Monday, November 23, 2009

Harman Professional Announces End-Of-Year Equipment Rebates

Applies to select JBL, AKG, Soundcraft, dbx, Lexicon and Crown Audio products

Harman Professional is offering including rebates, bundle offers and added values on select JBL, AKG, Soundcraft, dbx, Lexicon and Crown Audio products.

According to Scott Robbins, Harman Professional, Vice President of Sales, the rebate offer is intended to support loyal retailers during a challenging season and provide end customers with a remarkable opportunity to access the same technologies used by the production elite.

The full roster of special offers and rebates is available here www.harmanrebates.com/

For example, one package includes a $100 discount when buying one Lexicon Lambda USB interface, one pair of JBL Control 2P Studio Monitors, and one AKG Perception 120 USB microphone.

In another package,  Lexicon teams with AKG to offer a free D 5 microphone with the purchase of an I-ONIX U22, I-ONIX U45, or an IONIX U825 recording interface.

The other Harman brands also have similar deals. JBL and AKG are offering a package deal of one pair of JBL LSR2328P studio monitors with a free set of AKG 240 studio headphones. Soundcraft is giving a $250 discount with the purchase of a Soundcraft MFXI20 Mixer with an AKG WMS 450 wireless system, and Crown is offering similar deals with $75 off the XLS 602, $100 off the XLS 802, and $150 off the XLS 50000.

Harman Professional Website
Harman Rebates Website

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Posted by Keith Clark on 11/23 at 11:20 AM
Live SoundRecordingChurch SoundNewsProductAmplifierAudioBusinessLoudspeakerMonitoringProcessorSound ReinforcementStudioWireless • (0) CommentsPermalink

Production Rental Company AGC Ehitiki Brings First DiGiCo SD8 To Cyprus

“There was a clear market demand for us to offer DiGiCo consoles to our clients, so this was the ideal time for us to buy our first one." - Loris Demetriades, AGC Ehitiki

AGC Ehitiki, the largest production rental company in Cyprus, celebrated it’s 20th anniversary with the acquisition of a DiGiCo SD8 console.

Based at large premises in Kaimakli, close to the center of the island’s capital city of Nicosia, the company was founded by Loris Demetriades, who still takes a very hands-on role in the day-to-day running of the business.

“There was a clear market demand for us to offer DiGiCo consoles to our clients, so this was the ideal time for us to buy our first one,” Demetriades says. “The SD8 is also way ahead of any other digital console within its price range.”

The SD8 made its debut recently as the monitor board for a show by popular Cypriot singer Michalis Hatzigiannis at the 23,000-capacity GSP Stadium in Nicosia.

“It is a measure of our confidence in the SD8 that we were more than happy to give it its debut on such a massive show,” states Demetriades. “Michalis is a huge star in Cyprus, it was the final show of his summer tour and so we had to be absolutely certain that everything would run smoothly.”

Michalis’ Monitor Engineer George Panolaskos was also more than happy to work with the SD8, and the show was a complete success.

“The SD8 is a very easy to use, very flexible desk,” concludes Demetriades. “Sonically it’s excellent, with plenty of headroom, and the effects are brilliant. Personally I don’t see any reason for patching outboard units.

“It’s fully booked for all the future shows we have coming up and I haven’t had a single ‘no’ or complaint from sound engineers I have recommended it to.”

DiGiCo Website

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Posted by Keith Clark on 11/23 at 10:08 AM
Live SoundNewsConcertConsolesDigitalSound Reinforcement • (0) CommentsPermalink

Clear-Com Debuts New Two-Channel Model In Tempest Wireless Intercom Series

New model offers same RF interference resistance and performance as original

Clear-Com Communication Systems has announced the worldwide launch of its new two-channel Tempest wireless intercom system, the most recent addition to the company’s Tempest 2400 2.4 GHz wireless intercom family.

The two-channel system is an affordable option for those requiring fewer channels of communications but still offers the same license-free operation, exceptional RF performance and rich feature set as the four-channel Tempest2400.

As with all members of the Tempest family, the two-channel Tempest2400 utilizes a patented Frequency Hopping Spread Spectrum RF scheme that not only avoids the need for licensing and frequency coordination, but makes it fundamentally resistant to interference from other wireless devices.

This ensures that no matter how crowded the RF environment, communications will go off without a hitch. 

In addition, once registered to a base station, a beltstation needs no further configuration and roams freely within a single coverage area, making it perfect for theatrical productions, concerts and other highly coordinated events.

“Our new two-channel Tempest2400 is ideal for customers who simply don’t require the capacity of the four-channel system, but still want all the benefits of its comprehensive feature set, such as the wireless ISO and Stage Announce features,” says Chris Barry, Product Manager, Clear-Com.

“Its simple set-up requirements and portability also make it a tremendous asset for those looking to use it for multiple types of mobile applications, so it’s a win-win for the customer in all aspects.”

Each Tempest2400 base station supports up to five full-duplex, two audio-channel digital wireless beltstations; by stacking up to 10 base stations together, 50 independent, full-duplex wireless beltstations can operate together in a single system.

Other features include advanced 2xTX Transmission Voice Data Redundancy and its interoperability with Clear-Com and other intercom systems through two-wire and four-wire connections.

Additionally, the iSelect roaming feature allows a beltstation user to move from one coverage area to the next by simply changing the beltstation’s association to another base station; each beltstation can be paired with up to 64 different base stations. 

Each Tempest2400 wireless system includes T-Desk software, a powerful PC-based control and configuration application. T-Desk offers the ability to monitor and manage the entire wireless system from a remote location via Ethernet connection to a LAN.  A PC running T-Desk can also interface with a base station via direct connection over CAT-5 cable.

All members of the Tempest2400 family include an optional remote transceiver that allows remote antenna placement up to 1,500 feet (450 meters) from the base station; data connection and power supplied over standard CAT5e/6 cable with lightweight, fast-charging Lithium Polymer (Li-Poly) battery technology in the beltstation; access to one of six relay closures from the beltstation and vibrating call alert.

In addition, a flexible battery solution allows beltstations to operate on standard AA batteries for emergency use; beltstation battery telemetry is displayed in real-time on the beltstation and base station LCDs, indicating how much power is left in each pack.

Clear-Com Website

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Posted by Keith Clark on 11/23 at 08:47 AM
Live SoundChurch SoundProductDigitalInstallationSound ReinforcementSystemWireless • (0) CommentsPermalink

Friday, November 20, 2009

In Profile: No Doubt Engineer John Kerns & His Work On The 2009 Concert Tour

Veteran mix engineer John Kerns spent the summer of 2009 on the road with No Doubt, and the following Q & A provides background on John as well as his work with one of the summer's largest concert tours

Q. How did you get started in the business?
John Kerns: I played in bands when I was young.  We owned our own small PA and then I moved to California and started doing studio stuff for a little while. I just sort of lucked into the live thing.

