Article

Thursday, April 26, 2012

Kimbra Performs Around The World With Earthworks

Kimbra is quickly becoming somebody that you ought to know. With a slew of SXSW showcase gigs, nationwide tours in support of Gotyé and Foster The People, tens of millions of views on YouTube, and a cover of the hit Gotyé song she lends her powerful vocals to, “Somebody That I Used to Know,” performed on Fox’s hit show Glee, Kimbra is bringing her music to the masses, one powerful performance at a time.

And Earthworks is there to capture that sound all along the way thanks to Kimbra’s FOH Engineer Angus Davidson. Covering every instrument on stage, Earthworks microphones have brought Kimbra’s lively and moving performances to the masses as the intensely talented musical force tours the globe.

Angus Davidson, FOH engineer for Kimbra and Crowded House, is no stranger when it comes to using all Earthworks microphones on stage. “Previously, with Crowded House, we also enjoyed tremendous success using 29 Earthworks mics, which we used on vocals, drums and instruments,” Davidson noted. “Everything was miked with Earthworks microphones.”

For Kimbra’s performances, Davidson went the all Earthworks route once again. The new SR40Vs are being used on both Kimbra’s and her band’s vocals. On drums Davidson uses a pair of SR40s for overheads, 2 DP30s on snare (top and bottom), DP30s on toms, SR30 on high-hat and SR30 with KickPad on kick drum. SR30s are used on the guitar amps, as well as for audience mics, and some things are fed direct. “An integral part of sound for Kimbra and her band is the exclusive use of Earthworks microphones,” Davidson remarked.

Monitor Engineer for Kimbra’s tour is Rod Matheson, who worked with Kylie Minogue for 18 years, did a world tour last year with Bristish trip hop duo Massive Attack and has done monitors for countless other artists throughout the world. “In my opinion, Rod is one of the finest monitor engineers on the planet,” Davidson said. “He has an exceptional set of ears and is a very particular and fastidious engineer who gets great results. Rod was a little dubious at first, about using all Earthworks microphones, but after using them the first time, he agreed that these mics were in a league of their own.”

The newest addition to the Earthworks lineup, the SR40V, has found a welcome home on stage with Kimbra. “The Earthworks SR40V is an incredibly flat microphone. The benefit to me is that I can make it sound any way I want, depending on the application. Because of the super fast rise time of the small diaphragm, I can add and subtract EQ without it ever sounding ‘tubby or flabby.’ Its natural presence and tightness is quite unique,” noted Davidson. “Kimbra has an incredible vocal range from a whisper to a huge full voice. Regardless of how she sings, I can always place her vocal exactly where it needs to be in the soundscape. No other vocal microphone I have used can compete with the SR40V. For me it is simply the best vocal microphone ever.”

With a background that spans three and a half decades of live and studio work, Davidson has a unique perspective on microphone technologies over that time.

“When I was starting in this industry 35 years ago there was an enormous focus on the “rise time” and “transient response” of microphones,” Davidson noted. “Somehow over the past 30 years that function of microphone audio physics seems to have been lost.  David Blackmer’s unique microphone technologies have allowed all of this to work really well for the first time.

“Earthworks microphones very clearly illustrate how important rise time, transient response, fast diaphragm settling time and extended frequency response truly are.  These microphone characteristics provide an enormous depth of field, with incredible detail over the audio spectrum, not to mention their incredible phase coherency, particularly when we use these mics on every drum.”

Davidson remarked on his experiences with the technologies found in Earthworks mics specifically on a drum kit, “I’ve listened very closely to recordings where we’ve used Earthworks mics on every element of the drum kit. The detail, separation, and accuracy of the stereo sound stage is unbelievable. The drums are crisp, detailed and really natural sounding.”

“In contrast, you could select a number of other ‘Specialty’ mics to use on a kick drum and get that huge round fat bottom end with the click of the beater, but it is a very one-dimensional sound,” Davidson continued. “There are all sorts of microphones out there that do a great job, specifically on one thing, but there are very few microphones that I have ever used that do “everything” equally well, like the Earthworks do. I don’t use two microphones on kick drum; instead, I use a single Earthworks SR30 with a KickPad. This combination with the addition of a little EQ makes the kick drum really beautiful and natural sounding, with no B.S. to the sound.”

Davidson continues with this natural sounding approach beyond drums to each instrument on stage. “I also don’t want to be pulling ridiculous monster sounds for every instrument that’s on stage,” said Davidson. “Instead, I feel that we should be balancing what is there, and not trying to reinvent the wheel. I like the idea of creating a sound stage that makes you feel a certain way; by the way you mix it, and not having to make every instrument and vocal sound bigger than everything else. There seems to be a trend to create thunderous bass, but at the expense of everything else. Why would you keep applying EQ, compression and all types of signal processing and not treat the signal with the respect it deserves? I want to be able to look on stage, and be able to clearly hear everything I can see. The essence of this is to pick a microphone that will do that effectively, and then place it perfectly to best reproduce the source it is hearing.”

