or lulls people into a state of “anything is possible”—even though what they want defies the basic laws of physics!

One such situation occurred to me a while back when I was in Sydney Australia recording an album with an R&B band called The Rockmelons. (A rockmelon is Aussie for a cantaloupe if you were wondering) Australia is a wonderful and mystical place especially out in the middle of the country—a U.S.-sized desert.

So perhaps the laws of physics are suspended in parts of the “down under” but they were not for us at EMI 301 Studios in downtown Sydney.

The lead singer in the band, at that time, had the worst case of “red light” fever I’ve ever encountered—actually more like a severe headphone phobia. As soon as he heard the track and his voice in the cans, he acted intimidated and overwhelmed; he would stop, not sing at all, or sing terribly.

All of us were puzzled because at live gigs in front of an audience, he was wonderful—the main attraction. The most peculiar thing was that if we suddenly stopped the track’s playback, for a few measures he would sing the most spectacular soul riffs and melodies all acapella.

Robin Smith, the producer was obviously extremely concerned because if this guy couldn’t sing, we would not have an album—they were not an instrumental group.

The first flash of “can do” brilliance came from Smith when he had the second engineer setup a two-track tape deck so that it recorded the same audio feed from my vocal recording chain at the session’s multi-track received. This machine was to be kept it in record, rolling at all times.

The second engineer and the producer also worked out a system of hand signal routines—a rating system where the assistant would jot down the two-track’s tape counter number and a 1 to 5 rating whenever the producer heard a piece of a vocal he liked. These chunks of vocals might be used in the final vocal compilation process.

So for the rest of the day, whenever I would stop playback, we captured all these cool riffs and lyrics on the two-track. Later, after the singer left, the producer and I would “fly” in all the good bits into the multi-track master vocal take. To say the least, this was not a satisfactory record production method that grew very old very fast.

So after a couple of days when the singer did not get any better, we decided to tackled the problem at the root cause: what was it about the phones that put this otherwise great singer off? We played with volume; mix, compressing the phone mix, reverb and other effects, different brands of headphones—everything we could think of.

We determined that the singer was a sensitive fellow who just felt physically uncomfortable and a little paranoid wearing headphones—and it was worst when music was coming out of them.

So I remembered a trick I saw Crosby, Stills and Nash used ‘back in the day’ at a Hollywood studio called Sound Labs. I was working on another project in studio 1 and they were banging away in studio 2.

Of course everybody in those days would occasionally check out “who was in the other room” and what it sounded like. It was a great learning atmosphere with a very rich and free exchange of remarkable ideas from very talented people—artists, engineers and musicians who sometimes were not that technical but always open to any sonic experiment no matter how ludicrous-sounding.

I went in and saw they were using two Yamaha NS10ms or Auratone monitor speakers mounted on mic stands instead of headphones!

I think it was Graham Nash who said they had always harmonized as a group around a single microphone listening to each other more than the track. I went into the studio during a playback to hear how low in volume the speakers were. They were very quiet.

But the big revelation to me was when I stood equal-distant between the two speakers and discovered they were flipped in polarity—out of phase from one another.

Apparently I (being a recording engineer trying to be vigilant for such serious problems) was much more sensitive to this than CSN. It was also true that none of them ever stood exactly equidistant between the speakers.

Further, this “wrong” hookup was anathema to everything I was ever taught or had experienced. But this was situation where practicality outweighs technical correctness.

So I used that idea for my singer in Australia and it worked! The main point of this trick is to place the vocal mic exactly—dead on—in the null-point of the two speakers. I did this by playing the cue mix and listening to the mic channel in solo.

I put headphones to hear only the mic’s signal and would just move the mic around until I got minimum speaker spill or leakage.

Is the leakage a problem? No, not unless you want to do some wacky dance remix where the lead vocal is solo’d or you produce and record an entirely new track under the vocal.

The other ‘detail’ is to make sure you play only the most minimal track mix out on the speakers and try to keep it mostly monaural. You might have to play with panning positions etc. Play just enough elements of your track’s production for the singer to sing well.

I’d probably leave out most of the sweetening ideas, fancy percussion playing and vocal effects off. I’d also try to keep the bottom end not too big and the Yamahas will help in the regard.

Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Be sure to visit his website

Posted by admin on 12/10 at 07:00 PM

Friday, December 07, 2012

Etabeta Selects Clear-Com For Flexivan OB Project

Clear-Com is pleased to announce that Etabeta, the Rome-based outside broadcasting (OB) division of Italy’s Gruppo Comunicazione Italia, has selected its Eclipse-Omega digital matrix intercom for the Etabeta FlexiVan OB vehicle.

Clear-Com is pleased to announce that Etabeta, the Rome-based outside broadcasting (OB) division of Italy’s Gruppo Comunicazione Italia, has selected its Eclipse-Omega digital matrix intercom for the Etabeta FlexiVan OB vehicle.

The Eclipse matrix system and V-Series intercom panels will be used to coordinate complex broadcast productions, including sports, talk shows and conventions. The installation was specified and facilitated by Video Progetti, Clear-Com’s broadcast distributor in Italy.

As a leading production company for more than 25 years, Etabeta has created some of the most technically advanced television programs for such top networks as RAI and Mediaset.

As part of its commitment to providing quality productions to the global market, Etabeta develops programs of almost every style. Its latest OB vehicle, FlexiVan, serves as a broadcast production hub and connects multiple broadcasters’ OB vans.  FlexiVan also comes with a full ENG crew to accommodate events of any size at any location in the world.

The 240-port Eclipse-Omega system is the communications backbone of the FlexiVan operations, allowing tight collaboration among engineers and studio staff to efficiently manage all production activities. Eclipse seamlessly interfaces with third-party systems as well as telephone lines to provide more user connections.

The advanced V-Series key-panels are used among operators for fast audio routing, and offer sophisticated features, such as 10-character OLED display labels and a ‘Listen Again’ digital memory replay. Moreover, any system changes can be quickly achieved through the Eclipse Configuration Software (ECS) so that the FlexiVan team can easily adapt to last-minute production changes.

“Due to the many requirements of the FlexiVan, we carefully evaluated all possible intercom solutions on the market,” says Rosario Messina, OB Van Engineering Manager, Etabeta. “We found that the Clear-Com product line meshes perfectly with our needs for flexibility since it offered several communication paths, intelligent integration with other systems and scalability for each production.

“Also, because of Etabeta and Video Progetti’s long-running relationship with Clear-Com, we know the company provides high-quality products as well as excellent customer support.”

“The Eclipse-Omega digital intercom system with V-Series panels is the perfect system for a busy broadcaster like Etabeta,” says Karlie Miles, Director of EMEA Sales, Clear-Com. “Broadcast production teams do not necessarily know what new demands are around the corner, but Clear-Com’s intercom solution gives them the ability to adjust to and fulfill the demands of any production.

“We are thrilled that Etabeta has entrusted Clear-Com to handle communications for its FlexiVan OB vehicle, and we’re excited to continue helping the company far into the future.”


Posted by Keith Clark on 12/07 at 12:39 PM

Biamp Systems Releases Report On The Hazardous Levels Of Noise In Everyday Life

New report highlights the impact of noise at work, at school or at home

The amounts of sustained noise people are subjected to in everyday life have reached unsafe levels, according to a new report authored by leading sound experts.

Building in Sound (pdf), developed by Biamp Systems in collaboration with acoustics expert and TED speaker Julian Treasure, reports that everyday noise levels regularly exceed World Health Organization’s (WHO) recommended levels.

The study draws clear links between excessive noise and poor acoustics and ill-health, distraction and loss of productivity, even disruption to educational development.

Drawing on a variety of academic, government and industry body sources, the paper has identified the economic and social impacts noise can have on everyday life – whether in a city, at work, in a classroom or hospital.

Examples of the sort of noise levels urban populations are regularly exposed to include:

—An air conditioning unit puts out sounds of 55 decibels. At this level, sleep is impaired and the risk of heart disease increases. Yet an average busy office has been recorded at 65 decibels.

—Street traffic has been recorded at 70 decibels. Regular unprotected exposure to the same level of noise can lead to permanent hearing loss.

—The average noise of a motorway is around 85 decibels, the same point at which U.S. Federal Law mandates hearing protection for prolonged exposure. 

