Interconnect

Tuesday, June 11, 2013

Clear-Com Releasing New HelixNet Partyline Intercom System Linking Capability At InfoComm 2013

Networking function increases channel access to support complex workflows

Clear-Com is launching the enhanced HelixNet Partyline intercom system at InfoComm 2013 in Orlando (booth 921).

The new version will include a multi-system linking capability to deliver a cost-effective intercom solution for distributing many network digital partyline channels and program audio feeds to dozens of digital beltpacks.

Enabling this capability are two new modules: the HLI-ET2 Ethernet module, for main station-to-main station networking over fiber or a LAN (Local Area Network), and the HLI-FBS fiber module for daisy-chaining fiber interfaces to link main stations over long distances.

The expanded functionality of HelixNet Partyline permits a more sophisticated and cost-effective digital partyline intercom network for large stadiums or multi-campus venues.

In these applications, a high volume of intercom users in disparate locations are often required to stay connected over long distances. HelixNet Partyline’s new system linking capability supports these complex workflows by granting access to many HBP-2X digital beltpacks through the connection of networked HMS-4X main stations over fiber or a LAN.

Due to its unique design, the entire HelixNet network will only need a single, shield twisted-pair cable as its power source—saving cost and resources for setup and configuration.  Moreover, the new version of HelixNet easily integrates with other wireless or two-wire systems to create a complete intercom setup.

“Linking main stations gives HelixNet users the capability to share digital partyline channels plus program inputs in a network distributed system,” says Simon Browne, Clear-Com director of product management. “This means that any of the possible dozens of HelixNet Beltpack users can access any two of the available network partyline channels by simply selecting the desired channels on their beltpack.

“Since all these channels are running off a single, shielded twisted-pair cable, users no longer need to physically relocate multiple cables for new configurations, utilize expensive source-assign equipment or rely on multiple power supply units.”

One of the modules supporting the system linking function is the new HelixNet Partyline HLI-ET2 Ethernet module for main station-to-main station networking. Built to fit directly into one of the HMS-4X option module bays at the back of the main station, the HLI-ET2 Ethernet module can connect main stations directly through a LAN using standard IT switches.

A new HLI-FBS fiber module is also available for linking main stations over long distances. The fiber module has two fiber ports using SFP modules for simple exchange of fiber transceivers. It also allows linking of other main stations in a fiber daisy-chain. The standard connection is single-mode, with multi-mode offered as an option.

Clear-Com

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Posted by Keith Clark on 06/11 at 07:41 AM
AVLive SoundChurch SoundNewsProductAVEthernetInterconnectNetworkingSignalSystemPermalink

Manhattan Center Studios Extends Its Riedel Intercom System To Seamlessly Connect All Venues

One of the largest Riedel intercom integrations in the America

Manhattan Center Studios is extending its Riedel Communications intercom system to tie together all of its studios and performance halls, as well as a new client news network, Al Jazeera America, that will be completed this summer. Manhattan Center Studios is one of the largest Riedel intercom integrations in the Americas.

“We believe that the Riedel intercom is on another level from its competition, and with every software release, it becomes both more powerful and easier to program,” says Marvin Williams, director of engineering and operations at Manhattan Center Studios. “There is no request a production or client can make that we cannot accommodate.

“Now that we are completely Riedel — Artist matrix and keypanels, Performer digital party lines, and Acrobat wireless — the whole system is a pleasure to use. Investing in Riedel is one of the best decisions we ever made.”

The Manhattan Center is known for hosting performances and special events in which media and entertainment come together. With its unique location in the heart of midtown Manhattan, the facility is a premier mid-size venue for corporate galas, charity fundraisers, award shows, rock concerts, and fashion shows.

The Manhattan Center is home to the Hammerstein Ballroom, originally created as an opera house, as well as a full ballroom, two full TV studios, the Log Cabin and Studio 7 audio studios, and 20 plus edit suites, all of which are connected with Riedel intercom systems.

The new installation includes 64-port and 32-port Artist digital matrix intercom mainframes, both equipped with redundant power supplies and CPU cards and featuring fiber connectivity. MADI Client cards allow the transparent transport of up to 64 MADI signals through the Artist infrastructure. For accessing the intercom matrix, a combination of new OLED rack-mount (RCP-1112) and desktop (DCP-1116) key panels have been distributed across the facility.

In addition, their Acrobat wireless system was expanded with a new CC-60 system controller, ten new beltpacks, and additional antennas to enhance coverage. The Manhattan Center Studios also added two commentary control panels (CCP-1116) that can each support two commentators, for use during live events or voiceovers.

“Manhattan Center Studios was one of our first U.S. customers, and today it continues to be one of our greatest partners,” says Patti Gunnell, entertainment solutions manager, North America, Riedel Communications. “Customers like Manhattan Center Studios, who demand products that are both flexible and reliable, have made our Artist product the best digital matrix intercom platform available today.”

Riedel Communications

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Posted by Keith Clark on 06/11 at 06:29 AM
AVLive SoundRecordingNewsAVEthernetInstallationInterconnectNetworkingRemoteSoftwareStageStudioPermalink

Aviom AN-16/i v.2 Analog Input Module Now Shipping

Supports increased channel availability for new A360 personal mixer

Aviom has announced that the new AN-16/i v.2 input module, which supports the new A360 personal mixer, is now shipping. The AN-16/i v.2 converts 16 line-level analog audio channels to a Pro16e A-Net digital stream.

Based on an enhanced version of Aviom’s Pro16 A-Net protocol, Pro16e, the technology driving the new A360 personal mixer allows multiple 16-channel banks of monitoring content to be delivered over a single Cat-5 cable.

The AN-16/i v.2 includes an A-Net Input jack for connecting up to three additional input modules. The analog inputs of the AN-16/i v.2 are merged with this incoming A-Net stream to create the expanded Pro16e digital audio stream utilized by the A360.

The new A360 brings new levels of control and customization to the personal monitoring experience with a number of new features. It offers the ability to choose which channels are mixed on each unit in the system. Each A360 may use the default channels, or audio channels can be selected and organized individually for each personal mixer, drawing from a network pool of up to 64 channels.

According to Chandler Collison, Aviom’s vice president of sales and marketing, “One of the most frequent requests from users is more channels in the personal mixing system. The new version of the input module gets more channels into the system for the A360 and is a simple and cost-effective way for existing Aviom users to expand and upgrade their systems.”

The AN-16/i v.2 features eight four-position input level/gain sensitivity switches, stereo channel link switches for every channel pair, and per-channel signal present and clip LEDs on the front panel.