Before turning his attention back to the demands of the tour, Davidson offered these final thoughts. “Earthworks microphones look at the sound and give it an enormous amount of respect at the start, and that makes our job ten times easier. So, when I see someone playing something, I want to be able to hear it clearly and distinctly. The Earthworks mics make that job easier than it has ever been. They are so detailed and so clear. It’s like an artist walking around all the time with dark glasses on, and then one day taking them off and discovering that it’s ‘light out there’.” 

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Microfiles: Electro-Voice 664, The Legendary “Buchanan Hammer”

A single-element cardioid, dynamic type mic was the first model to incorporate the company’s patented Variable-D design

My Baltimore-area high school theater was outfitted with the first quality PA system I ever worked with.

It had JBL horns and cabinets in a center cluster, powered by Crown amplifiers, with a 6-channel TAPCO mixer in the sound booth and Electro-Voice 664 microphones on stage.

Initially, to my finely tuned 10th grade ears, the system didn’t sound very good – the performers could barely be heard, and there was a lot of feedback.

It wasn’t long before I figured out that the real problem was operator error, not the system.

I eventually got the hang of running it correctly, including learning the importance of proper mic placement.

And what mics they were! I fell in love with the EV 664 – far better sounding than my own Lafayette mics for my garage band, and built like a tank. Later, I wasn’t surprised to find out the 664 had the nickname “Buchanan Hammer” due to its rugged design.

The story behind the nickname, as I know it, is that during his legendary microphone lectures, the late Lou Burroughs (one of the founders of EV) would beat a 664 against a 2 x 4, and/or use it to hammer nails into a board, and then plug it in and use it for the rest of his presentation. The Buchanan part of the moniker refers to the town in Michigan where the company was headquartered for decades.

Timeless Design
Introduced in the mid-1950s, the 664 sported a cool “Art Deco” design, with a sleek yet curvy chrome body that evoked the popular automobile tailfins of the period. Inside was some serious technology.

The single-element cardioid, dynamic type mic was the first model to incorporate the company’s patented Variable-D design (U.S. patent number 3115207, awarded in 1963) still found in several EV mics to this day, including the broadcast-favorite RE20 and the recently introduced RE320.

Variable-D (“Variable Distance”) uses three ports to cancel sound from the rear, while the side ports (slots located on the sides) are coupled to the back of the diaphragm and help cancel high-frequency sounds.

The hole on the top (located toward the front of the body’s “raised spine”) works the same as the side ports, but has a longer path and added filters to affect mid frequencies. The single hole at the rear of the spine has a longer path and more filtering to address low frequencies.

This all combines to give the mic good pattern control over a wide frequency range and a reduced proximity effect. An ad from 1961 states “The 664 does not BOOM when performer crowds microphone.”

The 664 was available in three finishes: satin chrome, non-reflecting gray (664A) and a gold finish (664G). I also own one that is branded Executone EXCC, and it has a brown body with a gold windscreen, as well as a chrome model that is branded DuKane 7A160.

More Versions

There were actually two designs of the 664. The earlier one had a more classic base, three ports on each side, and the old script logo on the switch-plate. The later version had a more modern rounded base section, a single large port per side, and a larger switch-plate cover that featured the round, red EV logo. Both utilized a 4-pin EV screw-on connector that was popular on many of their models.

The 664 shipped with an 18-foot cable that was not terminated at the console end.

There was yet another version called the 664A, but it was supercardioid, a lot smaller, and had more modern styling. It still utilized the Variable-D design, but had a long plastic port along its spine instead of separate rear port entrances.

The 664 holds a special place in my collection. It was the first true professional caliber microphone I worked with, and the single port model shown here is the first mic I ever purchased just to collect and not use onstage.

Great mics and great memories!

Electro-Voice 664 Specs (original 3-port model)

Transducer Type: Dynamic, non-metallic Acoustalloy diaphragm
Polar Pattern: Cardioid
Frequency Response: 40 Hz – 15 kHz
Sensitivity:  -55 dB at 150 ohms
Nominal Impedance: Switchable high impedance or 150 ohms
Size:  7 3/16 x 1 7/8 inches
Net Weight: 28 ounces
1961 Price: $49.98

Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb.

 

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Church Sound: Going With What You Know When The Pressure’s On

Stacking the odds in your favor from the outset

Several years ago, I was invited to be a presenter at an audio industry trade show, and while there, I greatly enjoyed meeting some fellow presenters.

In fact, after the convention center hall closed, eight of us audio “geeks” went to enjoy dinner together, and it turned out to be a fun—and instructive—evening on many levels.

At one point, as we were seated around the table, someone in the group posed this hypothetical question: “If you got a call to do an event, had to be there in an hour, and weren’t told much of anything about the performers or performance, which microphones would you bring?”

This was quickly followed up with: “And oh, by the way, you’re limited to three models. Any manufacturer, but only three different models.”

I immediately jumped in with my take—a dozen dynamic vocal mics, a dozen dynamic instrument mics, and a dozen quality condenser mics. Everyone generally agreed.

Now, what models? This is where it got even more interesting, and more heated, as everyone weighed in with their views about which manufacturers and models were better—and why.

After about 20 minutes, someone suggested making a rule that we all, as a group, had to compromise and agree on the three models. (We also added a caveat that anyone who would not compromise would be on the hook for paying for dessert for the entire group.)

