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Wednesday, October 19, 2011

Symetrix Grows SymNet Solus Product Line With Debut Of New Solus 16

Developed for small to mid-sized installations not requiring I/O expansion

Symetrix announces an addition to the SymNet Solus product line – the Solus 16.

“The Solus 4 and Solus 8 already provide two of the most popular form factors requested by integrators,” said Trent Wagner, senior product manager at Symetrix. “But input counts run higher in many types of installations, and we received a barrage of requests for a higher input form factor. The Solus 16 answers that request without requiring a jump to networked DSP or separate expansion I/O devices maintaining the high value for which the Solus line is known.”

Solus is powerful SymNet DSP hardware, developed for small to mid-sized installations not requiring I/O expansion.

The entire family of SymNet hardware, including Solus, is configured using open architecture SymNet Designer software.

System designers have the option to use or modify Solus DSP design templates for basic projects, or, to create unique designs entirely from scratch.

The three Solus hardware offerings differ only in their audio input and output counts:

—Solus 16 with sixteen inputs and eight outputs;
—Solus 8 with eight mic/line inputs and eight outputs;
—Solus 4 with four inputs and four outputs.

Ethernet, ARC port, RS-232 port, two control inputs, and four logic outputs complete the control feature set.

To simplify set-up, a front panel LCD displays system settings. Solus supports Symetrix ARC wall panels, third party control systems, and SymVue, a SymNet end-user control panel application.

Symetrix

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Harman Pro Offering HiQnet System Architect And AVB Training Course In November

Will take place at Harman Signal Processing in Salt Lake City on Nov 17-18

The Harman Pro System Development and Integration Group (SDIG) has announced it will be offering a training session for its HiQnet System Architect software, as well as AVB networking.

The training will take place at Harman Signal Processing in Salt Lake City, Utah on November 17-18, 2011, led by Harman product development specialist Emilian Wojtowycz, an expert in the implementation of System Architect for installed sound applications.

HiQnet is a communications protocol that enables all the compatible devices in the audio signal path, from mixing consoles to loudspeakers, to seamlessly communicate with each other.

System Architect is the software used to set up and configure a HARMAN HiQnet and AVB system.

Among the topics to be covered are the following:

·    Design workflow
·    Overview of Ethernet AVB technology
·    Advanced design workflow
·    AVB networking
·    Day-to-day operation

“Our latest version of System Architect, version 3, is ideally suited to handle the unique challenges of sound system design and reduce the complexities of networked audio routing,” notes Adam Holladay, market manager, Harman SDIG. “This training course will provide attendees with invaluable, practical, time-saving knowledge to help acquaint them with the capabilities of HiQnet System Architect and AVB.”

For more details on session times and dates, go to www.harmanpro.com. Attendees can register for the course by contacting Staci Bash at 801-568-7555 or .(JavaScript must be enabled to view this email address).

Harman Pro

 

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Master Class In Fiber Optic Audio To Be Presented By Optocore Founder Marc Brunke At AES

A discussion of the discuss the technological underpinnings as well as real-world case studies of complex and multi-faceted broadcast, studio, and live performance applications

Optocore founder and chief engineer Marc Brunke will host a presentation entitled “The Fundamentals of Audio and Data Networks over Fiber Optics and Cat-5 Cabling” at the 131st Audio Engineering Society (AES) Convention.

Specifically, the program will begin at 9 am on Friday, October 21, at Room 1E08 on Level 1 (lower level) of the Jacob Javits Center in New York, site of the AES Convention.

Brunke will discuss his views on the direction the industry is taking, the constantly expanding role of fiber optics in entertainment technology systems, and he’ll also be available to answer questions.

Different approaches and different ways of dealing with synchronization and jitter problems will be described; each point will be supported with an example from real life applications, with all present known technologies taken into consideration.

Brunke will also cover topics such as component technology advancements and their cost/performance ratio as it relates to recognized AES industry standards such as MADI, along with other current (and future) platforms. 

The program will run for 90 minutes, providing ample time to discuss the technological underpinnings as well as real-world case studies of complex and multi-faceted broadcast, studio, and live performance applications.