On the rear panel, the unit has 16 analog audio inputs with balanced 1/4-in TRS jacks. The audio Thru jack for each input allows the AN-16/i v.2 to be inserted seamlessly into an existing audio signal path. Also on the rear panel are one A-Net In jack and one A-Net Out jack, along with a slide switch for assigning the channel bank.

The AN-16/i v.2 can be used with any line-level analog audio signal such as console direct outs, inserts, or aux sends.

The new AN-16/i v.2 Input Module works with the new A360 as well as the company’s current A-16II personal mixers, and the two personal mixers can be combined seamlessly into the same system.

In addition, the AN-16/i v.2 can be used as part of a Pro16 digital snake or audio distribution system. These systems support a range of flexible configurations, including 16 x 0, 32 x 0, 48 x 0, 64 x 0, 16 x 16, 32 x 16, 32 x 32, and 48 x 16, with an unlimited number of digital splits and no loss in audio quality.

Aviom

 

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Posted by Keith Clark on 06/11 at 05:21 AM
AVLive SoundChurch SoundNewsProductAVEthernetInterconnectMonitoringNetworkingStagePermalink

Monday, June 10, 2013

Universal Audio Now Shipping Flagship Apollo 16 Interface & New Apollo Software

New software delivers multi-unit cascading, virtual I/O, and more

Universal Audio (UA) is now shipping the new Apollo 16 audio interface and new UAD software v.7.0, which provides multi-unit cascading, Virtual I/O, and other enhancements to both the Apollo 16 and Apollo audio interfaces.

Apollo 16 is Universal Audio’s flagship 24-bit/192 kHz audio interface, delivering world-class conversion with 16 x 16 analog I/O.

The FireWire/Thunderbolt-ready interface combines superior sound and flexible routing with onboard UAD-2 QUAD processing, allowing the tracking of audio in real time through the full range of classic UAD analog emulation plug-ins — from Neve, Studer, Manley, Lexicon and more — on both Mac and Windows 7. (*Requires Windows 7 64-bit edition operating system and a qualified PCIe-to-FireWire adaptor. Thunderbolt option is Mac-only.)

Apollo 16 offers meticulous analog circuit design, top-end converters, and DC-coupled outputs. Two Apollo 16 units can be cascaded over MADI for an expanded system with eight UAD processors and 32 x 32 simultaneous analog I/O, capable of handling large professional mixes.

Apollo 16 also offers compatibility with Intel’s high-bandwidth Thunderbolt technology on Macs via a user-installable dual-port Thunderbolt Option Card (sold separately). Thunderbolt provides greater UAD plug-in instances, improved performance at high sample rates, and reduced UAD plug-in latency in the DAW versus Apollo’s standard FireWire connection.

Now shipping, Apollo 16 carries an estimated street price of $2,999 US. Apollo’s Thunderbolt Option Card is also now available for an estimated street price of $499 US.

UAD Software v7.0 provides significant enhancements to Apollo’s workflow and expandability and adds the new Ocean Way Studios, SPL TwinTube, and Sonnox Inflator plug-ins to the UAD Powered Plug-Ins platform.

Among these enhancements is multi-unit cascading, which allows for combining two Apollo 16 interfaces (for 32 x 32 analog I/O) or two Apollo interfaces (for 16 x 16 analog I/O, with eight Apollo mic preamps) into a single elegant system via FireWire or Thunderbolt.

The boost in Apollo connectivity is navigated via a redesigned Console application, offering better visual feedback, a new PT Mode which simplifies outboard hardware integration with Pro Tools, and a new Virtual I/O feature that allows for Realtime UAD Processing of DAW tracks and virtual instruments.

Download UAD Software v7.0 here.

Universal Audio

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Posted by Keith Clark on 06/10 at 01:16 PM
Live SoundRecordingNewsProductDigitalDigital Audio WorkstationsInterconnectProcessorSoftwareStudioPermalink

Friday, June 07, 2013

FiberPlex Introduces Live Production Toolbox For Convenient Fiber Optic Connectivity

A system with everything needed for transporting audio, video, control and data feeds of various formats

Debuting at InfoComm 2013, FiberPlex Technologies introduces the Live Production Toolbox, which includes everything needed to easily connect remote cameras, microphones, and other equipment over fiber optic cable regardless of media type or data format. 

“Fiber is really the only contender when it comes to transporting information like high-bandwidth 3G or 6G-UHD video, audio, lighting control, etc., because it is high capacity and noise-free from mile one to mile 99,” says Kyle Rosenbloom of FiberPlex. “The difficulty until now has been setting it up for different media formats. With this toolbox, we’ve literally made it easy to not only set up varying types of feeds, but also to change them out quickly as needed.”

Central to the toolbox is FiberPlex’ WDM16 active wave division multiplexer combining 16 optical channels onto a single pair of single-mode fiber, each channel supporting aggregate data rates from 155MB to 3GB.

The multiplexer automatically converts signals to the appropriate optical wavelength for transport purposes, taking the tedium out of fiber connectivity and opening up the use of optical links for on-demand, fast-paced applications such as live production.

“Inevitably, unforeseen challenges crop up at the last minute during a live production or broadcast, and, many times, techs are left scratching their heads and running around like crazy looking for a solution. With the WDM16, all they have to do now is plug in another SFP module and they’re good to go. We are making fiber connectivity a plug-and-play proposition,” says Rosenbloom. The WDM16 allocates any audio, video, or other data feed to a specific wavelength on a full spectrum (like a rainbow of colors that cannot be seen by the human eye), and then combines all of those wavelengths together onto a single pair of fiber.

Hot-pluggable SFP (small form-factor pluggable) modules for the WDM16 come in a variety of formats for interoperability between SD-, HD- and 3G-SDI as well as 10/100/1G Ethernet connectivity, MADI, varying wavelengths of multimode/single-mode optics, and single fiber BiDi (bidirectional). FiberPlex WDM16 frames also accept standard SFP modules from third-party providers.

Another significant device in the Live Production Toolbox is the new FOI-6010, a universal hot-pluggable SFP/SFP+ frame for interoperability between media formats up to 10GB.

The FOI-6010 can be used as a stand-alone unit for smaller applications needing just one or two fiber links, or for larger productions that require routing of multiple signals back to the WDM16 from, say, several camera and announcement locations around a sports stadium. Whether transporting standard definition or high definition (3G-SDI or newer 6G-UHD), Ethernet or various serial data formats, FiberPlex has it covered with this single powerful solution.

Rounding out the Production Toolbox is the FiberPlex LightViper digital audio management system to handle distributing audio throughout a venue. The LightViper system includes stagebox/mic preamps, analog and digital AES-3 audio, a high-quality word clock, and either a fiber optic pass through or fiber optic splits on every device.