Now it got really interesting! I watched (and participated) as these audio professionals became amazing salesmen, trying their best to sell their particular microphone models.

Finally someone said, “Wait a minute—if we don’t really know what we’re going to be facing at the gig, and in our scenario here, we all must agree on this mics, then wouldn’t it make sense to pick ones we’re all familiar with?”

This immediately brought clarity to the discussion, and it boiled down to this: “Has everyone used a Shure SM58 (ubiquitous dynamic vocal mic)? A Shure SM57 (ubiquitous dynamic instrument mic)? A Shure SM81 (all-around good condenser)?” 

The answer was yes for everyone.

And there we had it—those three mics would be the group’s choice in this particular scenario.

My goal here isn’t to promote Shure products (though I do like and use a number of Shure mics), but rather to get to this point: It’s always best to use what you know, especially if you’re in a high pressure situation.

The logic is simple:

1) You’ll know approximately where to set the preamp gain on the console.

2) You can quickly rough in the EQ based on what you are mic’ing because you’re familiar with the characteristics of the mic.

3) You’ll know the best placement of the mic in relationship to the sound source because you’re familiar with the pickup pattern and sensitivity of the mic.

4) You can best place stage monitors because you know the greatest rejection point of the mic.

In other words, you’re stacking the odds in your favor from the outset to deliver a quality result.

This logic applies to more than just microphones. A rule I strictly follow is to not use any unfamiliar piece of gear during a show. I don’t care if I’m told that it’s the best thing ever, and/or really expensive. Nope - I’m going to use what I know. 

Now when it comes to trying new things at rehearsals, you bet!  This is where it makes sense—generally it’s a lower pressure situation, and I have plenty of time to learn about the gear, and to learn what adjustments I need to make, and how to use it best in that particular application.

Still, even before trying something new at rehearsals, I must see the piece, check its specs, touch it, and play around with it in order to get comfortable enough to put it into use. I don’t want to waste the band’s time if the equipment does not work or if I have to take time to learn how to even make it functional. 

I love playing with new (and/or unfamiliar) gear. The only caveat is that I choose to use it in a controlled fashion so that it’s a win for all come the performance.

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.

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Wednesday, April 25, 2012

Church Sound: Choosing The “Best” Microphones

It may have better specs, but that doesn’t mean it’s the best choice

 
I received an e-mail inquiring about microphones; specifically what constitutes a good microphone.

The reader had seen my post on rechargeable batteries and noticed that I was using Shure SM58 capsules on the mics under test.

That made him wonder about the report he had just received from a consultant who had reviewed their church’s A/V systems.

From the report, to wit:

“Good quality microphones give the biggest performance increase for the money invested. If the right sound is not captured by the microphone, then no amount of technical gadgets is going to be able to get a good sound. Avoid vocal microphones with high proximity effect (increase in bass response) (e.g. Shure PG58, Shure SM58).”

I’ll start by stating that I disagree with most of that paragraph. Yes, good quality microphones are important.

However, when you rank them on the “benefit for dollars spent” scale, you only get big gain for dollars if you’re upgrading from those 3 for $19 deals you see in the Kingdom Electronics ads.

Once you get into mics that cost $100 or more, the differences are often subtle and in some cases, academic. Case in point; Bono quite often sings into an SM58. Should he be avoiding that microphone? I wonder if he’s ever tried the Shure PG58?

So why do I think microphones do not provide the greatest improvement for dollars invested? Simple: What we do is sound reinforcement in a live setting and as such, I think speakers better fit that description. I’ll unpack this more in a later post; let’s get back to microphones.

Now, don’t get me wrong; I’m a big fan of good mics. In fact, I’ve spent a fair amount of money recently improving the depth and breadth of our mic locker. A good mic can make a big difference. And right now, I’m buying new mics because I don’t have enough money to buy a new PA.

So even though my other sound engineers and I notice that the Heil Sound PR22 sounds a lot better on the snare than the SM57 it replaced, I’ve yet to have anyone come up to me and tell me that the snare sounds better. That’s because it’s a subtle difference and we’re listening for it (and we note how much less EQ is required to make it sound good).

Conversely, if we hung a new PA that had vastly better coverage, evenness, phase response, lower comb filtering and overall better fidelity, I think people would notice.

To be sure, it’s going to cost some coin to make that happen, and for the same amount of money, I could have bought a truckload of mics.

But I’m quite sure I could replace the e609 on our guitar amp with a U87 (roughly 30 times the price of a 609) and no one would notice.

So my recommendation to the reader was not to replace the drawer full of SM58s just yet, rather, investigate a new speaker system.

Once the system can faithfully reproduce what you send it, then start looking at better mics.

Now let’s get on to another part of the report that I mostly agree with.

“Microphones should be selected from a trial use after the rest of the sound system is brought up to standard. The more expensive microphones have a flatter frequency response (more natural sound, higher volume before feedback occurs), better off axis rejection (more volume before feedback, less pick-up of adjacent instruments or voices), lower proximity effect (tone changes at varying distance from mic), lower handling noise, better ‘pop’ filters.”

Generally all of this is true. What I take issue with is the notion of “more expensive” microphones are inherently better choices.