Brunke has almost 20 years of experience in fiber optic transport disciplines, and their influence on commercial design challenges.

For further information go to www.optocore.com. In North America, please contact Brandon Coons at the Optocore North America office at 1-(416) 287-1144.

Optocore

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Tuesday, October 18, 2011

Audio? Confusing? Learning Is A Life-Long Process

Consider the input types that may exist on a mixing console. I found all of these on units sitting around the shop...

Almost every Syn-Aud-Con seminar has attendees from other technical fields that need to learn about sound systems and audio.

These fields include networking, telephony, lighting, electrical and others.

Many tell us that audio is the most confusing thing they have encountered in their technical careers – and it is no wonder.

Consider the input types that may exist on a mixing console. I found all of these on units sitting around the shop.

There are nine (9) analog topologies and twelve (12) digital topologies. Each serves a purpose. Each works fine for its intended application. Each is defensible from a technical and practical point-of-view. Each will likely remain in use as other connector types and topologies emerge.

Consider also that some of these have several variations, such as the polarity convention on an XLR connector or AES3 on a DB25 connector.

It is ironic that digital I/O is often touted as making things easier, yet there are more digital connector types than analog! Add to this the confusion caused by the need to configure digital I/O for the correct sample rate, bit-depth, etc.

It’s no wonder that noise and distortion remain the weak links of most sound systems. They often result from feeding the wrong signal to the wrong jack.

We have all heard a DVD player over-driving a microphone input. Yes, you get sound, but in audio the presence of sound does not necessarily mean that you hooked it up right.

A modern digital mixer may be able to convert between any of these formats.

The signal may come in as some form of analog and go out as some form of analog or digital.

The user must often choose based on the required cable length, input options on the next device or some other criteria.

So not only does the audio tech have to understand the connectors and interface topologies, he must also know the characteristics of the devices at the other end of the cable.

The interconnect is the easy part. Consider the knowledge and experience required to configure a DSP for a 3-way loudspeaker.

Technical complexities aside, the most perplexing part of audio for non-audio people is the artistic side. While some levels can be set with voltmeters and analyzers, many other adjustments are based on subjective criteria – you just turn the knob until it sounds right. But what is right? There can be any number of “rights.” How confusing is that!

The really good audio people have strong theoretical and practical backgrounds. They have “must have” tools in their toolbox that you can’t buy anywhere. They have the ability to diagnose many system problems by just listening to a speech track played over the system. They can often make it sound way better by turning one knob a little bit.

Their personal study time may be divided between Sound System Engineering, product manuals and Einstein’s papers on relativity, and they understand that all three are completely relevant to what they do.

Most importantly, they know that they will never know it all.

Yes, audio is confusing. Background in another technical field can help, but learning audio is a life-long process.

Pat and Brenda Brown own and operate SynAudCon, conducting training seminars around the world in addition to providing in-depth web-based training.

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Wednesday, October 12, 2011

Optocore Unveils Y3R-TP SANE Card For Yamaha Digital Consoles

Designed for Yamaha mini-YGDAI slots, providing an interface on Optocore's SANE network

Optocore has introduced the Y3R-TP SANE card for Yamaha digital consoles, and it will be on display at the Optocore booth (# 667) at the upcoming 131st AES Convention at the Javits Convention Center in New York City.

The new Y3R-TP card is designed for Yamaha mini-YGDAI slots, providing an interface on Optocore’s SANE network. It features 16 input/16 output audio channels through these mini-YGDAI slots, with word clock to/from the slot, to/from the network, and word clock transport to any device in the network.

The two SANE ports enable connection of multiple Y3R cards to the stand-alone SANE network or as a Cat5-based extension to a 2Gbit Optocore ring.

There is also a LAN slot, since all Optocore devices create a 100Mbit LAN network, and a 10/100Mbit Ethernet virtual switch to any Ethernet-enabled Optocore device.

Also incorporated is a USB/RS232/Remote port for direct transport of the Yamaha HA Remote protocol via Optocore.

Finally, the card offers Yamaha Emulation Mode, making it possible to control Optocore preamps from four different consoles.