The Production Toolbox will be at FiberPlex booth (820) at InfoComm 2013.

FiberPlex

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Posted by Keith Clark on 06/07 at 01:06 PM
AVLive SoundNewsProductAVDigitalInterconnectNetworkingSignalPermalink

Church Sound: Technical Issues Aren’t Always Technical

Thinking outside of the "techie" box
This article is provided by Gowing Associates.

 
A couple of issues that arose in a couple of the churches I support bred the title for this post.

I realized after these events that it’s sometimes hard for techie types to think outside of the technical box. They sometimes get so focused that they can’t see the forest for the trees. Some of this is due to not having enough technical knowledge and experience. Some of it is due to the exact opposite. I know that doesn’t make much sense but bear with me.

Event #1
I received a phone call from one of the techs at a church. They’ve been there for a couple of hours checking to ensure that everything is working on the new computer that was installed the week before. Since they use split tracks, they were trying to figure out where the balance control was in Media Player 10 to select between the music and the vocal track.

Well, it turns out that while it used to be in the graphic equalizer plug-in in Media Player 9, Microsoft appears to have taken it out of 10. So I had them load up Winamp which took care of that issue since it does have a balance control. Problem number one solved.

Then they were attempting to play a video, but were only getting the video and not the sound. I had them download and install the K-Lite Mega Codec pack which solved Problem 2. So far so good. I hang up the phone and go about my day.

A couple of hours later I received another call from the tech saying that the monitor screen that is “Y-d” with the rear confidence projector feed is showing blurry and doubled. Since I had checked all three monitor outs when I set up the desktop the week before I knew that all of them were working correctly on the front and rear projectors and the control monitor when I left.

The tech had tried everything he knew. I’m thinking about all the possible permutations of what could cause it, from a bad graphics card, bad Y adapter, bad cable, bad driver, etc. Keep in mind the tech has been trying to get this resolved for the last hour or so. All I’m thinking about is that I hate computers at this point!

I asked him if the projector also is showing the bad graphics since I’m figuring that if the projector and the monitor are both showing the same bad stuff, chances are good that it’s either the Y adapter or the graphics card. He hadn’t tried the projector so I had him go upstairs to turn it on. No projector. It turns out the pastor had borrowed it and didn’t tell anyone.

So I asked him track down the projector and plug it in. Viola! Everything is hunky-dory!

Moral of this story: Don’t take for granted that everything is where its supposed to be. He could have saved himself a lot of time had he turned on all of the equipment because he would have noticed a missing projector.

Event #2
This one happened at my church. I was running sound during Thursday night rehearsal and everything sounded good, even having some fun times seeing just how loud our system can run (105 dB, and yes, the dust was coming off the HVAC ducts from the amount of bass that was pumping through the system.

Everyone goes home happy. Saturday night service rehearsal starts, and after the first song the worship leader asks “Did anyone hear that scratchy, ticking sound?” To which everyone in the band replied affirmatively.

We started diagnosing the problem by muting all the channels and going one by one through each channel attempting to determine what was causing it. We got to the acoustic guitar channel and the musician started playing. Well… waddaya know? There it is! It’s clicking and scratching like crazy. When he stops playing the noises went away.

The worship leader informed me that the same thing happened the week before, and they replaced the instrument cable, which seemed to get rid of it. Hmmm… So I started a decision tree in my head about what the cause might be and ruled most everything out.

I asked the acoustic guitar player to start playing, and noticed that he was wearing a long-sleeve dress shirt. I stopoed him and asked if his shirt had buttons on the cuff and opening. It does.

So I asked him to roll up his sleeves. No more scratching, no more ticking. Everyone was like, “seriously?” Yup. Problem solved

Moral of the story: Take a step back and observe before you dive into technical problem solving. The time and angst you save just might be your own.

The overall point of this post, or as my buddy Chris Huff at Behind The Mixer says, “The Take Away,”  is to avoid getting so focused on finding a technical solution to a problem that you overlook the obvious.

Sometimes looking at the environment will help you place the problem in perspective. While a significant number of technical problems can only be solved by technical solutions, don’t ignore the non-technical possibilities.

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

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Posted by Keith Clark on 06/07 at 10:39 AM
Church SoundFeatureBlogStudy HallEngineerInterconnectSignalSystemTechnicianPermalink

Thursday, June 06, 2013

Max Gilkes Upgrades Mastering Facility With SADiE 6 And Prism Sound

Using new tools on a variety of projects, including mastering an album for French artist Mayra Andrade

Producer Max Gilkes has upgraded his Brighton, England-based mastering and post production facility 1 Sonic with SADiE 6 software for mastering and a Prism Sound Orpheus FireWire audio interface.

Gilkes, who founded 1 Sonic eight years ago, says the support offered by Prism Sound was a key factor in his decision to invest in the company’s products.

“I was previously using Peak as a playlist editor and doing most other things on Pro Tools,” he explains. “But as it was time for an upgrade I decided to go with a company that I knew I could have a good working relationship with, and one that was also based in the UK.”

With a client list that includes Ninja Tune, Big Dada, Sony (France), Smart Move Productions, Mr. Bongo, Keep Up, Silverland and Mission, Gilkes is rarely out of the studio. His recent credits have included Fink, Dobie, Roots Manuva, The Skints, Deco Child, Raffertie, Eliza Carthy and Prince Fatty.

The facility itself is mainly used for mastering and mixing, but it also has a live room with eight tie-lines that is large enough to record drums.

“The outboard in the studio is primarily for tracking, so there are some nice pres and compressors,” Gilkes says. “I also have a master section by Audient, PMC and Yamaha monitors, Pro Tools, various classic guitar and bass amps and a range of microphones – not to mention an over-used tea pot!”

Moving from one software platform to another can be intimidating, but Gilkes says he has found the transition to SADiE 6 far less painful than he anticipated.

“It’s always hard work making the transition from one system that you are particularly well versed with to a whole new environment, but SADiE 6 is very intuitive and easy to use,” he says. “I’m really enjoying the facilities it provides. The editing facilities, in particular, are great and I like the PQ and DDP tools. However, the sound quality is the most outstanding feature and I am really impressed by that.”

Sound quality also played a part in Gilkes’ decision to buy a Prism Sound Orpheus FireWire audio interface.

“It was a no brainer, really,” he says. “The studio needed to make a significant step up in convertor quality and Orpheus delivered the transparency and clarity I was looking for, both as an A/D and a D/A converter. It has great sounding pre’s and I like the interface features for multi-tracking in Pro Tools.”