Case in point: When we bought our new wireless system, I specified one Shure KSM9 capsule that I planned on using that for our worship leader. Turns out, it doesn’t work for him. And as we’ve tried it on many of our vocalists, it doesn’t work for most of them either. In fact, some of them really don’t like it.

So here we have a capsule that’s over $500, and for the most part, we and most of our singers prefer capsules that sell for less than half that. Quite honestly, I’d be really ticked if I had ordered ten KSM9s instead of ten SM58s based on the notion that more expensive = better. In fact, I’m going back and ordering a few more Shure Beta 87s because in our PA, with our singers, they are a superior choice.

Does this make the KSM9 a bad mic? No! On paper, it is be head and shoulders above the Beta 87 or SM58. However, the less-flat frequency response, proximity effect and wider pattern make the latter two better choices for our vocalists.

And that brings me to the one part of the consultant’s report that I thoroughly agree with:

“Microphones should be selected from a trial use after the rest of the sound system is brought up to standard.”

Before you go out and commit big dollars on new mics, try them out. If you can get demos, do it. If not, buy from a dealer who will let you return them if you don’t like them. Try a large cross-section of mics if you can. The best choice might surprise you.

In our case, we much prefer a Heil Sound RC35 on our worship leader over the KSM9, even though the Heil is half the price. And our student worship leader sounds fantastic on a RC22. I’ve always been a big fan of the Neumann KMS105; we had a KMS104 on our worship leader and I thought it made him sound muddy with no clarity at all.

Most importantly, don’t let anyone sell you a microphone because it’s more expensive and therefore “better.” It may have better specs, but that doesn’t mean it’s the best choice.

Try it out and hear for yourself.

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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Heil Sound Releases Updated PR 35 Microphone

Heil Sound has released an upgraded version of its PR 35 handheld dynamic microphone.

Changes include a new chassis, resulting in cosmetic differences with the old model.

In addition, rear rejection has been increased to -42 dB with the upgrade.

A concealed two-position roll off switch replaces the former thumb switch.

The PR 35 ships with three interchangeable colored trim collars that can be mixed and matched.

The new PR 35 will be priced the same as the current model and is available now.

The PR 35 was first introduced in 2008 and has since been used by artists such as Joe Walsh, Charlie Daniels, Stevie Wonder and others.

Heil Sound

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In The Studio: Tips For Mixing Rap Vocals, Part 3 - Compression

Compression is a powerful tool that many struggle to fully understand

 
Check out Mixing Rap Vocals Part 1 and Part 2 before reading this article.

Compression is a difficult subject because there is a lot you can do with it.

So let’s look at the main reasons to grab a compressor before getting into some of the more intricate uses.

Quick Macro-Dynamic Control

Macro dynamics refer to words and phrases. These are the clear dynamics you can hear as “this part is louder, that part is softer.” The most transparent way to get things sounding even is to actually automate the vocals manually.

But sometimes time doesn’t allow for this approach. So if you aren’t automating, a light ratio, slow attack, slow release, just catching the louder moments with the threshold is a good way to even things out.

Micro-Dynamic Control

What volume automation might not catch is the very quick dynamic changes – loose spikes at the fronts of words. These spikes aren’t heard so much as “volume” but more as an overall quality to the vocal.

The issue with these spikes is two fold – first, they eat away at your headroom pretty quickly– second, they will trigger any compressors you are trying to use for purposes besides micro-dynamic control.

It can be useful to dedicate a compression stage toward pulling back these vocal spikes. Generally a fast attack and release, and a light ratio does the job. The light ratio is to retain the articulation of the word and minimize frequency skewing.

The key is to set the threshold low enough to catch as much of the peak as possible while effecting the body of the signal as little as possible. I try to avoid using limiters for this purpose. I like the Empirical Labs Distressor for this (especially for controlling peaks while tracking), as well as digital style compressors such as the Logic or Pro Tools stock compressors or the Waves C1.

The attack setting is very important – it’s usually between a number of nano-seconds and two or three milliseconds in the digital world, and on the faster side of things for the analog world (totally varies unit to unit).

Getting A Vocal To Stay Audible Through A Mix

The power of compression is that you can make something louder while not actually raising the peak volume of the signal. This becomes extremely useful for making something cut through a dense mix or to come forward. This is probably where the majority of compression work for rap vocals come in.

Rap is generally an in-your-face, visceral style of music. The kick is physical, the snare is physical, subtlety isn’t really the overall goal. And the vocals are paramount. I’ve mixed a number of rap records where the vocals are lower in the mix, but never have I thought it was a good idea.

Generally I want the vocals to be equally as strong as the drums or stronger, and I want them as “forward” as possible. Compression is usually a part of that equation.

Optical Compressors

The smoothest way to get those vocals forward is through optical compression. The rounding quality of the attack and the unique shape of the knee in an optical compressor makes them ideal for vocal work.

Examples of optical compressors would be the CL1B, the LA2A, LA3A, your stock Logic compressor has an optical mode, RComp has an optical mode – and don’t quote me on this but RVox has an “optical” sound to it, as does the “smooth” setting on the UBK-1.

One of the advantages to opticals is that they tend to have easy access. Many have just one knob to control the degree of compression.

Attack And Release Time

Of course you’re not limited to simply optical compressors or fixed time settings.