Q. How long have you worked on A-level tours like No Doubt, doing big stadium shows?
JK: Since 1986 or 87 for arena/shed type tours. This is my fourth tour with No Doubt, and I’ve also worked with Sum 41, Avril Lavigne, Rogue Traders, Bruce Springsteen, Shania Twain, and Stevie Nicks, among others.

What have been the biggest advantages of moving from analog to digital (consoles)?
Obviously, the ability to recall everything is the first and foremost advantage, especially with all the traveling and waiting around we do.

Though it actually wasn’t until three or four years ago that I ran into digital desks that I really liked. That’s what facilitated the actual jump over to digital. 

How do you approach mixing from a creative standpoint?

I really just try to reproduce the band, but in a live setting. Obviously, there are certain bands that have [elements] on songs or records that I try to make come across live.

But all of the bands are so different. Mixing No Doubt versus Sum 41 vs. an Australian band I work with –they are all different acts, so I approach them all differently.

Yet, I still want everyone out there in the crowd to say, ‘Wow, you know that band was great. They played great. They sounded good.’ They don’t have to say that the band sounded exactly like the record.

John Kerns (click to enlarge)

I don’t think people really come to just hear the CD played really loud. They want a unique sound that you can only get by physically being there.

How does the ability to recall a whole show on the fly affect the way you mix?
It really hasn’t affected the way I mix at all. It certainly has affected all the traveling and all the one-offs. For Sum 41, we did shows all over the world, and it was usually on budget.

So being able to take just three rack spaces and a USB key is a lot easier than flying a bunch of hardware over there. It’s a lot easier than it ever used to be.

You are using a (Digidesign) VENUE system now on your shows. Do you recall what the learning curve was on the first show you mixed on it?
It was easy. It’s the easiest of all the digital consoles I’ve used. It’s laid out just like an analog console, except you are working on one channel at a time instead of a bunch.

click to enlarge

I didn’t delve into plug-ins too much to start. Everything on the console surface itself is totally usable and allows me to work the way I have always worked.

How’s the reliability of the VENUE?
It’s been great. In my entire experience, I’ve had one minor hiccup, and I’m not even sure it was related to the VENUE. I believe it was due to some conflicting authorizations.

Has your soundcheck process improved?
VENUE saves me a ton of time in sound checks. There are a lot fewer things that you have to worry about. You don’t have to worry about whether a cable is working or not. You don’t have to rely on whoever your local system guy is. And when you travel it is so easy. Everything is in one package.

What plug-ins do you use?
For reverbs, I like the TC (Electronic) stuff. I like the TL Space. I use CraneSong Phoenix, and use a bunch of McDSP stuff. I find it’s really good. I also use Analog Channel and Channel G, and a lot of it is to try to emulate old pieces of gear.

Though if I had to do a show with just the surface stuff and not a single plug-in–other than reverbs and delays– with keys and compressors, we’d still have something there.

Some engineers go back and look at a list of what plug-ins were used on different records or what analog gear was used on records, and they’ll try to recreate that. Is that your approach?
I certainly don’t want to take anything away from that at all, but that’s not my gig. To mold them isn’t my gig.

When you look at the stage you see what everybody is playing; you can hear what everybody is playing. I don’t know if organic is the right word, but my approach is pretty straightforward.

Have you delved into the recording capabilities (of VENUE) at all?
Yes, that’s how we record our show. We just use the recordings for sound check the next day.

For the uninitiated, can you explain what virtual sound check is and how it works?
Basically, you are just recording your previous show straight off the preamp heads and when you switch over your console into HDX mode the next day, you can play back.

As long as your input and output assignments are all correct, you’re playing back through your actual channels with a real dynamic performance from the band.

Short of leakage into any vocal mics, you are about as close as you can be to exactly what your band played or what your stock sound should be at the start of your sound check.

click to enlarge

How do the bands react to the kind of new workflow that virtual sound check enables?
Bands dig it. They love it. Because they can come in and instead of me asking them to stick around and play three or four things, they can come in and run half a song and as long as they’re good, you know, as long as their mixes are good, we’re set. It makes life easier on everybody.

I think when you get up to this level, everyone is pretty open to new technology and they probably all know more about Pro Tools than I do. They can probably teach me things.

Can you talk about how VENUE works in a festival environment?
It works great. In analog world, you have 60 or 70 inputs, you are still dialing knobs, looking at pieces of paper to see what you had on the last show. VENUE is a lifesaver.

It’s not the be-all-save-all, however. You’ve still got to use your head about it. If you come in and you have 300 different plug-ins that you use, you better make sure that they are in the console because you are not going to have time to load them.

It won’t do your job for you, but it will make it a lot easier.

This is one of your final nights and a hometown gig in Los Angeles. Does that feel like a special experience for the band?

It may be for the band. It’s just another show for me. I have the same approach to every show.

You can’t not give it your all just because you aren’t doing a show in a good market. It’s the same show.

But it is a line of thinking, definitely a prevalent line of thinking. You always get the ‘Ah, LA, it’s a big show.’

But I think to myself, ‘Oh, really…did three more trucks show up? It looks like the same stuff to me!’ It doesn’t matter whether we are doing a club gig in EL Paso or doing four nights here in their hometown.

The whole crew is the same way. No matter what show we are doing, everyone brings their A game every time. If they don’t want to do it every day, then why do it?

 

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Posted by Keith Clark on 11/20 at 09:46 AM
Live SoundFeatureConcertDigitalEngineerMixerSound Reinforcement • (0) CommentsPermalink

Lab.gruppen Joins Open-Platform Crestron Integrated Partner Program (IPP)

The new Crestron module features full amplifier fault reporting, muting, and soloing at the per-channel level, as well as power on/off and real time metering

Lab.gruppen is now a fully certified member of the Crestron Electronics Integrated Partner Program (IPP), following the co-development of a Crestron Integrated Partner control module which allows Lab.gruppen amplifiers and Lake products to operate seamlessly with other hardware in an open-platform Crestron systems environment.

IPP applies to audio/video, security, HVAC, lighting and other controlled devices that may be incorporated into commercial systems.

The module is a custom control protocol developed for Lab.gruppen by Crestron and is available at www.labgruppen.com for Lab.gruppen C Series and FP+ model ranges, and will be available shortly for the PLM Series of Powered Loudspeaker Management systems that feature Lake Processing inside.

Access to the C and FP+ Series amplifiers is provided via the NLB 60E NomadLink Bridge & Network Controller, which allows control and monitoring of up to 512 amplifier channels within a single network.