Gilkes is now using SADiE 6 and Orpheus on a variety of projects, including mastering an album for French artist Mayra Andrade on Sony (France), recording tracks for Congo Natty, completing a debut album for Chevron, recording drums for the next Grasscut (Ninja Tune) album, finishing backline tracking for a band called Bentcousin, which is releasing a three-track single through Team-Love NY and completing an album for a local brighton band called the Meow Meows.

And if that work schedule isn’t intensive enough, Gilkes is also producing and co-writing with the legendary Russell Stone of R&J Stone fame.

“The Russell Stone project is very interesting as all tracks are vocal/non-verbal and based on free improvisations,” he says. “The album, which is called HUMU, is very nearly finished and should be out in September.”

SADiE
Prism Sound

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Posted by Keith Clark on 06/06 at 10:34 AM
Live SoundRecordingNewsDigital Audio WorkstationsInterconnectProcessorSoftwareStudioPermalink

Tuesday, June 04, 2013

Alford Media Services Expands Portfolio Of Riedel Equipment

Adds range of Performer digital partyline intercom systems, Artist digital matrix intercom systems, and more

Alford Media Services, a supplier of event technology support to corporate clients across the United States, has upgraded and expanded its portfolio of Riedel Communications equipment.

With this investment, Alford has increased both its capacity for providing complete top-quality event communications systems and its ability to offer comprehensive training for new and existing users of Riedel gear.

“Riedel Communications offers flexible, state-of-the-art intercom solutions that address our customers’ requirements for performance, audio quality, and ease of deployment,” says Rich Tate, director of creative support at Alford Media Services. “We strive constantly to maintain versatile and highly efficient solutions that can be configured quickly and easily to support high-end corporate events, and our new Riedel gear meets this test very well.”

Alford has continued to systematically upgrade its inventory with a comprehensive range of Performer digital partyline intercom systems, Artist digital matrix intercom systems, and related control panels and stations, split boxes, interfaces, beltpacks, and headsets. Making these systems available from its Coppell, TX facilities, the company offers another U.S. source for Riedel gear rental and training.

“Alford is an extraordinary rental company that not only has invested in and supported our products, but has also taken on the task of making product training a priority for its staff and clients,” states Patti Gunnell, Riedel entertainment solutions manager, North America.

“We are seeing rising demand for our products in markets around the world, and the U.S. is no exception,” notes Christopher Street, general manager, North America at Riedel Communications. “Alford’s expansion and upgrade of its Riedel systems will make it easier for U.S. companies in the corporate space to take advantage of our renowned communications technologies, and its ramped-up training capabilities will ensure the availability of freelance operators who are familiar with our intercom product lines.”

Riedel Communications

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Posted by Keith Clark on 06/04 at 05:04 PM
AVLive SoundNewsAVBusinessEthernetInterconnectManufacturerNetworkingPermalink

Crest Audio Announces New Tactus Digital Mixing System

Built in partnership with Waves, new platform delivers quality audio, custom configuration and pro features

Crest Audio has partnered with Waves to create the Tactus digital mixing system, providing a simple, customizable touch-screen operating interface that puts all of the audio routing, processing and mixing tools within immediate reach. It’s designed to work with the new Waves eMotion mixer for SoundGrid.

The Tactus hardware is comprised of the Tactus.FOH, the audio processing core for the system, and the Tactus.Stage, a 32-in/16-out remote stage box. The modular capability of the Tactus Digital Mixing System, which interconnects via standard Gigabit Ethernet cabling and an internal Gigabit switch, allows multiple configurations in a range of channel counts to best suit each application.

The Tactus.FOH is powered by the Waves SoundGrid audio processing/networking platform, providing extremely low latency with precision audio processing. The Tactus.FOH frame provides 8 local microphone/line inputs and 8 line outputs, and serves as the audio signal processing engine.

The Tactus.Stage includes 32 microphone/line inputs and 16 line outputs. The digitally controlled high-quality mic preamps offering low input noise, high slew rate, low THD, and 66 dB adjustment range in 1 dB steps to maximize dynamic range. An integrated gigabit Ethernet switch allows for easy networking. Users also benefit from the option of combining Tactus.Stage I/O interfaces for a total of 64 stereo inputs by 32 stereo outputs.

Tactus.FOH highlights:
• Primary server for the Tactus mixing system, hosting and supporting the eMotion LV1 software mixing platform.
• Will function as a standalone core for an 8 in, 8 out mixing platform.
• 8 balanced mic/line inputs, XLR, TRS and 1/4-inch Phono
• 8 balanced outputs - First 4 are male XLR connectors, 2 of which have parallel TRS jacks, plus 4 additional TRS jacks.
• Mic input accepts a +27 dBu input signal without pad
• 66 dB adjustment range in 1 dB steps to maximize dynamic range
• Gain changes on zero crossing to minimize adjustment noise.
• 48-Volt phantom power switchable on every mic input.
• Integral gigabit Ethernet switch for easy networking
• Waves SoundGrid Digital audio network protocol
• Balanced output maximum level switchable between + 24 dBu and + 18 dBu
• Supports sample rates of 44.1 kHz, 48 kHz, 88.2 kHz and 96 kHz
• External word clock input and output
• Two AES digital outputs
• Headphone output with level control
• MIDI in and switchable out or thru
• 3U rack mount package
• Universal input power supply

Tactus.FOH


Tactus.Stage highlights:
• 32 microphone/line inputs
• XLR mic and TRS 1⁄4-inch line inputs
• 16 Line outputs
• Mic input accepts a +27 dBu input signal without pad
• 66 dB adjustment range in 1 dB steps to maximize dynamic range
• Gain changes on zero crossing to minimize adjustment noise.
• 48 Volt phantom power switchable on every mic input.
• Integral gigabit Ethernet switch for easy networking
• Waves SoundGrid Digital audio network protocol
• Balanced output maximum level switchable between + 24 dBu and + 18 dBu
• Supports sample rates of 44.1 kHz, 48 kHz, 88.2 kHz and 96 kHz
• External word clock input and output
• Two AES digital outputs
• Headphone output with level control
• MIDI in and switchable out or thru
• 4U rack mount package
• Universal input power supply

Tactus.Stage


eMotion LV1 mixer highlights:
• 64 input channels (mono/stereo)
• 32 stereo buses + return channels
• L/R, center, mono main buses + master channels
• 8 DCA faders, 8 mute groups, 8 user-assignable function controls
• Cue/SIP and talk-back
• EMO-Q4 Equalizer: 4-band EQ with HP and LP per channel
• EMO-D5 Dynamics Processor: Comp/Gate/Expander/DeEsser/
• Limiter/Leveler per channel
• Up to 6 plugins per channel
• Connect to multiple DAWs for recording and playback
• Save and transmit sessions and snapshots from any DAW to eMotion
• Connect and share SoundGrid I/O devices, drivers and applications
• Up to 1,000 Scenes/Snapshots
• Up to 96 kHz sample rate
• Automatic plugin delay compensation

Tactus touch-screen interface


Waves
Crest Audio

 

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Posted by Keith Clark on 06/04 at 12:51 PM
AVLive SoundChurch SoundNewsProductAVConsolesEthernetInterconnectMixerNetworkingProcessorSoftwareSound ReinforcementPermalink

Monday, June 03, 2013

The Old Soundman: A Rare Glimpse Into The Tool Bag Of A Legend

A rather seedy individual pushes OSM's buttons, giving him the chance to rant -- his favorite thing!