The study also looks at solutions to the issues – given that road traffic noise is estimated to cost between 30 and 46 billion Euros a year ($39 and 60 billion USD a year), or 0.4% of GDP in the European Union.

It calls for an integrated approach to acoustic design that incorporates cutting edge sound technology with a more thoughtful approach to architectural design and construction.

Properly executed, managing sound can lead to higher employee productivity and job satisfaction, lower crime rates in urban environments, and increased sales in business.

“Noise is a major threat to our health and productivity – but until now we have been largely unconscious of its effects because of our obsession with how things look,” says Treasure, who is chairman of The Sound Agency. “We need to start designing with our ears, creating buildings and public spaces that sound as good as they look. If we do that, we can transform the productivity and well-being of office workers, patients in hospitals and children in schools, among many others.”

“This isn’t a call for silence, but an appeal to start considering the effects poorly managed sound can have,” adds Graeme Harrison, vice president of marketing at Biamp Systems. “The right sound and acoustics can transform education, healthcare and work, but we have to address the problem now because it’s only going to become more difficult in the future.

“We have the technology and expertise to manage the acoustics of new and existing environments, but now’s the time to act and build in sound.”

Download the pdf report, Building In Sound (pdf), here.

Biamp Systems


Posted by Keith Clark on 12/07 at 09:07 AM
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Radial Introduces ChainDrive 1x4 Distro And Line Driver For 500 Series

Accepts balanced or unbalanced source and distributes the signal to four front-panel 1/4-inch TRS outputs

Radial Engineering has introduced the ChainDrive 1x4 distro and line driver, the latest in the company’s ever-growing range of 500 series modules.

The ChainDrive is a single-wide audio distribution module that accepts either a balanced or unbalanced source via the 500 series power rack input and distributes the signal to four front-panel 1/4-inch TRS outputs.

Four conveniently positioned front panel level controls make it easy to optimize signal-to-noise levels.

For those equipped with a Radial Workhorse, the Omniport is set up as an unbalanced to balanced converter for manipulating hi-Z signals in the pro-audio domain.

The ChainDrive can also distribute a full stereo program using TRS connectors following the tip-left, ring-right, sleeve-ground convention. 

Once connected, the ChainDrive lets you create exciting signal chains such as multi-band compression, feeding several guitar amps and effects simultaneously, or even taking a stereo program and sending it to various digital processors, dynamic controllers and effects at the same time.

According to company president Peter Janis: “With so much happening around the 500 series, I would not be surprised to see some contractors and PA companies beginning to migrate to the standard. The ChainDrive is probably one of the modules that is best suited to bridge the gap between studio and commercial installations.” 

As with all Radial products, the ChainDrive is built to the highest standards. It begins with a 14-gauge steel outer casing with a galvalume shield to protect the circuit board.

Each potentiometer is outfitted with a steel shaft and output jacks are made from extra tough glass-filled nylon with nickel-silver contacts that will not tarnish.

Full surface ground plane with double sided solder points further ensure part integrity remains while noise is kept to a minimum.

The ChainDrive is now shipping. Estimated retail price: $350 USD.

Radial Engineering

Posted by Keith Clark on 12/07 at 07:52 AM
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Thursday, December 06, 2012

In The Studio: Microphone Techniques To Produce Warm, Spacious Stereo

Explaining and exploring stereo recording microphone techniques

Renowned recording engineers Jack Renner of Telarc Records and Marc Aubort of Nonesuch Records describe spaced omnidirectional microphones as their favorite technique for stereo recording of symphonic ensembles.

Of all the stereo mic techniques available, the “spaced-omni” method is especially good at providing a warm, full sound (deep low-frequency response) and a spacious sense of ambience. It adds up to a pleasant listening experience.

We’ll explore how this technique - and others - can be beneficial.

Spaced Pair
With the spaced-pair method (also called AB), you place two identical mics a few feet apart and aim them straight ahead (Figure 1). The mics can have any polar pattern, but omni is most popular for this method. The greater the spacing between mics, the greater the stereo spread.

How does this method work? Instruments in the center of the group produce the same signal from each mic.

When you monitor the mics, you hear a phantom image of the center instruments midway between your loudspeakers.

If an instrument is off-center, it is closer to one mic than the other, so its sound reaches the closer microphone before it reaches the other one. Both mics produce the same signal, except that the farther mic’s signal is delayed compared to the closer mic’s signal.