The Crestron module features full amplifier fault reporting, muting, and soloing at the per-channel level, as well as power on/off and real time metering, of both individual channels or user-defined groups.

According to Lab.gruppen product manager Doug Green, the new Crestron module is an essential tool for all Lab.gruppen users working in the installed sound sector.

“With our entry into the Integrated Partner Program, our products can be seamlessly integrated into any Creston-based, and Crestron-controlled system anywhere” said Green. “As one of the world’s leading providers of third party control hardware and touch panels to the installed sound sector, this is a great way of making Lab.gruppen products even more usable.

“It’s also good for Crestron, as it offers them greater choice of hardware when developing their network solutions.”

“Installed systems are growing ever more complex,” continued Green, “and often the biggest headaches faced by system designers and specifiers are about getting pieces of equipment to communicate and integrate effectively with each other.

“Ideally we’d like as many manufacturers as possible to be able to control our products – indeed, we already have a generic version of our own protocol on our website that allows users to custom design their own control software so that they can incorporate our products into their systems.

“Crestron’s control module is another part of the puzzle, and particularly important for large scale, complex projects that require remote control, fault monitoring and integration with Voice Alarm systems. Crestron is already in the process of developing a similar module for PLM, which we look forward to announcing in the very near future,” he concluded.

Lab.gruppen Website

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Posted by Keith Clark on 11/20 at 08:35 AM
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Heavy Usage Of Shure Microphones, Wireless At 10th Annual Latin Grammy Awards

The show show featured 15 live performances by some of the biggest stars in Latin music, the vast majority of whom used Shure microphones for their vocals and backline

Shure microphones and wireless systems were in heavy rotation at the recent 10th Annual Latin Grammy Awards Show in Las Vegas, where Calle 13, a Puerto Rican alternative/urban hip-hop duo and Shure endorser, won all five categories in which they were nominated.

The awards show, televised to more than 12 million viewers on Univision, took place at the Mandalay Bay Events Center in Las Vegas, featured 15 live performances by some of the biggest stars in Latin music, the vast majority of whom used Shure microphones for their vocals and backline.

Performance highlights included Mexican singer Juan Gabriel, who was honored as the Latin Recording Academy’s 2009 Person of the Year. Using a black Shure UR2/KSM9 expanded a planned ten-minute medley into a full concert set.

The broadcast music mixer was Eric Schilling, noted for his studio engineering for artists like Gloria Estefan and Shakira who has a long track record of mixing award telecasts.

As the night’s biggest winners, Calle 13 took home awards for Record of the Year (the single “No Hay Nadie Como Tú” featuring Mexican band Café Tacvba) and Album of the Year (Los De Atrás Vienen Conmigo).

The duo, consisting of singer/songwriter René Pérez (a.k.a. Residente) and multi-instrumentalist Eduardo José Cabra (a.k.a. Visitante), also won the categories of Best Urban Music Album, Best Alternative Song, and Best Short Form Music Video.

Another highlight of the evening was Calle 13’s collaboration with Rubén Blades for an elaborate staging of the group’s hit “La Perla,” which received the award for Best Short Form Music Video.  The number began with a huge set of Japanese taiko drums played by performers from Cirque du Soleil’s “Mystère” troupe.

“That performance spanned two full stages,” noted Schilling. “It started with the taiko drums, which are physically huge, on one stage. Then the band started playing on top of that from the other stage, then finally Residente and Rubén Blades came in with their Shure wireless Beta 58s. It looked great and sounded fantastic.”

While Schilling has his preferences, the choice of lead vocal mics is left up to the performers as a matter of policy. “We want them to be in their comfort zone as much as possible,” said Schilling. “But if there’s no strong preference, I will usually suggest either a KSM9 or a Beta 58, depending on the artist. This year, I think all but a couple of the live performers went with Shure, and the results were outstanding.”

Schilling also used a lot of Shure mics on the backline. “Typically in our shows, we use Shure mics for 60 or 70 percent of our backline. On drum kit, I like to use KSM32s for overheads. That’s a great mic. Of course, I use the SM57 on snare, and I like the KSM137 for hi-hat. The kick gets a Beta 91 boundary mic and a Beta 52 dynamic, which lets the house PA mixer, Ron Reeves, and I to get the right sound for our different needs.”

Of course, Latin music uses a lot of percussion. “With a couple little exceptions, all the percussion is miked with Shure. I love SM57s on congas and bongos,” said Schilling. “On the rest of the backline, I use either KSM137s or KSM32s for all the guitar amps. I use the same mics for saxophones, too.”

To prepare for the live broadcast, three days before the show are dedicated to rehearsals. “We get about an hour with each live act, and I tape everything so I can work on each mix before the show,” said Schilling. “It’s all recallable on the console, but that just gets me into the ballpark.

“After a band is playing live, they’re never quite the same as in rehearsal. So there’s constant fine-tuning. Plus, there’s always something unexpected. Having mics I can count on is a key element.”

Shure Website

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Posted by Keith Clark on 11/20 at 08:20 AM
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Barry’s Toolkit: First Look At The New Lexicon Native PCM Reverb Plug-In

Barry provides detailed look and hands-on review of Lexicon's Native PCM Reverb plug-in bundle, the company's long-awaited entry into a software-based reverb

The Lexicon Native PCM Reverb plug-in offers great, usable factory preset reverb sounds and, at the same time, “deep as you want” intense programmability for modifying or designing your own unique reverberant treatments.

Programming is now more precise due to the vast collection of algorithmic specific parameters and the three-dimensional RTA (real-time analyzer) color display that allows the visualization of the reverb’s shape, character and evolution over time.

Ported in VST, AU and RTAS Native formats for both MAC and PC, this bundle consists of seven plug-ins each using its own specialized reverb algorithm.

The seven reverbs are categorized as: Chamber, Concert Hall, Hall, Plate, Random Hall, Room, and Vintage Plate. Because each reverb plug-in has a dedicated “engine”, each comes with its own set of relevant parameters that are fully adjustable within the intuitive GUI.

Yet the interface’s look, feel and operation remain consistent across all seven plug-ins. For any one who has ever used the LARC to control either a Lexicon 480L and 960L reverb,

Lexicon Native PCM is an immediate “install and use” reverb - for the most part—no manual reading required. For those new to reverb tweaking will find the most important, salient parameters immediately available and easily adjustable.

The manual gives excellent descriptions of all the parameters and how they change the sound of the reverb.

It’s All In The GUI
Once you’ve decided upon and instantiated one of the seven reverbs best suited for your application, you must define or specify its exact nature using the Category pull-down at the top of the GUI.

The Chamber plug-in has subcategories called small medium, and large chambers; Concert Hall reverb is divided into Rooms, Small Halls, Medium Halls, Large Halls and effects; Hall has Small Spaces, Small Halls, Medium Halls, Large Halls, and Huge Halls.