An individual who corresponds to the description of a fugitive known to viewers of “America’s Most Wanted” and “Cops” known only as “JR” sent in this question, which gives me a chance to pontificate, something that I very much enjoy doing.

The beautiful thing is that I can go on about stuff that my questioners are correct about, or just as well when they are off the mark. It’s all fuel for my turbine, my reactor, my photovoltaic converters.

What’s in the essential gig bag for OSM? Does he carry a duffel with anything besides a drum key, bottle of bourbon and crescent wrench?

Let’s take things in order.

I just bought two new drum keys due to guys like you making off with the last one I had. It always happens, either the drummer doesn’t have one (what’s up with that?) or his tech doesn’t.

So I lend them mine, then I go out to have a smoke, and forget to persecute the wicked to get my drum key back. I mean, they only cost a few bucks! What’s wrong with humanity?

Just a few weeks ago, I was mixing a show at someone else’s club, and the house guy told the drummer that his rack tom was resonating. But neither one of them had a frickin’ key! So I saved the day. I did it for everybody’s sake, for the sake of the show. Capisce?

Number two: It has been many years since I felt a heightened sense of excitement by hoping to be like Keith Richards clutching his stupid bottle of Jack, or Slash with the eternal Marlboro.

Sure, I smoke and drink in moderation, but it is my personal feeling that Jack Daniel’s and Jim Beam are more consistently violence-producing than heroin or crack cocaine. Oh, I’m sorry, were those companies about to buy a banner ad? Whoops…

The crescent wrench—now you’re preaching to the choir! I got my crescent, my needle nose, my clippers, my big electronics store tweezers, my soldering gun and solder, my beer bottle openers, Sharpies of different colors, fine point sharpies, my pliers, my personal nobody-else’s-loogies talkback mic.

My Sony headphones, with their ridiculous bag with the seams that split down the sides; why haven’t they rectified this awful design flaw in the last decade? Everybody I know hates these things!

I also have some generic wire to repair XLRs with. I have some black trick line, and some cable ties. I have some orange mountaineering line. I have a polarity tester and an XLR tester, a bag of XLR Y-cables, pin one lifts, and sex changers.

I have RCA to quarter adapters. One of your buddies stole my vise grips. Foam earplugs for sensitive friends and suffering strangers.Tweakers. Greenies. A reversible big-ass screwdriver. Flashlights.

Yes, it’s heavy, Soundman! You couldn’t handle this bag!

Extra batteries. Guitar tuner. Lanyards. One-inch gaff, two-inch gaff. AC ground lifts. Electrical tape. In-ear monitors. Mini stereo to quarter-inch stereo jacks. Orange three-banger ac combiners. A razor knife. Multimeter. Hex key sets, inches and metric. Ten-foot tape measure.

What’s the Ultimate Sound Man (USM) need to get the job done?

Hey, bro? One more time—I handle the jokes around here. I make funny. You laugh. That is the sequence, not the other way around.

I do not resemble an Ultimate Support keyboard stand. I would not mind being compared to the Ultimate Support percussion table, those things have many uses. I am not an Ultimate Support mic stand with the squeezy thing. I am also not a low profile collapsible Ultimate Support guitar stand, but those were a cool innovation, wouldn’t you agree?

Enough. Begone!

Luv,
The Old Soundman

There’s simply no denying the love from The Old Soundman. Check out more from OSM here.

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Posted by Keith Clark on 06/03 at 06:46 AM
Live SoundFeatureBlogConcertEngineerInterconnectMonitoringSignalStageSystemTechnicianPermalink

Friday, May 31, 2013

Systems Engineer Philip Reynolds Chooses Focusrite RedNet For The Killers

Reynolds made the decision to use RedNet in order to maintain the audio within the digital domain

Focusrite RedNet audio I/O technology is being implemented for the front-of-house rig for The Killers world tour, where specifically, a RedNet 1 eight-channel A-D/D-A and RedNet 4 eight-channel mic preamp perform a number of duties for systems engineer Philip Reynolds.

Reynolds made the decision to use RedNet in order to maintain the audio within the digital domain — A-D conversion takes place side-stage as soon as the mics plug into the stagebox — for as long as possible before conversion to analog.

In addition, he uses audio analysis software coupled with a reference mic to match the sound system response as closely as possible to the output of the front-of-house console. RedNet handles all of his inputs and outputs to facilitate that process.

RedNet 1 routs audio to and from his test system, and also to the FOH desk for house music playout and the drones that are used during the show’s encore. This material comes from a pair of Mac Minis, one primary and one backup, which all connect to the RedNet network using an Dante Virtual Soundcard (DVS) driver. RedNet also handles the press and video feeds, as well as any venue needs, such as the hearing assist system for the hard of hearing.

Previously, Reynolds used the on-board preamps in a conventional Firewire audio interface before switching to the RedNet 4. “I was blown away. The RedNet 4’s preamp is flat, and the phase coherence is perfect,” he says. This accuracy is important, with his ability to make fine tweaks to the loudspeakers relying on the analysis software’s precision.

Because Reynolds uses his own gear on tour — rather than rental equipment — the RedNet system represents a significant personal investment.

“Because it’s mine, I know it works and that I’m going to have it where I go,” he states. “RedNet does everything I want it to do, and the expandability is endless with this setup. I rely on my system 100 percent, and it just works.”

Over time, he intends to expand his RedNet system with an additional RedNet 1 and a RedNet 3 digital I/O, the latter of which will enable him to drive the front-of-house PA from the RedNet network.

“RedNet units have allowed me to design a system to bring a fully digital system on the front end,” he concludes. “With the RedNet 3 providing primary digital connection and RedNet 1 acting as my analog backup, we can bring the latency down and keep the quality of audio the same, which has been a goal of mine for the last year.”