Also at AES, Optocore creator and president Marc Brunke will be hosting a tutorial entitled “Fundamentals of Audio and Data Networking Over Fiber Optic and Cat5 Cables,”  where he will look at the various networking technologies on the market today and how they address key points such as word clock, latency and distribution. This session will be held in room 1E08 of the Javits Center on Friday, October 21 at 9 am.

Optocore

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Extron USB Matrix Boards For SMX System MultiMatrix Now Shipping

Designed to route up to eight Host CPUs to up to four peripheral locations equipped with one or more USB 2.0 devices

Extron Electronics has announced the immediate availability of two new USB matrix switcher boards for the SMX System MultiMatrix modular, field-upgradeable matrix switcher.

The SMX USB matrix boards are designed to route up to eight Host CPUs to up to four peripheral locations equipped with one or more USB 2.0 devices, such as keyboards and mice, web cams, personal media players, or portable hard drives.

They support data transfer rates up to 480 Mbps and are compatible with USB 2.0/1.1/1.0 specifications. Host and Peripheral Emulation is provided on all ports for reliable, problem-free boot up, even without a tie being made to a connected device.

SMX USB matrix switcher boards are ideal for use in the creation of KVM - keyboard, video, mouse matrix applications when combined with available SMX DVI, HDMI, or VGA matrix switching boards. SMX USB matrix boards are available in two I/O sizes: 4x4 and 8x4.

“For KVM applications, the flexible, modular design of the SMX MultiMatrix provides the ideal platform,” says Casey Hall, vice president of sales and marketing for Extron. “With the wide range of HDMI, DVI, VGA, audio, and now USB matrix switcher boards available for the SMX, AV integrators can create unique and versatile KVM systems that match their customers’ needs.”

The SMX System MultiMatrix Series of digital and analog multi-plane matrix switchers combines multiple, independent matrix switchers in a truly modular, field-configurable frame.

SMX frames are available in sizes from 2U up to 5U that are capable of supporting up to 10 separate matrix boards, which can be switched independently or simultaneously, under a single point of control.

Applications include medical imaging systems, conference and training facilities, and other mid-sized applications that require the switching of different signal types, and it is a cost-effective upgrade path for ongoing I/O or signal format changes.

Extron

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Tuesday, October 11, 2011

Lectrosonics ASPEN Processor Deployed At Florida Emergency Operations Center

In addition to managing all the audio and signal processing for the command center room, the ASPEN processor handles paging and audio routing throughout the entire complex

The new Clay County Emergency Operations Center (EOC) in Green Cove Springs, Florida (just south of Jacksonville), which commenced operation in May, incorporates a Lectrosonics ASPEN SPN1624 16-input/24-output audio processor to manage a wide range of audio signals inherent in a facility of this nature.

Jacksonville-based Florida Sound Engineering Company, a commercial A/V design-build firm serving the greater North Florida and South Georgia region, was contracted to handle the design and deployment of the EOC’s new audiovisual system.

Robert A. Cole, the firm’s president and owner, and project manager Kevin Schnarr, selected the Lectrosonics ASPEN processor over a competing system during the design stage of the project. Cole discussed those attributes that made the ASPEN processor the best choice.

“For the system we designed,” explains Cole, “the mixer must service the entire facility, which is separated into eleven audio zones. This mixer needed to accommodate eight computer audio inputs, three wireless microphone inputs, two cable TV receiver inputs, two BluRay player inputs, and an input for telephone paging. Control of the system is handled by a Crestron Rack2 controller with two wireless touch panels and four wall-mounted wired touch panels.

“Given all that we had to pull together, we were very impressed with the fact that the Lectrosonics ASPEN processor integrates mixing, signal routing, audio processing, and control into a single unit. This was a distinct advantage over the primary competing system we looked at that would have required more of a ‘system assembly’ approach.”

The Clay County EOC is a two-story building consisting of 75 rooms on the first floor and 36 rooms on the second floor. Housed in the building is an office for the Clay County Fire Chief, an office for the Clay County Communications Director, a 12-station Communications/Dispatch Center, and a 36 seat Incident Command Center, along with offices for support staff.