Sample graphic: Lexicon PCM Plug-in Concert Hall Multiband Bandpass. (click to enlarge)

The Plate plug-in has Small, Medium, and Large Plate Reverbs, and Random Hall offers Small Spaces and Small, Medium, and Large Random Halls.

Lexicon’s Room reverb provides Small, Medium, Large Rooms, Drum Room, Small and Medium Halls, Large Halls, Exterior Places, and Effects.

Finally, the Vintage Plate reverb contains plates designed for Instruments, Vocals, Live Sound, Drums and Percussion uses.

Each of these subcategories of reverbs comes with 50 or more evocatively named factory presets sonically tailored in very specific ways using all the different combinations of that reverb’s individual set of parameters.

All parameters are available for editing and all are automatable within your DAW’s automation system.

Sample: PCM Plug-in Concert Hall Multiband Notch. (click to enlarge)

Below the Category and Preset pull-down menus are stereo I/O level meters, two equalizers available for both the Early Reflection signal and the Tail portion of the reverb, and a real-time audio analyzer display.

The I/O meters are standard LED trees with a handy signal present indicator. The EQ section has choices of one-pole (6 dB/octave) or two-pole (12 dB/octave) filters in Lo Pass, High Pass, Bandpass or Notch topologies.

There is a graphical display showing the filter curve imposed on the early reflection sound, colored in blue, and the reverb tail EQ curve shown in red.

Real-Time
The real-time analyzer in the center of the upper half of the GUI is a spectacular, three-dimensional multi-band tool for actually “seeing” the reverb sound build up, play out and evolve over time. The time line runs from right to left with reverb level represented vertically.

Once a reverb signal is present, different color-coded audio waveforms parade across the screen. Each of the moving waves represents a frequency band from 50 Hz to 12.5 kHz with the lower frequency bands in the back and higher frequencies in the front.

I found the RTA edifying—perfect for adjusting the Early Reflection and Tail EQs, predelay, Bass RT, Reverb Out level and Size parameters.

Being a summed mono display, for stereo parameter settings rely on your ears, as you should for all adjustments. The RTA confirms what you are hearing and is mesmerizing too.

There are also two other ways to “read” the reverb sound: a more conventional and colorful 2-D spectrum analyzer bar graph and a linear moving amplitude waveform display. 

Sample: PCM Plug-in Hall Frequency LoPass. (click to enlarge)

The manual points out that these dazzling displays require additional host DSP so after impressing that client watching over your shoulder, save some DSP and turn them off before closing the plug-in’s window.

Reverb Controls
The remaining lower 2/3 of the GUI is devoted to the control faders. I found these faders to “mouse” very responsively without delay or glitching like other plug-ins.

Depending on the reverb plug-in you’ve called up, this section can be populated with up to nine parameter faders.

I liked that the most salient and useful faders are shown and for the most part, similar parameter faders from one reverb plug-in to another reside in the same slot—such as one of the most changed parameters—RT (reverb time).

Sample: PCM Plug-in Hall Impulse LoPass. (click to enlarge)

For the most part, there is little need to drill deeper than this page.

There are so many presets in each Category, I found toggling through them quickly gets you close to what you want and then a quick fader move or two gets you all the rest of the way there.

Each parameter fader position is named and has a value box for manual entry. Clicking on Edit allows any of the parameter faders to be re-tasked or changed to control any of the other available parameters by using the pull-down menu under the Modifier button.

So if you’d like the parameter Reverb Time to always be the first fader or any have other parameter(s) that does not normally show up here to be visible, you can make it so.

While in Edit mode, for deeper programming, the Soft Row becomes visible for selecting another row of parameters for access to more of the plug-in’s under-the-hood divinity.

Depending on the reverb plug-in instantiated - I’ll use the Chamber reverb as an example—the Soft Row will have the button names: Input & Mix, Reflections, Reverb and Echoes. When any of them are clicked, sub-parameters faders appear.

For example, when Input & Mix is clicked, Mix (wet/dry), Diffusion, Shape, Spread and Predelay parameters become available.

If applicable, additional buttons above these sub parameter buttons will show up.

In the above example, Predelay (as do all delay parameters in all plugs) has a toggle switch for either Absolute (the fader positional value) or Tempo-based from a 32nd note to a half-note value predicated upon the session’s tempo.

Sample PCM Plug-in Vintage Plate Display Off LoPass. (click to enlarge)

It sounds more complicated here than it really is.

Know that there is the good ol’ Compare button to show the preset default parameter settings if you get lost in a wilderness of bewilderment.

Lastly, the Lexicon Native PCM bundle has its own comprehensive interface for managing, naming, and saving modified user presets.

As user presets are accumulated, they are listed in the aforementioned Category pull-down menu.

User presets are stored within the plug-in itself, instead of in the DAW’s plug-in folder, as usually the case.

Sample: PCM Plug-in Vintage Plate Multiband Notch. (click to enlarge)

This means that they, along with all the Native PCM reverbs, are available for other programs in your computer. If you sequence in Logic and mix in Pro Tools, you can keep the reverb sounds consistent across both platforms.

You could also share them (as XML text files) with other systems or between Mac and PCs. (Now that is cool!)

Let’s Use This Thing!
I installed Lexicon Native PCM Reverb into a MAC PPC Quadcore running OS 10.4.11 and open it as an RTAS plug-in in Pro Tools HD session.

While the software runs fine in this old OS, it runs better in 10.5 or above. It authorizes via iLok and there are mono, stereo and mono (input)-to-stereo (output) versions available.

Great room sounds might be the ultimate quest - the ‘holy grail’ for high-end reverberators with percussive sounds the most challenging sound sources.

How does the reverb algorithm handle percussive attacks and simulate the thousands if not millions of reflections that happen in a real room? Does it ‘boing’ when a sharp-sounding snare drum is put to it?

In short, how realistic does a synthesized room sound?

I have to say I was impressed from the moment I first heard this plug-in at the recent New York AES convention - the true, great sound of a Lexicon room in a plug-in! I am so ready for this!

In Session
Here at my Tones 4 $ Studios, I was in the middle of a hard rock project at the last tweak stages and I wanted to replace other reverbs with Lexicon Native PCM to see if I could get something better going on.

Sample: PCM Plug-in Vintage Plate Frequency Notch. (click to enlarge)

The current pop/rock music aesthetic does not call for much long reverb treatment but the producer and artist wanted the drums to sound like they were recorded in a big room - which of course, they were not.

The first song moved at a fast clip - 153 BPM so I couldn’t add a giant room with a lot of lengthy aftermath. It had to build quickly and decay fairly quickly.