Focusrite

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Posted by Keith Clark on 05/31 at 01:57 PM
AVLive SoundRecordingNewsAVDigitalEngineerInterconnectNetworkingProcessorSoftwareSound ReinforcementTechnicianPermalink

Dirty AC Power? Where To Lay The Blame For System Noise Problems

Replacing myth and misinformation with knowledge and clear understanding

The idea that “dirty” power causes audio system noise problems has a nearly irresistible intuitive appeal - and there are dozens of companies ready to cash in on this widespread but mistaken belief.

For example, here is a quote from a well-known manufacturer of power conditioning products: “Today’s residential systems contractors face unprecedented challenges where high resolution, trouble-free operation is required.

From inducing AC ground loops, video hum bars, static bursts, damage from AC line surges and variable audio and video performance, comprehensive control and conditioning of AC power is no longer an option.“

In fact, the power line doesn’t cause ground loops at all—and no amount of power “cleansing” or “purification” will prevent them!

Obviously, if every highway were smooth as glass, our cars wouldn’t need suspension systems. But it’s simply unrealistic to expect such highways - we pretty much have to accept them as they are.

The same is true for the AC power line. It’s a utility used by all sorts of appliances and equipment - and it’s certainly not pristine. Further, the power distribution systems in our buildings unavoidably create small voltages and currents that can potentially contaminate our signals.

Therefore, we need “suspension systems” to isolate our audio signal paths from the power line. Any pathway that allows coupling between the two is the fundamental problem causing noise in sound (and video and computer) systems.

Even though this is demonstrably true, and based on real science, it’s often difficult to persuade folks that “bad” AC power isn’t to blame. Audio systems routinely suffer from hum and buzz even when AC power is pristine.

Figure 1: Cable couples, audio-frequency noise only. (click to enlarge)

In unbalanced interconnections, the noise is usually coupled in the audio cables. It isn’t that the cables are poorly shielded; rather, it’s due to the basic properties of wires.

A simplified equivalent circuit of an unbalanced audio cable (Figure 1) shows that the shield, like any wire, has both DC resistance and inductance and that this inductance is magnetically coupled to the inductance of the center conductor, creating a kind of transformer.

The impedance of an inductance increases in direct proportion to frequency. Therefore, when current flows in the shield at frequencies below about 10 kHz, most of the voltage drop occurs across the resistance and very little across the inductance. It is this voltage drop that adds noise to the audio signal and is responsible for 99 percent of noise problems in unbalanced interfaces.

But, at higher frequencies, most of the voltage drop occurs across the inductance and, through transformer action, induces an equal voltage in the center conductor, thus reducing the coupling as frequency increases.

We may conclude that noise coupling in unbalanced interfaces is a significant problem only at audio frequencies. Balanced interfaces are generally immune to this coupling mechanism, but can fall victim to others.

Normal & Common
First, let’s briefly define some terms used to describe power-line noise. “Noise” in this context is generally defined as any voltage at a frequency other than 60 Hz (in the U.S.). Harmonics of 60 Hz, generally 3rd, 5th, 7th, etc. are usually the largest components of noise (heard as “buzz” when they enter the audio path). Differential or “normal-mode” noise is that between line and neutral, while “common-mode” noise is that between neutral and safety-ground.

Generally, normal-mode is larger because it’s created not only by local loads on the branch circuit but can also be brought in through the main service panel from outside power lines. The combined normal-mode noise on each “phase” is conducted to all its branch circuits via the main service panel.

Common-mode noise can be created only by equipment on each branch circuit because neutral and safety ground for each branch circuit are bonded at the main service panel (this assumes there are no “shared neutrals”). See Figure 2, which shows a simplified schematic of a power-line filter. Such filters typically reduce normal-mode and common-mode noise only above about 50 kHz.

This is not very useful at reducing audio noise - audio cables themselves have significant rejection starting at high audio frequencies. Note that the normal “leakage” currents that flow in capacitors “Cy” couple noise from both line and neutral to safety ground.

Figure 2: Typical AC power filter. (click to enlarge)

Remember, safety ground is normally the reference “ground” for each piece of equipment. When additional noise currents flow into the safety ground wiring, more noise voltage is created between outlets. It’s these voltage drops that can couple into our audio at some vulnerable point!

When any line filter, conditioner, or isolation transformer is used, electrical code requires that both it and the equipment it feeds remain connected to safety ground, as shown for the isolation transformer (Figure 3). Because capacitances in both transformers and filters divert additional 60 Hz and high-frequency noise currents into the safety ground system, they frequently aggravate the problem they claim to solve.

Second, the touted noise attenuation figures for virtually all these power line devices are unrealistic. Laboratory measurement setups connect the device and test equipment to a large metal ground plane.

Figure 3: AC power isolation transformer.(click to enlarge)

The resulting specifications can be impressive, but in my opinion they don’t represent performance in a real-world system where the ground connection is to safety ground wiring or conduit.

However, these devices can be very effective if installed at the power service entrance, where all system safety grounds are bonded to each other.

Seductive Idea
So-called “balanced power” – or more properly, “symmetrical AC power” - is another seductively appealing idea.

Explanations of the concept often mistakenly assume that the internal capacitances from power line to chassis that exist in all equipment (C1 and C2 or C3 and C4 in the figure as shown in Figure 4) are equal.

Of course, if this were true, the normal “leakage” currents in these capacitors would completely cancel because the voltages across them are equal but of opposing polarity.

But this assumption is not valid for typical real-world equipment, where one capacitance is often several times larger than the other. Therefore, real-world noise reduction is usually less than 10 dB and rarely exceeds 15 dB, a fact even promoters admit.

In audio systems, 10 dB improvements rarely “solve” noise problems. But 10 dB might be cost-effective if it makes a “hum bar” disappear from a video display. Increased cancellation of noise would require manufacturers to better match power line to chassis capacitances in their equipment, which is unlikely.

In systems where all equipment uses two-prong (ungrounded) power connections, filters and isolation transformers will have little effect on noise, but balanced power may offer some improvement. However, in systems where some or all equipment uses three-prong power connections, the effects of leakage currents pale in comparison to magnetic effects in premises wiring.

Figure 4: Balanced power attempts to cancel ground noise current. (click to enlarge)

By transformer action, current flow in line and neutral wires create a magnetic field that can induce significant voltages over the length of the safety ground wire. Although this is a major source of ground voltage differences between outlets (contributing to system noise), power conditioning has no effect on it whatsoever.

Explaining Benefits
When power conditioning is installed, usual practice is to power most, if not all, system equipment with its output. The fact that all system equipment is powered from very closely spaced outlets may drastically reduce ground voltage differences between pieces of equipment.