If you send the same signal to two speakers with the signal in one channel delayed, the sound image shifts off center. With a spaced-pair recording, off-center instruments produce a delay in one mic channel, so they are reproduced off center.

The spaced pair codes instrument positions into time differences between channels. During playback, the brain decodes these time differences back into corresponding image locations.

A delay of 1.2 millisecond (msec) is enough to shift an image all the way to one speaker. You can use this fact when you set up the mics. Suppose you want to hear the right side of the orchestra from the right speaker.

The sound from the right-side musicians must reach the right mic about 1.2 msec before it reaches the left mic. To make this happen, space the mics about 2 to 3 feet apart.

This spacing makes the correct delay to place right-side instruments at the right speaker. Instruments partway off center produce interchannel delays less than 1.2 msec, so they are reproduced partway off center.

If the spacing between mics is, say, 12 feet, then instruments that are slightly off center produce delays between channels that are greater than 1.2 msec. This places their images at the left or right speaker. I call this “exaggerated separation” or a “ping-pong” effect.

On the other hand, if the mics are too close together, the delays produced will be too small to provide much stereo spread. Also, the mics will tend to emphasize instruments in the center because the mics are closest to them.

The spaced-pair method tends to make off-center images unfocused or hard to localize. Why? Spaced-pair recordings have time differences between channels.

Stereo images produced solely by time differences are not very sharp. You still hear the center instruments clearly in the center, but off-center instruments are harder to pinpoint.

Spaced-pair miking is a good choice if you want the sonic images to be diffuse or blended, instead of sharply focused.

Another flaw of spaced mics: If you mix both mic channels to mono, you may get phase cancellations of various frequencies. This may or may not be audible.

Spaced mics, however, give a “warm” sense of ambience, in which the concert-hall reverb seems to surround the instruments and, sometimes, the listener. Here’s why: The two channels of recorded reverb are incoherent; that is, they have random phase relationships. Incoherent signals from stereo speakers sound diffuse and spacious.

Because spaced mics pick up reverb incoherently, it sounds diffuse and spacious. The simulated spaciousness caused by the phasiness is not necessarily realistic, but it is pleasant to many listeners.

This lack of correlation between channel signals shows up on a correlation meter. For example, the IK Multimedia T-Racks 3 mastering software includes a correlation meter, which shows almost zero correlation between channels for the spaced-pair technique. It shows 0.3 to 0.5 correlation (out of 1) for the N.O.S near-coincident technique.

Another advantage of the spaced pair is that you can use omni mics. Because of the physics of their operating principles, an omnidirectional condenser mic has deeper bass than a unidirectional condenser mic.

A typical studio omni condenser mic—of any size—has a flat response down to 20 Hz. A typical small-diaphragm cardioid condenser mic is flat down to 100 Hz, and 3 to 4 dB down at 50 Hz at 1 meter from the sound source. (These specs are from ACO Pacific, Neumann and Schoeps.)

Let’s compare the spaced-pair method to other stereo techniques.

Coincident Pair
With this method (also called XY), you mount two directional mics with grilles touching, diaphragms one above the other, and angled apart (Figure 2).

For example, mount two cardioid mics with one grille above the other, and angle them 120 degrees apart. You can use other patterns too: supercardioid, hypercardioid, or bidirectional. The wider the angle between mics, the wider the stereo spread.

How does this technique make images we can localize? Recall that a directional mic is most sensitive to sounds in front of the mic (on-axis) and progressively less sensitive to sounds arriving off-axis. That is, a directional mic puts out a high-level signal from the sound source it’s aimed at, and produces lower-level signals from sources to the side of the mic.

The coincident pair uses two directional mics that are angled symmetrically from the center line (Figure 2). Instruments in the center of the group produce the same signal from each mic. When you monitor the mics, the same signal comes out of each speaker. Identical signals from two speakers produce a phantom image midway between the speakers. So you hear the center instruments in the center.

If an instrument is off-center to the right, it is more on-axis to the right-aiming mic than to the left-aiming mic. So the right mic will produce a higher-level signal than the left mic.

When you monitor the mics, the right speaker’s signal is louder than the left speaker’s signal. This reproduces the image off-center to the right. So you hear the right-side instruments toward the right side.