I chose the Medium Vocal Concert 1 in the Medium Halls category of the Concert Hall plug-in. I set Predelay to a 32nd note, Reverb Time to 2.3388 seconds, Reverb Out Frequency to 7125Hz, and Diffusion to 70 percent.

Tail Width sets the stereophonic width of the reverb’s tail and I narrowed it down to 38 degrees because the default wider stereo setting tended to wash the stereo drum stage out.

In the Reverb sub parameters, I set Bass RT to 2.75X because I put some of this reverb even on the kick drum. Size was set to 39 meters. The GUI looked like the Barry’s EXAMPLE 1 graphic here.

Barry’s EXAMPLE 1 graphic. (click to enlarge)

By adding a small amount of this reverb, I got drums to sound like they were playing in small hall or large reverberant club show room.

The drum sound was present, realistic and powerful sounding. While a drummer playing at this tempo cannot do many fills or anything else, whenever there were breaks in the music, the reverb tail sounded like room decay.

The second song was at a more stately tempo of 66 BPM—an old-school MOR ballad the producer wanted “big boy” reverb cake on the drums and most everything else.

I started with the Large Hall preset in the Large Halls Category in the Hall Reverb plug-in. I set Predelay to 16th-note and Reverb Time to 1.9801 seconds. Tail Width was at 99 degrees but check this in mono—you may not like what you hear and return it to normal stereo.

Reverb Out Frequency was 6,500 Hz but RT Hi Cut was at 6,250 Hz (default) and I also kept the default size to 32 meters, as shown in the Barry’s EXAMPLE 2 graphic below.

This sound, including the 16th note predelay, was perfect for this song. This Hall preset sounded huge with super clean tails that faded into the mix’s noise floor (what little there was).

I subsequently added another stereo reverb to fill in the 16th note predelay gap and the phasey tail out of the first reverb. I wish reverbs had a blend control where you could “leak” around the predelay section for filling in that space.

In both of these two brief examples, I got the reverb sounds very quickly - but, in fact, there are so many excellent preset choices with so many ways to “dial them in,” making decisions takes more time than actually getting a sound.

It’s easy to go very surreal with a lush, wonderful and huge reverb as in the case of the ballad, or more realistic with a tougher and harder sound I developed for the faster rock song.

Barry’s EXAMPLE 2 graphic (click to enlarge)

I found it super important to learn the exact nature of the seven reverb plug-ins - the Categories.

For me it was like learning the categories in the Lexicon 480L or 960L reverbs: once I developed a sonic familiarity of each, I usually “nailed” my initial choices first time rarely starting over by changing categories.

I have no other reverb, plug-in or hardware, with this much versatility - but then I don’t own any Lexicon reverbs except this one! If I were allowed only one reverb bundle, this would be the one for me!

I am anxious to get into my next mix project where I can start fresh and run several Lexicon Native PCM reverb plug-ins working together.

Lexicon Native PCM sells for $1,899.95 MSRP. Lexicon will start shipments of the plug-in to authorized dealers the first week in December. Purchase and download of the plug-in directly from http://www.lexiconpro.com will occur in January 2010.

For more information, go here, and also be sure to check out videos here.

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Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Visit his website at www.barryrudolph.com

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More Reviews & Articles By Barry Rudolph On PSW:
A Wide Variety Of Microphone Techniques For Recording Drums
The Tale Of A Project-Saving Monitoring Technique
Test Driving The Focusrite Saffire PRO 40 Firewire Audio/MIDI Interface
Rhythm Section Tracking In The Studio
Does The WAVES Hybrid Line Of Plug-Ins Enhance The Creative Process?
Creative Uses For Loudspeakers To Enhance Your Recordings
The Shure 55 Microphone Has Deep Roots, But How Does It Hold Up Today?
Thumbs Up Or Down For The Marshall MXL V89 Studio Condenser Microphone?
Inside The Peluso P12 Tube Condenser Microphone
Barry’s DAW Toolkit: Review Of The Novation Nocturn With Automap 3 Pro
Barry’s Recording Tips: Figure Of Eight Royer For Electric Guitars
Review Of The X-Tempo Pok DAW Wireless Footswitch Controller
Barry’s Toolkit Of Handy DAW Products
Recording Gear Hits At The 2009 Winter NAMM Show
Working At Recording Success: Taking Elemental Steps Can Make All The Difference
Recording Tip: Successfully Dealing With A Dead Room

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Posted by Keith Clark on 11/20 at 07:28 AM
RecordingFeatureProductReviewDigital Audio WorkstationsProcessorSoftwareStudio • (3) CommentsPermalink

Britannia Row Supplies Outline Butterfly For Return Of Robbie Williams At London’s Roundhouse

Butterfly has become the most specified system for the Roundhouse due to its high power and even dispersion together with its camera-friendly look

Britannia Row supplied Outline Butterfly VLA line arrays at the request of the BBC and London’s Roundhouse for the 2009 Electric Proms season, which recently kicked off with the return of Robbie Williams backed by a 34-piece orchestra plus his own touring band.

The system comprised 12 Outline CDH 483 hi packs per side, supplemented by three Mantas “wide boxes” in each hang, as well as 12 Subtech 218 subwoofers. Outline T9 amplifiers drive all loudspeakers, with control via Dolby Lake 412s.

Butterfly has become the most specified system for the Roundhouse due to its high power and even dispersion together with its camera-friendly look, which is essential to televised performances from the venue.

Derrick Zeiba was in charge of the orchestra, while Snake Newton served as Williams’ mix engineer.

“The addition of the Mantas wide element has meant that you no longer need in fill/out fill boxes - the 120 degrees produced by Mantas does everything that you need. This was proven to great effect especially in the Roundhouse, which can be very unforgiving,” Zeiba notes.

“It’s my favorite small system - very little system or channel EQ was used to get unbelievable results,” adds Newton.

The Electric Proms continued its season with performances by Dizzee Rascal, Dame Shirley Bassey, The Doves, and Smokey Robinson, all utilizing the Butterfly arrays.

Outline Website

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Posted by Keith Clark on 11/20 at 07:08 AM
Live SoundNewsPollAmplifierConcertLine ArraySound ReinforcementSubwooferSystem • (0) CommentsPermalink

Thursday, November 19, 2009

Brian Wilson Completing Unfinished George Gershwin Works At Ocean Way Recording

The newly completed works may remain as instrumentals or may develop into full-fledged musical pop songs

Beach Boys co-founder Brian Wilson is recording the music of George Gershwin at Ocean Way Recording in Hollywood. 

Wilson has been authorized by Gershwin’s estate to complete unfinished songs Gershwin left incomplete when he died in 1937. 

Initially, two songs are being recorded for a proposed album due out in 2010.