This may explain most of the benefit usually attributed to the conditioner. But this can be done with an ordinary outlet strip or, at the very least, by powering all system equipment from the same branch circuit.

Although ground loops often involve safety ground connections, it cannot be emphasized enough that disabling them with “lifters” or ripping out the third pin is both highly dangerous and illegal. I’ve seen pricey audiophile power “accessories” whose most notable features were an internal disconnection of safety ground and, of course, lack of a UL label.

Generally, the most dramatic and cost-effective solution to system noise is to locate and eliminate the ground loops or other problems that allow noise to couple into signal paths in the first place. This approach solves the fundamental problem, which tampering with safety grounds does not.

Bill Whitlock has served as president of Jensen Transformers for more than 20 years and is recognized as one of the foremost technical writers in professional audio.

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Posted by Keith Clark on 05/31 at 11:56 AM
AVFeatureBlogStudy HallAnalogAVInstallationInterconnectMeasurementPowerSignalPermalink

To Infinity & Beyond: The Intersection Of Modern Production Technology

There’s no mistaking that our beloved entertainment and communication industry is in a state of transition; the question is the direction it’s headed

As the world continually progresses, new technologies and methodologies inevitably present themselves.

Most developmental efforts are intended to aid technology users improve their work output in respect to quality and efficiency, while also aiming to make a profit.

This cycle occurs in every industry from heavy construction to microscopic surgery. We, as the providers and operators of entertainment and communication equipment, are constantly in the midst of it all.

And we must adapt to these changes if we wish to stay relevant.

Original ideas come from many sources. Often, it’s a manufacturer that envisions a better way to accomplish a given task, and that vision drives the creation of a new product. Smart phones are a recent, topical example.

Other times, end users pick up the torch and run with it, wishing to make their workflow more efficient and themselves more marketable. Think smart phone apps, frequently developed by individuals or small groups, which foster productivity and convenience while sometimes also earning income for their creators.

The Urge To Merge
Humans are builders, developers, creators, designers, and informed users. It’s in our DNA and it’s there to stay.

Case in point: a new way of looking at the big picture is emerging in professional production. Formerly disparate practices (think sound versus lighting versus video) are beginning to intersect.

Soundcraft’s recently debuted Si Performer small-format audio console includes on-board DMX lighting control, and is an example of an ongoing paradigm shift in product design and live production capability.

Imagine how this might be taken even further. What if an audio console had a second small fader and a few knobs that controlled lighting, located next to each audio fader? A single engineer could bring up a lead guitar solo in the mix, while simultaneously controlling the brightness and color of the lights that are focused on the guitar player. Ditto for keys, drums, vocalists, and so on.

It would be hard to deny that this could be a valuable intersection of technologies with a potentially wide appeal, especially in respect to budget-challenged events.

With lighting controls adjacent to audio controls, there would be no distraction to the console operator. “Tap” buttons might also be added to each channel to allow rhythmic flashing of lights. MIDI, DMX, or another protocol yet to be developed could make rapid scene changes that alter outboard audio effects, onboard plug-ins, and lighting scenes, all at the same time.

Or, perhaps the noise gates on the drum channels could be used to trigger various synchronized lighting effects within the same console, changeable as desired for different segments of a performance, either on the fly or by means of scene presets.

A future implementation might also include a button on each channel that would instantly repurpose the audio controls for lighting or video operation, thereby reducing the number of physical controls, the size of the console, and the cost of manufacturing.

Master Of The Universe
As far-fetched as it may seem right now, such “master controllers” might very well be the consoles of the future.

We’ll leave it to the product managers to conduct the market research to determine if such ideas are valid – or just vapid speculation.

But no one can deny that the economic value of employing a single system operator – instead of two or three – might make all the difference between economic success and failure for a wide range of production technology users, such as small and mid-sized churches, clubs, and various live events.

Moreover, the operator would naturally come to embrace the holistic properties of the event or presentation, rather than only concentrating on one narrow aspect such as sound or lights.

Of course this scenario is not likely to fit large events that employ large crews that are already overworked. But such events are few and far between in contrast to the thousands and thousands of modest events that take place every day, all around the world.

A Preset Is A Preset?
Numerous presentations and shows are tightly scripted. A scripted event is in sharp contrast to, well, let’s call it an “open” performance, one in which the engineer must make decisions on the fly that might vary significantly from performance to performance. Jazz artists, jam bands, some rock bands, and most artists who improvise as part of their live performance fall into this latter category.

That said, the vast number of theatrical shows, AV presentations, worship gatherings, and choreographed pop concerts are often highly predictable. Most of the heavy lifting went into setting up scene presets, usually by a gifted sound designer, and usually well in advance of the performance run.

So if scene presets are already being used to control audio, why not have the same scene presets control lighting as well? Sure, the lighting cues might not fall directly on the same clock cycle as the audio cues, but that could easily be controlled with a simple timer routine.

“Cue #10” might fade the lights to onehalf and then five seconds later change audio to a series of new settings. But it would still be Cue #10 – instead of having no relationship to the audio cue list.

Dream Come True
The concept could go further by including motion control for scenery and property changes. Theme parks have been using synchronized control for their attractions for many years; how cool would it be to have all the control programmable and operable from one console, not several that are merely sharing time code? The value and simplicity of keeping a single piece of gear in stock for emergency replacement, rather than many, would make it a maintenance supervisor’s dream come true.

Events that require complex motion control, such as major concerts and more elaborate AV presentations, could benefit by intersecting truss movement, scenery changes, fog machines, and other motion control applications into the same master system that handles sound, video and lighting.

In the future, it ’s not unlikely that we’ll see “master consoles” (with optional remote tablets, of course) that handle all of the show requirements, particularly when many of the routine tasks are per-programmed and the operator is not expected to run complex audio, lighting and video on the fly, but rather, to step through scenes that were programmed during pre-production.

The point is that significant advantages could be realized when all production aspects are managed by a master controller, instead of individual, unrelated controllers that are the norm of the methodology used today.

Where’s The Copper?
And then there are data transport systems. We now have bi-directional digital delivery systems that can carry audio, video, Ethernet, RS -232 and RS -422 control signals, production communications, and in some cases, almost any other signals that need to get from point A to point B.

Data transport systems not only preserve signal integrity, but some allow for I/O break-outs anywhere they might be needed, and without requiring additional cable infrastructure.

When you think about it, a comprehensive data transport system is just a step or two away from becoming a console, or more to the point: a futuristic master controller.

The I/O is already there, microphone preamps are already present, so the next step is to add one or more control surfaces, DSP, control software, and perhaps DMX, AVB, MIDI, NTSC, HD, and whatever else might be needed to accomplish a full suite of tasks. Voila! The new multimedia Master Control Console has just been born.