That is how coincident stereo miking works. The coincident pair codes instrument positions into level differences between channels.

The brain decodes these level differences back into corresponding image locations. A pan pot in a mixing console works on the same principle. If one channel is 15 to 20 dB louder than the other, the image shifts all the way to the louder speaker.

Suppose we want the right side of the orchestra to be reproduced at the right speaker. That means the far-right musicians must produce a signal level 20 dB higher from the right mic than from the left mic. This happens when the mics are angled apart by a certain amount.

Instruments partway off center produce interchannel level differences less than 20 dB, so you hear them partway off center.

Listening tests have shown that coincident cardioid mics tend to reproduce the musical group with a narrow stereo spread. That is, the group does not spread all the way between speakers.

A coincident-pair method with excellent localization is the Blumlein array. It uses two bidirectional mics angled 90° apart and facing the left and right sides of the group.

A special form of the coincident-pair technique is the mid-side (MS) recording method illustrated in Figure 3

It uses a “mid” microphone facing the middle of the orchestra and a bidirectional microphone aiming to the sides. The middle mic is most commonly cardioid, but it can be any pattern.

In a device called a matrix, the signals from both mics are summed (mixed together) to produce the left-channel signal and are differenced (mixed in opposite polarity) to produce the right-channel signal.

You can remote-control the stereo spread by changing the mid/side ratio in the matrix. This remote control is useful at live concerts, where you can’t physically adjust the mics during the concert.

You can also control the stereo spread during mixdown rather than during the recording. For example, you can use a computer DAW to vary the stereo spread without using a matrix device.

A recording made with coincident mics is mono-compatible. If you expect that your recordings will be heard in mono (say, on TV), then you’ll probably want to use coincident methods.

Near-Coincident Pair
In the listening tests available here in mp3 format, I compared a spaced-omni pair to a near-coincident pair. In the latter method, you angle apart two directional mics, and space their grilles a few inches apart horizontally (Figure 4).

Even a few inches of spacing increases the stereo spread compared to a coincident pair and adds a sense of ambient warmth or air to the recording.

The greater the angle or spacing between mics, the greater the stereo spread.

How does this method work? Angling directional mics produces level differences between channels.

Spacing mics produces time differences. The level differences and time differences combine to create the stereo effect.

If the angling or spacing is too great, you get exaggerated separation. If the angling or spacing is too small, you hear a narrow stereo spread.

A common near-coincident method is the O.R.T.F. system, which uses two cardioids angled 110 degrees apart and spaced seven inches (17 cm) horizontally. (O.R.T.F. stands for Office de Radiodiffusion Television Française—French Broadcasting Organization.)

Usually this method gives accurate localization. That is, instruments at the sides of the orchestra are reproduced at or very near the speakers, and instruments halfway to one side are reproduced about halfway to one side.

The N.O.S. (Dutch) system uses two cardioids angled 90 degrees and spaced 12 inches (30 cm), while the D.I.N. (German) system is 90 degrees and 7.9 inches (20 cm).

Compared to O.R.T.F., those methods have less off-axis coloration because the mics are less angled away from the center instruments.

Also their 90 degrees angle between mics is easier to set up visually than the O.R.T.F. 110 degrees angle between mics.

Baffled Omni Pair

This method uses two omni mics, usually ear-spaced, and separated by a hard or padded baffle (Figure 5).

To create stereo, it uses time differences at low frequencies and level differences at high frequencies.

The spacing between mics creates time differences. The baffle creates a sound shadow (reduced high frequencies) at the mic farthest from the source.

Between the two channels, there are spectral differences—differences in frequency response.

Some examples of baffled-omni pairs are the Schoeps or Neumann sphere microphones (Figure 5), the Jecklin Disk, and the Crown SASS-P MKII stereo microphone (Figure 6).

The omni condenser mics used in the baffled-omni method have excellent low-frequency response.

A special form of the baffled-omni pair is binaural recording with an artificial head (dummy head). The head contains a microphone flush mounted in each ear.

You record with these microphones and play back the recording over headphones.

This process can recreate the locations of the original performers and their acoustic environment with startling realism.