Gershwin’s great-nephew Todd Gershwin is a trustee of the family trust and has stated that both Gershwin and Wilson were musical visionaries known for combining various musical genres in their compositions. 

The newly completed works may remain as instrumentals, or may develop into full-fledged musical pop songs.

The Gershwin project was the result of overtures by Walt Disney Records president David Agnew, who proposed a two-album contract with Wilson. 

Both composers are known for their highly successful pop music, as well as for their explorations into more serious orchestral work.

Ocean Way Recording Website

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Posted by Keith Clark on 11/19 at 04:29 PM
RecordingNewsDigital Audio WorkstationsStudio • (0) CommentsPermalink

Yamaha Digital Consoles Cast For Stage Production Of “SAMMY”

Production's sound designers John Shivers and David Patridge incorporate the digital consoles

The Tony Award-winning Old Globe in San Diego, California is currently hosting the hit play “SAMMY” based upon the life of legendary entertainer Sammy Davis, Jr., the production’s sound design by John Shivers and David Patridge incorporating Yamaha digital consoles.

Masque Sound (East Rutherford, New Jersey) provided elements of the sound system for Shivers and Patridge. “It’s always a pleasure to work with clients as skilled and as organized as John and David,” states Scott Kalata of Masque Sound. “They are especially capable of designing well thought out systems that are extremely reliable and sonically pristine. They’re both right up there with the best sound designers in the business.”

A Yamaha PM5D digital audio console is ats the center of the sound system.

“We know the PM5D to be a quality digital mixing platform that presents a good combination of feature set versus economics. For shows on the scale of most regional theatres in the U.S., the PM5D offers a solution that has a reasonable footprint, is reliable, and extremely operator friendly, explains Patridge. “On SAMMY, we took advantage of the ability to cascade the inputs from a Yamaha MY16-AT card to buss the sound effects system into the PA.

“Sound effects for SAMMY were programmed using QLab V2 that we used as a master cue list to recall scenes in the PM5D. QLab’s audio was routed over optical from an RME Fireface 800 to the PM5D’s card slot, cascaded to the busses and, in turn, routed to various speakers.”

Patridge says that a Yamaha DM1000 digital console was paired with two AD8HR external preamp units to serve as a sub-mix for both drums and percussion as the sound design for SAMMY easily exceeded the inputs available on the PM5D.

“We like the higher quality microphone preamps available via the AD8HR and the ability to pair them with Yamaha’s product line of digital consoles using MY slots and serial control of preamp functionality is very convenient.”

Front of house mixing duties are handled by Old Globe engineer Erik Carstensen, the second project he has mixed for Shivers and Patridge. “Erik combines a good deal of industry experience as a sound engineer with a calm and good-natured demeanor - an excellent combination for a venue that could easily produce 15 different works in a season,” Patridge says.

“The selected channel programming and routing on the Yamaha PM5D is intuitive and flexible,” adds Carstensen. “I find that changing and storing EQ and dynamics settings, for example, is achieved quickly - vital in the production process where audio usually doesn’t get a lot of time to get it right. On top of that, using the PM5D in combination with our DME64 has been very reliable, show after show.”

With a cast of 16 of Broadway’s best singers and dancers, the world premiere musical, written by two-time Academy Award and Grammy Award winner Leslie Bricusse, depicts Sammy Davis Jr.‘s days as a child working in vaudeville through his time with Dean Martin and Frank Sinatra as a member of the Rat Pack.

Yamaha Commercial Audio Website

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Posted by Keith Clark on 11/19 at 10:55 AM
Live SoundNewsAudioConsolesDigitalMixerMonitoringSound ReinforcementStage • (0) CommentsPermalink

Klein + Hummel To Be Integrated Into Sennheiser & Neumann

"We would like to benefit from the strength of the Sennheiser and Neumann brands in this business area, too." - Volker Bartels, Sennheiser

Sennheiser electronic GmbH & Co. KG will integrate its subsidiary K + H Vertriebs- und Entwicklungsgesellschaft mbH, manufacturer of studio monitors and installed sound products, into the Sennheiser and Neumann companies.

“We would like to benefit from the strength of the Sennheiser and Neumann brands in this business area, too,” states Volker Bartels, speaker of the Sennheiser executive team. “Neumann and Sennheiser are firmly established in the studio and installed sound areas, respectively, and promise a much better market penetration than is at present possible under the brand name of Klein + Hummel.”

Georg Neumann GmbH, Berlin, will become responsible for the studio monitor business, while an optimized installed sound portfolio will be continued by Sennheiser.

“Under these well-known brands, we hope to considerably increase sales of these premium products,” says Bartels. “Neumann – as the studio brand – will expand its portfolio, and Sennheiser will further strengthen its installed sound business, which is the company’s third ‘pillar’ so to speak, besides the consumer and professional business.”

Wolfgang Fraissinet, president of marketing and sales at Neumann in Berlin, adds, “In its long company history, Neumann has set many milestones in studio technology worldwide. We will now also use this expertise for studio monitors and offer optimum solutions to our customers, to international artists in the areas of TV and radio broadcasting, recording and audio productions.”

Bartels estimates that the integration of Klein + Hummel will be completed by the new year. For the most part, the employees of K + H Vertriebs- und Entwicklungsgesellschaft will be employed by Sennheiser and Neumann.

On March 1, 2005 Sennheiser had taken over the areas of sales and development of Klein + Hummel GmbH, Ostfildern and thus the worldwide rights in sales, products and the brand.

Sennheiser Website
Sennheiser USA Website

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Posted by Keith Clark on 11/19 at 08:32 AM
Live SoundRecordingNewsAudioBusinessLoudspeakerMonitoringSound ReinforcementStudio • (0) CommentsPermalink

Fun With Studio Funding: Cash Poor But Laughing About It

Finding it tough to get enough cash to stock your home studio? Our man Jackson B. Jackson, president of the “JBJ First National Bank,” offers financial advice virtually guaranteed to get you nowhere - except perhaps into a better mood!

To: JBJ First National Bank

Dear Sirs:

I am just starting into the recording business and need financing to build a studio and buy equipment. Could you please send me information on your financing packages, what your current rates are, and if there points involved? Zero down and 0 percent would be nice.

Thanks,
Swap Happy

P.S. I plan on quitting my job after you lend me the money to make time for running the studio, so could you please make the terms for 30 years?

——————————————————————————-
Dear Mr. Happy :

Here at JBJ First National Bank we strive to help small business owners achieve their dreams. We have many lending plans that may suit your needs. Please take a moment to browse through our selection of small business loan packages.

1. “Zero down, Zero payments, Zero money”
That’s right, you pay NO down payment, NO additional payments, and we give you NOTHING in return! Simply make an appointment with one of our qualified lenders, come down to one of our branch offices, and you will be promptly ignored.