Many other compressors work very well for rap vocals – in fact, any decent compressor can yield great results if set properly.

The key is setting the attack and release times appropriately. People will suggest milliseconds or time ratings for the best attack and release for vocals but the issue is that 300 ms on one compressor might give you the same results at 75ms on another.

So, I’d rather advise your compression technique based on the expected results. Your goal is to pull up as much of the “sustain” of the voice – the weight of it – while minimally affecting the articulation. Taking notes? – It’s about to get heavy.

When dealing with the articulation of the words, you’re primarily gauging your attack time. There’s generally a substantial range of attack speeds that work for vocals. What you don’t want to do is set the attack too short, or the shaping of consonants will be blurred. Nor do you want to set the attack too long, because you’ll allow the consonants to poke through too hard. So you want to find a middle ground.

A good way to experiment is to temporarily pull the threshold down a little farther than you normally would and find your attack setting that way, as the effect of the attack time will be more exposed with the lower threshold.

With the release time, my goal is to pull up as much of the body of the voice as possible. So I’m going to set the release on the faster side. I don’t want the voice to distort or become unnatural sounding, but I want as much body as possible before I get to that point.

In terms of both the compressor ratio, and the release time, I tend to be a little more aggressive with rap vocals than “softer” music. The “integrity” of the vocal sound is not really as important as the prominence of it. For a more relaxed, natural sound I might do a medium release and 3:1 or 4:1 ratio.

For a rap vocal I’m going for a pretty quick release, and I’m doing 4:1 up to 8:1. Rap isn’t really supposed to be “pretty,” so I don’t worry if the compression becomes a bit audible.

Thicker Vocals

Another great use for compression on vocals is to make the vocal sound thicker – particularly in rap. Rap is frequently recorded in home studios, even by big name artists. And home studios rarely produce the thick, full vocal sound that one can get at a professional facility.

So being able to thicken and give weight to a vocal is an extremely important skill. In order to do it right though, you need a little more than compression. You need an EQ to make sure the vocal is as even and smooth as possible. Then you need some “saturation.” Saturation is just a nice name for friendly distortion. Saturation moves and enhances the harmonics of the vocal. Over saturating will sound like crud, but just the right amount gives the impression of a richer sound.

So the formula is – get the vocals sounding clean, saturate to get the vocal sounding richer, and compress that signal. You have to tread carefully though as over EQ’ing, over saturating, or over compressing will make your vocals sound horrible. Unless you do all of that in parallel!

Parallel Compression

Parallel compression basically means making a copy of the signal, compressing the snot out of it, and then blending that parallel signal back in with the original signal.

The advantage here is that you can get really liberal with the effects, and just blend it in to where it doesn’t sound unnatural. This is great is you are trying to fill-out frequencies that weren’t really there in the original recording, because you can really saturate and compress the parallel signal and generate some very consistent dense harmonics.

Then just blend that in until just before it starts to sound too effected. One of the reasons I really like the UBK-1 for vocals is because it gives you a saturation stage followed by a compression stage and the ability to blend both in parallel.

Distortion Free Equalization

Lastly, compression can be used to tame frequencies without the artifacts from EQ. Often with vocals you’ll have moments where the vocalist changes their tone.

A common example is vocalists will often become more midrangy as they project because they tighten their neck and push more air through their nose. If you simply notch out some midrange – that might work, but it might also take away some of the energy of the overall performance, or some of the frequency information you need to make the vocal stand out.

A good alternative is to use a compressor with either an adjustable side chain, or an external side chain input. The side chain signal is what the compressor reacts to – so if you can EQ the side chain to target problematic frequency areas, the compressor will “intelligently” react to those tones and pull them down.

Conclusion

Compression is a powerful tool that many people struggle to fully understand, so try to get your hands on one and start experimenting. As always I’ll keep an eye on the comments in case there is anything that needs clearing up. I also encourage you to share your own compression tips here!

Matthew Weiss records, mixes, and masters music in the Philadelphia, New York, and Boston areas. Find out more about him here.

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

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Tuesday, April 24, 2012

Lectronsonics Announces Release Of Venue V5.1 Firmware

Lectrosonics is pleased to introduce the release of Version 5.1 firmware for the company’s Venue Wideband modular receiver system.

The V5.1 firmware provides an important enhancement that will be most beneficial to broadcast, location, and sound reinforcement professionals—the capability of a “TalkBack” function compatible with the newly introduced HH handheld transmitter.

When the transmitter is set to TalkBack mode and the function is engaged (activated when the user presses the multi-function button on the HH transmitter), the audio will switch to a different, predetermined, XLR on the back panel of the receiver. This functionality enables the transmitter to control audio routing, which can be very useful for situations that require real time communication between talent, director, monitor engineer, or other band members. In order for the TalkBack functionality to operate on the Venue receiver, at least one empty slot (and, thus, an associated unused XLR) is required in the mainframe.

If more than one receiver modules are installed upstream to the empty slot and set to TalkBack compatibility mode, they will share the next available empty slot as a TalkBack active output. In the event that multiple upstream receivers enter a TalkBack state at the same time, their audio will be mixed on the shared TalkBack output. Hence, it is possible to have multiple transmitters using a common TalkBack channel XLR.