Perhaps the hardware comes from one source and the application software from another. Just as smart phones can run apps created by thousands of unrelated developers, perhaps apps will be superimposed on data delivery systems so that a client can shape them into whatever he/she might want. Now that would indeed be a true intersection of skill sets!

Same Or Different?
Will adherence to the traditional approach of dividing the workload among different production departments make this speculative outlook no more than a pipe dream? Or will the economic advantage of reducing headcount override the long-standing tradition of dividing tasks and responsibilities?

It will be interesting to see which manufacturers (and users) embrace a new approach to intersecting technologies, and which remain tied to the present mentality. Like most new ideas, there will be some who will immediately be excited while others will utterly dismiss it.

But growth and advancement is built into our genetic coding, making it highly likely that some form of the concepts we view as being far-fetched today will become standard operating procedures tomorrow.

I recently spoke with representatives from several audio console manufacturers. Some indicated that they do not see media and control intersection as being important at this time, while others said they’re watching this very closely.

But there’s no mistaking that our beloved entertainment and communication industry is in a state of transition; the question is the direction it’s headed.

As production technology inevitably marches forward, stay attuned to ways to maximize your personal value in the new paradigm of intersection.

Engineer #1: “The only constant is change.”
Engineer #2: “Yes, well it used to be that way.”
Engineer #1: “Oh. What happened?”
Engineer #2: “It changed.”

Ken DeLoria is senior technical editor for ProSoundWeb and Live Sound International and has had a diverse career in pro audio over more than 30 years, including being the founder and owner of Apogee Sound.

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Posted by Keith Clark on 05/31 at 11:14 AM
ProductionFeatureOpinionTrainingProductionAudioLightingProduction SoftwareRecordingVideoBusinessConcertConsolesDigitalDigital Audio WorkstationsInterconnectNetworkingSoftwareSound ReinforcementPermalink

Tuesday, May 28, 2013

Prism Sound Ships New Audio Interface

Lyra – A New Audio Interface For The Studio Producer, Musician and DJ Market

British manufacturer Prism Sound has announced that the first Lyra audio interfaces are now shipping to its worldwide dealer network.

“We were hoping to ship Lyra earlier in the year but due to some manufacturing issues this was delayed until now,” says Prism Sound’s Sales and Marketing director Graham Boswell. “Testing Lyra on all of the software and hardware platforms took longer than anticipated, but now we have commenced deliveries.

“I am confident that this new family of audio interfaces will delight all audio content producers including musicians, composers, project studio owners, DJs and re-mixers.”

Lyra, which made its debut at the 2012 AES Convention in San Francisco, brings to an even wider audience Prism Sound’s high performance audio converter technology.

This technology is used in the world’s leading recording facilities by recording artists and for blockbuster movie soundtracks such as Harry Potter, Lord of the Rings and the Hobbit. Based on the award-winning and critically acclaimed Orpheus interface, Lyra allows music recording professionals to access the power and sophistication of the Orpheus audio path and clock circuitry, but in a smaller package and at a much more affordable price point.

Lyra connects seamlessly with both Macs and PCs via a simple USB interface, making it ideal for recording professionals who don’t need eight channels of analogue I/O.

Prism Sound plans to launch a number of different variants of Lyra and has started the ball rolling with Lyra 1 and Lyra 2. Both incorporate new ARM Cortex processor design offering class-compliant USB interfacing, plus DSP and a low latency ‘console-quality’ digital mixer for foldback monitoring. Both products also have optical SPDIF capability and Lyra 2 also supports ADAT.

Lyra 1, which retails in the UK at £1,349 plus VAT, will be of particular interest to the musician and project studio market. This unit offers two analogue input channels – one for instrument/line and one for mic/line – plus two DA output channels and optical-only digital I/O. With Lyra 1, musicians can connect a guitar and a microphone through the input channels, plug into their software mixer via a simple USB connection and start laying down basic tracks in a matter of minutes.

Lyra 2, which retails in the UK at £1,849 plus VAT, takes the concept a little further by offering two AD input channels with switchable microphone, instrument or line input modes and four DA output channels. Both optical-only digital I/O and copper S/PDIF are available on this version of Lyra, which also offers wordclock In/Out enabling synchronization with other digital devices.

Both products are ergonomically designed to look as good as they sound. The front panel has a master volume control assignable to selected output channels, while the unit’s small size – just 11 inches wide – makes it very easy to transport for musicians, producers and DJs on the road. For studio use, Prism Sound can supply dedicated rack mounts as an extra.

“We know there is a market for Lyra because our customers have been demanding this product ever since we launched Orpheus,” Graham Boswell adds. “However, we are very protective of our reputation for delivering the highest possible audio quality so we were not going to bring any product to market until we were sure that it could live up to our exacting specifications.

“Lyra does just that, and we are very proud to introduce it.”

Lyra is fully supported by Prism Sound’s acclaimed technical and after-sales service staff.

Prism Sound

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Posted by Julie Clark on 05/28 at 02:07 PM
Live SoundRecordingNewsProductDigitalInterconnectStudioPermalink

Friday, May 24, 2013

Asterope And Fishman Support Audio Engineer Week At AES Nashville

Event included many notable audio engineers and producers

In support of Nashville Audio Engineer Week, Asterope and Fishman sponsored the 4th annual Nashville Recording Workshop + Expo 2013, held recently at the Rocketown Event Center in Nashville. The event included many notable audio engineers and producers, and introduced attendees to new Asterope technology. 

To help attendees hear the “Asterope difference,” the company created a listening environment at the event where attendees compared Asterope with competitive products in a one-on-one demonstration, using an electric guitar, acoustic guitar, or a vocal microphone.

A “cable of choice” among professional musicians, Asterope’s core line of music instrument products is also used in recording and live sound environments. The company plans to continue expanding into the pro audio market.

Later in the week, Asterope lent its support to the 16th Annual AudioMasters Benefit Golf Tournament held at the Harpeth Hills Golf Course in Nashville. The tournament is the primary fund-raiser benefiting the Nashville Engineer Relief Fund (NERF) and is a cooperative effort of the Audio Engineering Society Nashville Section and NERF, Inc.

“Both Asterope and Fishman have long-standing ties to the Nashville music industry,” said Dariush Rad, president of Asterope. “We were honored to be able to participate in the week’s events and introduce our products to such noted industry professionals.”

Asterope
Fishman

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Posted by Keith Clark on 05/24 at 02:34 PM
Live SoundRecordingNewsEngineerInterconnectSound ReinforcementStudioTechnicianPermalink
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