2. “Five Finger Plan”
In building any small business we recognize that supplies are needed. With our “Five Finger Plan” you can simply go to the appropriate supplier (in your case, your local pro audio shop) and ask for Larry. When the sales person goes to the back of the store to look for anyone named Larry, simply place any needed merchandise in your pants and exit the store quickly. If anyone should ask you what you think you’re doing, or (for reasons unknown), accuse you of “stealing,” simply notify them of your FFP loan with JBJ First National.

3. “Audio Engineer Special”
This package was designed specifically for home recording and project studio owners.  We loan you enormous amounts of money filtered through a special broker called “Your Wife”. “Your Wife” will tell you EXACTLY what you can and cannot spend on your studio. This is a very popular package, and it isn’t hard to find professionals in your field who have recently signed up for this plan or who have had a contract for years!

Finally…

4. “The NO Limit Plan”
How much we’ll actually lend you is irrelevant, because as soon as you spend what you’re given, your gear will simply not be good enough for you any more. The equipment you wanted so much yesterday… now that it’s actually made it into your hands, it will simply lose it’s appeal. But this is NOT a problem! We’ll simply loan you more money so you can go out and buy more gear in hopes that those shiny boxes with the pretty flashing lights will cover up the fact that you have no idea what you’re doing in the studio!

I’m sure you will find a package suitable for your needs, and all of us here at JBJ First National look forward to helping you build a great big financial hole in your life!

And by all means, quit your day job today! You, my friend, are on your way!

Insincerely,
Jackson B. Jackson
President and CEO of JBJ First National Bank

More articles by Jackson B Jackson on PSW:
The Guide To Better Mixing Through “Mixing”
Watch Out For The Tall Tale: “We’ll Fix That During Mastering”
Keeping Fresh The Art Form Of Recording

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Posted by Keith Clark on 11/19 at 06:27 AM
RecordingFeatureOpinionPollAudioBusinessDigital Audio WorkstationsStudio • (0) CommentsPermalink

Wednesday, November 18, 2009

Don’t Miss The New PSW Video Series: Introduction To Compression

The first in PSW's video educational series

ProSoundWeb has just launched the first in what will be a fast-growing series of instructional videos for audio professionals and neophytes alike.

The first program, Introduction to Compression, covers the necessary basics for success, as well as the keys to effective uses of compression, presented in 12 easy-to-digest entries.

All 12 installments are easily accessible via our video player here.

Don’t miss it!

 

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Posted by Keith Clark on 11/18 at 02:58 PM
Church SoundNewsVideoAudioDigital Audio WorkstationsEducationProcessorSoftwareStudio • (0) CommentsPermalink

Ashly Processors And Amps Help Boston Landmark Improve Sound And Save Energy

Audio/video integrator Balanced Input (North Attleboro, Massachusetts) helped "green" the Park Plaza by replacing the old 96-space rack system with a much smaller system centered on energy-conserving, and using great-sounding Ashly DSPs and amplifiers.

The Boston Park Plaza Hotel and Towers was recently inducted as a member of the prestigious Historic Hotels of America. In addition to its upscale accommodations, the Park Plaza is well-known for its six ballrooms: the very large Imperial and Plaza ballrooms, and the smaller Georgian, Arlington, Berkeley, and Clarendon ballrooms. The Park Plaza is an ideal location for meetings and parties of all kinds.

Recently the hotel had grown increasingly frustrated with the audio system that served the ballrooms. A pair of seven-foot racks housed enough power amps to fry a dozen eggs while idling, and their opaque user controls flummoxed even the most audio-savvy hotel staff.

At its root, there was no core control and no time alignment, equalization, or limiting.

Audio/video integrator Balanced Input (North Attleboro, Massachusetts) helped “green” the Park Plaza by replacing the old 96-space rack system with a much smaller system centered on energy-conserving, and using great-sounding Ashly DSPs and amplifiers.

The original audio system was installed in 1996. Mark Waker, principal at Balanced Input, consulted with the hotel and offered a plan for a system that would give them ease of operation, while giving them a substantial energy savings.

“The bones of the system were quite good, EAW boxes and Electro-Voice ceiling speakers, some tweeters needed but nothing major,” Waker admitted, “but most of the amps had issues and the processing was simply missing.

“I knew we could save the speakers but the system desperately needed modern control and amplification to deliver.”

The hotel is configured with one equipment closet for the two larger ballrooms and a second equipment closet for the remaining four smaller ballrooms. Each of the closets contained a seven-foot, 48-space rack. Between them, ten amplifiers powered the speakers.

“The fact is, these systems idle most of the time,” said Waker. “They don’t do much most of the day, but then when they have to work, well… they have to work!

“At any rate, they spend most of their time using energy to make heat, which the building’s A/C system then has to use energy to cool down. With power-hungry amps in there, it was a big waste.”

To each of the replacement racks Waker added an Ashly ne24.24M Protea DSP, “fully loaded” with modular expansion cards to offer a tremendous amount of processing power within two rack spaces.

Each of the larger ballrooms received an Ashly ne8250 eight-channel power amp, whereas one ne8250 served all four of the smaller ballrooms. In addition, mp3 players and power conditioners added features that were entirely absent from the older system.

Despite this, the rack size for the larger ballrooms shrank from 48 spaces to just twenty. Even more impressive, the rack size for the smaller ballrooms shrank from 48 spaces to just eight.

“Ashly amplifiers are ‘energy savers,’ but unlike most other ‘energy savers,’ they actually sound good,” said Waker. “We wouldn’t use them if they didn’t. In fact, we use Ashly PE Series amps and Ashly Protea processors in our live rig.

“In addition, Ashly products are resolutely reliable, which greatly reduces service calls - a benefit to both the hotel and us, as getting into Boston is no picnic! All that power and reliability comes with a very reasonable price tag.

“In fact, they helped us to come in below the figure that the hotel had budgeted for, so we helped the hotel spend that money on LED lighting.”

The old systems used to draw in excess of three amps while idling, more in the case of the bigger system. The new systems each draw less than one amp while idling. That represents a conservatively-estimated 60% reduction in power consumption. In addition, with Ashly processing, any hotel staff member can use the system.

“They just plug in a mixer or a mic, and clean audio comes out, no feedback, no fuss,” said Waker. “We also installed Ashly wall remotes so that banqueting staff can adjust levels without needing to call a tech. Genius!”

Ashly Audio

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Posted by Keith Clark on 11/18 at 12:00 PM
Live SoundNewsAmplifierAudioDigitalInstallationProcessorSound ReinforcementSystem • (0) CommentsPermalink

Live Concerts In Surround? Despite Some Obstacles, It Can Indeed Be Done

A well-balanced left-center-right sound system is a beautiful thing, and there are many examples of them producing great stereo sound for an entire audience. But we want more than that; we want surround sound. How can this be achieved?