Karl Winkler, director of business development at Lectrosonics, commented on the new V5.1 firmware offering, “We’ve received numerous requests over the years for this kind of talkback functionality on our handheld transmitter. With the new HH handheld, we felt it was the perfect time to incorporate this feature along with the corresponding setup in the Venue receiver. So far, the response has been really great.”

The latest version of the Venue firmware can be downloaded and installed by users via USB or by visiting the website.

Many audio field technicians are now in possession of measurement systems that can be used to assist the listening process in equalizing sound reinforcement systems.

But, they’re often surprised to find that the measured system response correlates poorly with subjective impression of how the system sounds.

In other words, the system can sound good when it looks bad on the analyzer, and it can sound bad when it looks good on the analyzer.

As a result, some users have become frustrated and distrustful of analysis systems in general.

Let’s look at why the eye and ear do not always agree on what is best regarding the response of the sound system.

First, consider the most popular methods of measuring the response of the sound system.

By “response,” I am referring to the magnitude of the frequency response as displayed on a dB (vertical) vs. logarithmic (horizontal) scale. The goal of technical system equalization is to produce a “flat” horizontal line on this display.

WORKING IN REAL TIME

The real-time analyzer (RTA) is essentially a bank of meters, each driven with a 1/n-octave constant percentage bandwidth filter so that only the level of a limited range of frequencies is displayed by each meter.

The original RTAs used analog meters, but current versions use a vertical row of LEDs for each 1/n-octave band. One-third octave resolution is the most popular, and correlates well with the response of the human auditory system. 

The RTA input is fed from an omnidirectional test microphone located at a listener position. Omnis are used because they typically have a very flat, “benign” frequency response over most of their band pass.

RTAs can also be software-based, utilizing the sound card on a personal computer to provide the A/D conversion of the microphone output voltage. A mathematical algorithm (the FFT) is used to produce the previously described dB vs. frequency display.

These “digital” analyzers emulate their analog counterparts in how the information is displayed, but differ in that the filters and display is the product of a computer algorithm rather than analog filters. This type of RTA is more versatile, as the octave-fractions, colors, etc. are under software control.

Regardless of which type is used, the standard method-of-use is to drive the sound system with pink noise (equal energy per 1/n octave) and adjust the system equalizer for a “flat” magnitude response on the analyzer display.

RTAs are powerful tools when certain guidelines are followed, but indoors they can indicate a system response with poor correlation to what the listener is hearing. The major consideration is the placement of the measurement microphone.

If the mic is placed in the near field of the loudspeaker (typically less than 10 feet), the correlation with human hearing is pretty good. At this position, the direct energy from the loudspeaker dominates what is being observed on the analyzer and very little of the reflected energy from the room is included in the displayed response. Adjustment of the equalizer for a flat direct sound field on the analyzer produces a desirable result.

The down side to the near-field placement is that the measured response is very sensitive to small vertical movements of the microphone when the loudspeaker has offset vertical components (as most do). This sensitivity can be reduced if the microphone is moved to a greater distance from the loudspeaker (into the far field) since the path-length difference back to the individual components becomes more equal.

But, as the microphone is moved further away, the reflected energy from the room begins to dominate the displayed response.

GIVING EQUAL WEIGHT

Microphones have no “perceptual” abilities. They do not localize sound or discriminate early sound energy from late energy like humans do.

A listener at a distance remote from the loudspeaker will pay more attention to the direct field of the loudspeaker than sound that is building-up in the room.

A microphone gives equal weight to all energy without regard to where it is coming from.

A simple experiment to verify this is to stand at the microphone position and listen to the loudspeaker and then route the mic through a headphone amplifier and listen to it through headphones - not the same thing at all.

Low frequency sounds tend to linger in rooms longer than high frequency sounds, because most rooms have more high frequency absorption than low frequency absorption.

As such, the room becomes “bass heavy” when the total sound field is considered. This extra low frequency information will dominate what is observed on the RTA, and the knee-jerk reaction is to attempt to “flatten” the response by boosting the high frequency bands on the equalizer.

The result is a system with excessive high frequency output and a resultant “harsh” sound quality.

When RTAs are used in this manner, it is important to equalize to a “target curve” rather than for a flat frequency response. The popular “X” curve for theaters is flat to 2 kHz, where it starts rolling off the high frequency response at about 3 dB per octave. It is -10 dB at 10 kHz relative to 2 kHz.

This represents 1/10th power at 10 kHz relative to flat response. The one-third-octave analyzer and the target curve have served sound practitioners well for years, and remains a viable approach to system calibration.

RECENT METHODOLOGIES

Technology has yielded some new methods for acquiring the system response at a listener position. A complex comparison (both time and frequency information) of the input and output of a system is called the transfer function. It includes both the magnitude and phase response of the loudspeaker/room at the microphone position.

This has become a popular method of analysis, as it allows any input stimulus to be used to test the system, since the displayed response is just the difference between “what you put in” and “what you got out.”

Transfer function analysis has the added advantage of the ability to use a “time window” to exclude late arriving energy from consideration in the response. This can prevent the low-frequency build-up problem that plagues traditional real-time analysis. With proper implementation of a time window, the system response can be adjusted without the need for frequency weighting via a target curve.