The main control center measures roughly 60 feet by 120 feet and consists of two large screen projectors and six large tables with computers and ancillary equipment. In addition to managing all the audio and signal processing for the command center room, the ASPEN processor handles paging and audio routing throughout the entire complex.

“During the programming stage,” adds Cole, “we encountered some challenges in terms of controlling the ASPEN over RS-232 via the Crestron in the manner we had originally envisioned. Our programmers contacted Lectrosonics’ technical support staff, who worked with our team to help them gain the control we were looking for. They also helped us write some macros for a few routines we wanted to incorporate.

“The Lectrosonics support team was extremely helpful and responsive. I was very impressed with their ability to quickly understand what we were looking to accomplish and provide the guidance we sought.”

With the Clay County EOC fully operational, Cole reports that his client has been extremely complimentary of the overall A/V system design and its ability to manage the wide range of audio sources.

Florida Sound Engineering Company
Lectrosonics

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Thursday, October 06, 2011

Aviom Pro64 Network Manager Version 2.0 Now Available

Update delivers power and flexibility of crosspoint switch matrixing and supports new AllFrame Multi-Modular I/O system

Aviom has announced the availability of version 2.0 of the Pro64 Network Manager PC control application, which delivers the power and flexibility of a full crosspoint switch on both inputs and outputs throughout a Pro64 audio network.

Utilizing Network Manager, any audio input or output can be assigned to any Pro64 network audio slot.

Version 2.0 also adds support for Aviom’s AllFrame Multi-Modular I/O System, the newest addition to the Pro64 Series of audio networking products.

A field customizable and configurable modular digital solution, the AllFrame is comprised of a host frame, audio I/O cards and a range of mounting options.

“This latest release of the Network Manager brings new flexibility to the Pro64 system,” says Ray Legnini, Aviom product research and development manager. “With a crosspoint switch at every network node, users can matrix audio however they want and make more efficient use of hardware resources and more dynamically distribute audio throughout their system.”

Pro64 Network Manager is a free download from Aviom.com/NetworkManager and includes firmware upgrades for all Pro64 Series devices.

Aviom 

 

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Wednesday, October 05, 2011

Roland Systems Group Now Shipping The R-1000 Multi-Channel Recorder/Player

Captures up to 48 channels of discrete audio all as separate broadcast wave files ready to open in a DAW of choice

Roland Systems Group has announced the shipping of R-1000 48-track recorder/player through the company’s network of authorized resellers.

The R-1000 is a stand-alone, dedicated recorder/player designed to work with the V-Mixing System for live events and productions. Setup and control parameters of the R-1000 can be done directly from any V-Mixer or through the PC/Mac control utility (R-1000 RCS).

In addition, the R-1000 can be connected and used with any digital console that has MADI output capabilities by using the Roland S-MADI REAC MADI Bridge.

The R-1000 captures up to 48 channels of discrete audio all as separate broadcast wave files ready to open in a DAW of choice. As a playback device it can be used in live events to play back selected channels to augment a live performance or as a multi-channel playback deck in a theater or amusement park application.

Two units can be synched for a 96-channel recorder/player, or synched to video with SMPTE (LTC), or via black burst. All files are stored on the included 500GB removable hard disk drive (HDD. Material can also be transferred via USB to a connected drive.

The R-1000 also fosters virtual rehearsals when integrated with a Roland V-Mixer console. Not only does it become a powerful training tool, it greatly reduces sound check time for bands/productions.

Using a song previously recorded on the R-1000, switch to playback mode and all the sources play back through the appropriate channels on the console. Adjust the preamp gains on the console as you would if the band was live and the R-1000 takes care of the gain compensation. Then set compression, EQ, monitors, and effects. When the band takes the stage you can be confident it will sound the way it did during the virtual rehearsal.

Setup and configuration can be done using the color LCD touch panel on the front panel or with the PC/Mac Remote Control software via a USB connection.

The R-1000 is based on REAC (Roland Ethernet Audio Communication) and eliminates the bulk and noise susceptibility typically associated with analog snakes and replaces it with Cat-5/6 (Ethernet/LAN) cable.