You can clip a pair of miniature omni or cardioid mics onto the earpieces of eyeglasses. Each mic is on the opposite side of your head, either in your ears or on your temples.

To compensate for the acoustic effect of the head, the signals need some EQ (a broad dip around 3 kHz).

Boundary (surface-mounted) mics can be used for any type of stereo miking.

Listening Comparison
I recorded a symphonic band simultaneously with a spaced-omni technique and a near-coincident technique. For spaced omnis I used two ACO Pacific MK224PH measurement mics at 12 feet from the ensemble and 30 inches apart.

These phantom-powered units have extremely low noise, flat response from 20 Hz to 20 kHz, and an XLR output.

For near-coincident mics I used two Neumann KM-140 cardioid condensers in the N.O.S. configuration at 20 feet from the ensemble. I boosted 100 Hz by 6 dB on the KM-140s to make their low end match the omni pair.

To me at least, the spaced omni’s have a wider stereo stage and more envelopment. The equalized cardioids don’t sound quite as natural to me as the spaced omni’s, but they have sharper imaging. Feel free to draw your own conclusions.

Comparing The Four Techniques
Coincident pair:
• Uses two directional mics angled apart with grilles touching.
• Level differences between channels produce the stereo effect.
• Images are sharp.
• Stereo spread ranges from narrow to accurate. 
• Signals are mono compatible.

Spaced pair:
• Uses two mics spaced a few feet apart, aiming straight ahead.
• Time differences between channels produce the stereo effect.
• Off-center images are diffuse. 
• Stereo spread tends to be exaggerated unless a third center mic is used, or unless spacing is under 2 to 3 feet.
• Provides a warm sense of ambience.
• Provides excellent low-frequency response if you use omni condensers.
• Tends not to be mono compatible, but this may not be audible.

Near-coincident pair:
• Uses two directional mics angled apart and spaced a few inches apart horizontally.
• Level and time differences between channels produce the stereo effect.
• Images are sharp.
• Stereo spread tends to be accurate.
• The hall sounds more spacious than with coincident methods.
• Tends not to be mono compatible.

Baffled omni pair:
• Uses two omni mics, usually ear-spaced, with a baffle between them.
• Level, time, and spectral differences produce the stereo effect.
• Images are sharp.
• Stereo spread tends to be accurate.
• Excellent low-frequency response.
• Good imaging with headphones.
• The hall sounds more spacious than with coincident methods.
• Stereo spread is not adjustable except by panning the two channels toward the center.
• More conspicuous than other methods.
• Tends not to be mono compatible, but this might not be audible.



AES and Syn Aud Con member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.

Posted by admin on 12/06 at 10:08 AM
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Wednesday, December 05, 2012

TransAudio Group Introduces Daking Comp 500 VCA Compressor/Limiter

500 series module offers classic FET qualities

TransAudio Group, the worldwide distributor of products by Geoffrey Daking and Company, is introducing the Daking Comp 500, an easy-to-use VCA compressor/limiter module compatible with 500 series rack enclosures.

The semi-automatic Daking Comp 500, intended for both tracking and mixing, has been designed to perform and sound similarly to the FET circuitry found in Daking’s other compressor products, even though it uses a VCA. 

The front panel controls of the Comp 500 are simple to use. Compression is set with one knob, ranging from “Less” to “More.” Release time is switchable between Fast (0.5 ms) and Auto.

The Auto setting engages a dual time constant, which initially releases quickly before slowing, effectively eliminating “pumping,” and provides the punch associated with the 1970s UK company Audio & Design Recording’s Compex unit.

Attack may be switched between Fast (1 ms) and Slow (16 ms), while Ratio is switchable between Compressor and Limiter modes, respectively setting a 3:1 or 15:1 ratio.There is also a hard-wire Bypass switch.

A Stereo switch is provided that allows up to six units to be accurately linked. The 8-segment meter is selectable between gain reduction or output levels, and offers VU ballistics with floating peak.

The class A circuitry of the Daking Comp 500 incorporates discrete transistors and single-sided amplifiers, offering a fast and transparent signal with plenty of headroom and a low current draw.

All components are lead free and compliant with the Europe Union’s RoHS directives.

MSRP for the Daking Comp 500 is $795.

TransAudio Group

Posted by Keith Clark on 12/05 at 09:12 AM
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