A major difference between transfer function analysis and 1/n-octave real-time analysis is that the former requires the removal of the signal delay between the two signals being compared. The stimulus (the reference signal) always has a much shorter path back to analyzer input than the output of the measurement microphone. Sources of delay include the travel time through the air and the latency of digital processors.

Failure to properly synchronize the reference signal and the microphone’s signal will result in an erroneous display of the system’s response. The length of the time window must also be selected - in other words, “how much of the room decay do I want to include in the response?”

Unfortunately, there is not an optimum size for the entire spectrum. A short time window excludes much of the room decay at the expense of low-frequency resolution. A long time window improves frequency resolution at the expense of gathering too much of the room’s decay. A compromise is required.

The human auditory system perceives pitch on a proportional (logarithmic) frequency scale. This is one reason that we use constant-percentage bandwidth filters for tuning audio systems - the bandwidth grows with increasing frequency.

Frequency-dependent bandwidth suggests that the length of the windowing function used in transfer function analysis should be varied in the same manner - a decreasing length with increasing frequency.

This produces a somewhat “anechoic” response at high frequencies with increasing frequency resolution as frequency decreases.

The time window length is a function of frequency, with even the longest window (highest frequency resolution) excluding much of the late energy from the room.

Another caveat of this type of analysis is that much greater frequency detail is possible than with the typical 1/3-octave banded display. Phase interference effects from reflections or multiple drivers are clearly visible on the analyzer.

Such anomalies are almost always position-dependent, so careful “corrections” at one seating position will be inappropriate for another.

Both the loudspeaker and the measurement microphone should be carefully positioned to avoid the creation of very early high-level reflections.

SPECIAL EFFECTS?

The “floor bounce” effect is a common example of a very early reflection (typically within a few milliseconds of the first sound arrival) that produces a unique acoustic response for each listener seat for all but the lowest octaves of the spectrum. This is an example of “less is better” when measuring the response, as a 1/3-octave display lacks the resolution to observe the effect in detail and produces less of a temptation to “fix” it.

The floor bounce effect can be minimized by use of an appropriate frequency-dependent time window or by simply laying the measurement microphone on the floor, or on a board placed across the listener seats. The effect usually disappears with the presence of an audience, so we do not wish to consider it when tuning the sound system.

The use of variable-length time windows and the synchronous transfer function allow the system to be tuned in a manner similar to the near-field RTA method (flat response on a log frequency display), even at remote positions in the room. It is superior to the RTA method in that the effects of air absorption are readily apparent and can be compensated for via equalization. Near-field techniques do not include air absorption for the simple fact that the sound has not traveled very far before it is picked up by the microphone, so it hasn’t passed though enough air to be significantly attenuated.

By far, the biggest problem with tuning sound systems is failure on the part of the technician to recognize anomalies that cannot be corrected with equalization.

The equalizer is a “global” device, meaning that its response curve will be impressed on all of the sound radiated from the loudspeaker, regardless of the direction in which it is radiated.

Many, if not most, of the anomalies observed on the analyzer are unique to each listener position. The technician must learn to recognize and ignore such events. They include:

—Floor-bounce effect

—Interference between multiple drivers

—Reflections from objects near the mic or loudspeaker

Events that produce a more global effect, and can therefore be addressed with equalization include:

—Boundary-loading of loudspeakers

—Coupling between multiple low-frequency drivers

—The direct-field loudspeaker response

With training and experience, the system technician can implement methods that reveal system imperfections that are correctable, and hide those that are not - regardless of the analysis method used. Better yet, system designers can design systems with fewer “un-equalizable” problems.

BAD IS ALWAYS BAD

The old adage “An ounce of prevention…” could never be truer.

System equalization then becomes meaningful and fast, providing the “icing on the cake” of the performance of a sound system.

It makes a good loudspeaker sound better, and brings the system to its fullest potential given the acoustic environment into which it is placed.

A bad room is a bad room, regardless of how we process the electrical signal that drives the loudspeakers.

When used properly, the traditional 1/n-octave real-time analyzer is a useful tool outdoors at any distance. Indoors, the effects of reflected sound and non-frequency-uniform room absorption produce some problems for this method at measurement distances remote from the loudspeaker.

One solution is to utilize a weighting curve that reduces the target level of the high-frequency portion of the spectrum. Attempts to achieve a flat system response at remote listener positions without the use of a weighting curve can result in harsh-sounding systems and even component damage.

Transfer function analysis addresses some of the shortcomings of the 1/n-octave RTA, but it requires greater expertise on the part of the user. Failure to properly compensate for the time differential between the reference and measured signal can produce wildly erroneous results.

The time window length must also be selected by the user, and different lengths will produce different displayed responses. A frequency-dependent time window produces a display that correlates well with human perception.

The most important feature of either measurement method is a knowledgeable operator - one who understands the caveats of each approach along with the basic characteristics of the human auditory system.

None of the questions raised here have a single, correct answer. This means that experience, good judgment, and common sense rooted in Newtonian physics are still a part of the measurement process.

Sound is a relatively easy quantity to measure, but measurements that correlate with human perception are much more difficult. Analyzers driven by omni directional microphones do a poor job of emulating the human listener. At this point one could ask, “So why measure at all? Why not just listen?”