About 15 or so years ago, I went to a rock concert at a stadium in Chicago. Having heard this band’s best-recorded music on some excellent small systems, including high-end four-channel setups, I looked forward to this concert with great anticipation.

They were known for having elaborate and advanced concert setups. I expected many stacks of loudspeakers and amazing “big, immersive” sound along with a huge video display.

Unfortunately, the outcome proved just the opposite - talk about disappointing! It was barely loud enough, muddy, and distant-sounding.

I couldn’t even tell whether the giant loudspeaker stacks on each side of the stage were supposed to create stereo sound.

They were just too far away, and their coverage of the audience was inadequate.

The show may have been better for those who were close to the stage, but the rest of us could get superior sound with our home stereos.

Sound reinforcement (in general terms) has improved a lot since then, and good sound for every audience member is now possible in most venues. Although concerts are commonly recorded and produced in 5.1 format for DVD, surround sound for live concert reinforcement is still almost nonexistent.

Because progress is inevitable, it seems likely some day concerts will be routinely presented in high-quality surround sound.

But even if the budget is large enough, the laws of physics still create problems for design of a large surround sound system. Fortunately, we can make useful compromises, and as loudspeaker designs improve, we can more readily achieve effective stereo sound for an audience.

This is a good basis for surround sound, which can utilize many of the same principles.

Start With Stereo
Achieving good stereo sound for a large audience requires careful design with close attention to acoustic issues. The basic requirement is for each listener to hear each of the loudspeakers at nearly the same level, nearly the same time, and with matched frequency response.

This is a tall order for any size venue, and requires certain compromises.

The biggest problem is the timing of the sound heard from different loudspeakers. The arrival time of the sound from each loudspeaker will be different for every member of the audience.

Due to the precedence (or Haas) effect, the sound source heard first usually determines the perceived direction the sound is coming from.

When panning between adjacent loudspeakers, the sound from each loudspeaker (or cluster) needs to arrive nearly simultaneously.

To produce reasonably accurate panning, the sound arrivals need to be within about a millisecond (ms) at each listener.

For an audience of more than a few people, this won’t happen because 9 ms delay is created by every 10 feet of distance from the sound source.

The audible effect of this time error between left and right loudspeakers at a listener depends on both the time difference and the relative level difference.

When the levels are about equal, just 5 or 10 ms can cause significant image shifting toward the earlier sound source.

Longer time differences, up to about 40 ms or so, will yield significant precedence effect - the earlier sound source will set the direction.

Also there will be coloration and loss of clarity. Time differences over 50 ms can be heard as echoes, which of course is a major problem.

Dealing With Delay
Keeping the loudspeakers high enough above the audience will reduce the relative delay, due to the geometry of the layout. This is certainly helpful, but except for the smallest venues, it alone does not solve the problem.

One way to cope with delay is to control the relative amplitude of the loudspeakers in specific audience areas. If the delayed sound has a higher level, this can partially compensate for the delay and help to restore the spatial balance.

In the simplest implementation, left and right loudspeakers are toed in and across. This insures that the opposite side of the audience area is on axis and thus covered with higher output from the loudspeaker than those on the same side and nearer, who are then off axis.

It may offer an improvement, but typically suffers from degraded off-axis frequency response, as well as more sound energy being dispersed away from the audience where it’s not needed.

A better approach is to use loudspeakers with a directivity function tailored to the specific audience area to be covered.

The loudspeaker further from the listener should operate at a higher level to compensate for its longer delay. This requires a specific loudspeaker directivity function in which the output level is dependent on the angle off axis, while maintaining good frequency response.

A cluster of two or more loudspeakers can be designed to approximate this function. Figure 1 illustrates the principle. Note that the higher-level cluster components (shown in yellow) have higher directivity and are aimed toward the farther part of the audience.

Figure 1: Covering the farther part of the audience with a higher level (loudspeakers shown in yellow) helps to restore the balance for off-center listeners (click to enlarge)

The right-side loudspeakers mirror the left. The center loudspeaker helps greatly to expand the effective stereo listening area, and it needs to provide even coverage for the entire audience.

Even with the most optimum directivity control, there is a limitation in how far apart the adjacent-channel loudspeakers can be. If they are more than about 40 feet (12 m) apart, then the relative delay causes serious problems for off-center listeners, including loss of intended localization and a noticeable echo.

The use of a left-center-right system increases the possible width of the front sound stage, as long as channel assignment or panning is kept pair-wise to adjacent channels (left to center, or center to right), so that the same signal is not heard from both left and right.

Long Distance Surround
A well-balanced left-center-right sound system is a beautiful thing, and there are many examples of them producing great stereo sound for an entire audience.

But we want more than that; we want surround sound. How can this be achieved?

Sound propagation in air takes about 90 ms for every 100 feet (30.5 m) of distance, and this creates a very bad situation for large venues that may be hundreds of feet across the audience area.

Additional loudspeakers at the back or sides cause problems because even a medium-sized facility has enough distance from front to back to produce echoes in both directions, as shown in Figure 2.

Listeners in the front will hear the back loudspeakers too late, and listeners in the back will hear the back loudspeakers too soon, compared to the sound propagating across the audience from the front stage.

In theater and cinema surround sound systems, these delay problems are largely mitigated through the limited size of the venue, and judicious application of the program material to the surround system.

Figure 2: Listener at “Position A” hears the real loudspeaker 50 ms too late, while the listener at “Position B” hears the rear loudspeaker 50 ms too soon. (click to enlarge)

Rarely is a particular component of the mix routed to both front and rear simultaneously; this would cause a problem with clarity and intelligibility.

But high-quality surround music needs more precise timing to provide sound from all directions nearly simultaneously for all (or at least most) listeners. One might conclude that there is no way to implement this in a large venue because of the great distances involved.

However, compromises can be made–sometimes with great results. The careful use of electronic delay and a number of well-placed, well-behaved loudspeakers can work wonders.

Whether the goal is to provide additional envelopment, bring the audience “into” the performance, or simply to present sounds from directions other than the front, we need the ability to provide sound that the entire audience can hear specifically from the sides or rear - without delay problems.

Go here as I present some possible solutions to these issues.

Michael Miles has two decades of product and system design experience, and he holds an Electrical Engineering (EE) degree from the University of Michigan.

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Posted by Keith Clark on 11/18 at 10:55 AM
Live SoundFeatureOpinionStudy HallAudioConcertLoudspeakerProcessorSound ReinforcementSystem • (0) CommentsPermalink
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