Next time, we’ll have a look at this provocative question.

Pat and Brenda Brown own and operate SynAudCon, the leading independent professional audio education source, with training sessions held around the world and online. For more info go to www.synaudcon.com.

 

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Hosa Technology Introduces Second-Generation Elite Series Microphone Cables

Hosa Technology has introduced a significant upgrade of the company’s popular Elite Series microphone cables, which are now available with Neutrik XX-Series connectors plus a new nylon webbing over the cable’s PVC jacket.

Available in both Lo-Z (XLR3F to XLR3M) and Hi-Z (XLR3F to 1/4-inch TS) configurations, the cables use 20 AWG Oxygen-Free Copper (OFC) conductors that reduce resistance in order to facilitate maximum signal transfer.

Polyethylene dielectrics reduce capacitance for enhanced high-frequency transmission, while conductive PVC reduces handling noise.

Further, a 95 percent OFC braided shield is employed for noise-free signal transmission. The PVC jacket is cut- and abrasion-resistant for added durability.

The Neutrik XX-Series connectors employ gold-plated contacts for corrosion resistance and superior signal transfer, and they also utilize a zinc die-cast housing for rock-solid reliability. Polyurethane “glands” prevent cable kinking for longer cable life and chuck-type strain enhances cable retention.

“These second generation Elite Series cables offer exceptional audio performance and have been field tested for maximum reliability,” says Jonathan Pusey, Hosa Technology vice president of sales and marketing. “With a rich feature set and a highly competitive price, we are extremely optimistic that this product line will find favor with a wide range of customers.

“The Elite Series has always been an exceptional product and now, with the addition of Neutrik connectors and abrasion-resistant nylon webbing for a lifetime of dynamic, noise-free sound quality, I’m confident these mic cables offer the best possible combination of performance and value.”

The Hosa Elite Series microphone cables are available in lengths from 3 to 100 feet and carry MSRP pricing that ranges from $42.75 – $203.55. The Elite Series is slated to become available late Q2, 2012.

Hosa Technology

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Friday, April 13, 2012

New Audio-Technica Video: “Get Your Own Mic” Part Deux

Audio-Technica is at it again. providing another humorous public service announcement that graphically illustrates why NOT to share vocal microphones. Like the first installment (see it here), it’s the brainchild of Gary Boss of A-T.

 

Audio-Technica

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Shure Mics, Wireless & IEM Systems Heavily Utilized At 2012 JUNO Awards

Honoring the best of Canada’s music industry, the 2012 JUNO Awards in early April at Scotiabank Place in Ottawa utilized numerous Shure microphones, wireless systems and IEM systems for live performances, as well as host William Shatner and other applications.

Transmission Squelch was charged with the operation of dozens of wireless systems utilized on the broadcast w. The wireless needs for this year’s show included 42 microphones, 28 in-ear monitoring systems, and 28 intercom systems, including 8 channels of Shure Axient wireless microphones and 18 channels of Shure PSM 1000 in-ear monitors, supplied by Solotech.

Christian Pageau, co-owner of Transmission Squelch, was impressed with the Axient system, which utilizes frequency diversity and advanced interference detection technology. “Solotech offered the opportunity to try the Axient and I was very interested, so we agreed to try it during rehearsals,” he notes. “We did a lot of tests during the week, and it worked really well. In fact, it was my decision to use it in the show, and to put it on the main host.

“The filtering on the receiver is really good, so you can put more frequencies in the same range,” Pageau continues. “I also love having remote control of all the transmitter functions. To change the gain, or mute the RF, or even change the frequency at a distance, is incredible.”

During the telecast, Pageau dedicated two Axient channels to host Shatner, with additional channels assigned to the award presenters. “The two channels for the host were in frequency diversity mode, which was set up to prompt me if a frequency problem was detected,” he says. “They can also be set up for automatic operation, but I prefer to keep my hand on the steering wheel.””

For the show’s musical performers, Solotech provided 24 channels of in-ear monitors, 18 of which were Shure PSM 1000, which offers diversity reception. “Transmission Squelch had tested the PSM 1000 previously on a show in Montréal called Star Academy, and the results were very good,” reports Pageau. “So we asked Solotech to supply them. The diversity worked well everywhere in the venue, even backstage. Everyone was very happy with the system.”

To make all the on-stage wireless work smoothly along with the intercom systems in use by the crew, Pageau used frequency calculation software developed by a Canadian company EazyRF. He also designed and installed the antenna system for the show.

“We ran all the systems for the show with only four antennas – two to receive and two to transmit,” he states. “It was a full active system. For the reception, I was doing a star system, going through a distribution amplifier. The antennas were about 35 feet above the stage on either side. The transmitting system was active combining, with one helical antenna for intercom and another for all the in-ears.”

Shure Wireless Workbench 6 software helped keep the systems organized, to complement by an Anritsu Spectrum Analyser. “Workbench 6 was a beta version of the software for the Mac, and it worked perfectly,” Pageau notes. “I use it two ways. I use it to transfer labeling data, which is quicker than dialing each name on each receiver. Then, of course, I use it to monitor the systems during the show.”

Shure

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