The R-1000 records superior audio by capturing the converted sound connected to the Roland digital snake systems. Analog inputs and high-quality mic preamps are located close to the source where audio is immediately converted to 24-bit digital streams and sent over Ethernet.

Using REAC, the pristine digital audio signal is transferred throughout the complete system path en route to the R-1000 hard drive and then back to any outputs and on to limitless split positions.

Roland Systems Group

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Friday, September 30, 2011

Aviom Now Shipping New AllFrame Multi-Modular I/O System

Eliminates several expensive and labor-intensive stages of the signal chain while reducing cable clutter

The new Aviom AllFrame Multi-Modular I/O System is now shipping. It is suitable for multiple applications, bringing the company’s Pro64 audio network to wall boxes, stage boxes or floor pockets.

Designed for houses of worship, theaters, schools and conference centers, as well as touring and portable live sound applications, the AllFrame Multi-Modular I/O System eliminates several expensive and labor-intensive stages of the signal chain while reducing cable clutter.

Each device supports Cat-5e cable runs up to 400 feet (120 meters) between devices, with no loss of fidelity. Since it requires only a single Cat-5e or fiber connection, it eliminates the need for soldering, terminating and testing scores of analog connections as well as installing conduits and pulling separate cables for each audio signal, significantly reducing setup time.

At the heart of the system is the F6 Modular I/O Frame, a multi-purpose network frame with six field-configurable I/O card slots and integrated Cat-5e and fiber optic connectivity. The F6 can be outfitted to meet the audio I/O requirements of nearly any application.

The unit can be mounted on a wall or in a wall using a standard NEMA Type 1 enclosure. Using the RK6 rack mounting kit the F6 can be rack-mounted, with connectors facing the front or rear.

Other mounting kits that will be available soon include the FK6 and SK6, which allow for AllFrame I/O boxes to be installed in existing floor pockets or used as drop-boxes on stage.

The F6 can be powered through a four-pin XLR connector, a Euroblock connector or one of the Cat-5e A-Net ports. Installations that use Power Over A-Net will require the POA80 Power Over A-Net power supply.

The system offers users multiple connectivity options via modular I/O cards. The C4m Mic/Line Input Card provides four remote-controllable analog mic- or line-level inputs. With the same mic preamp circuitry found in Aviom’s 6416m Mic Input Module, the C4m provides clean, transparent, archival quality sound.

Inputs from the C4m can be controlled from an MCS Mic Control Surface, select Yamaha digital consoles or Pro64 Network Manager software. The C4o Output Card provides four XLR analog outputs with variable output levels.

For those looking to stay digital throughout the signal chain, there will be a C4dio Digital I/O Card with 4x4 AES3 I/O with Word Clock I/O.

“We couldn’t be more pleased to announce the availability of the AllFrame for purchase,” says Carl Bader, president and CEO of Aviom. “Having just celebrated our 10th year in the industry, we look forward to continuing to offer streamlined solutions for our customers and that includes the AllFrame. The system was designed with customer research in mind to help make digital more affordable and flexible, keeping analog cabling to a minimum and reducing system complexity and labor costs.”

“Our customers will appreciate the fact that the AllFrame can be used in conjunction with the complete line of Pro64 audio networking products and Pro16 Series Personal Mixers,” he adds. “Because Pro64 supports any combination of serial and parallel wiring topologies without affecting signal flow at any point, I/O points can be placed anywhere the application requires.”

Aviom

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Thursday, September 22, 2011

Addition Of Dante Card To Allen & Heath iLive Enhances Audio Flexibility At Florida Church

Dante provides full 64-channel bi-directional signal routing between the iLive’s MixRack I/O center and an external Ethernet device via a single Cat-5 cable

Like many house of worship facilities with a strong media ministry, First Baptist Church of Altamonte Springs, Florida, was anxious to upgrade its audio system.

With an Allen & Heath iLive digital mixing console already in place, the church was interested in expanding the system’s extensive capabilities in a way that would simplify training of its volunteer audio staff.

John Williams, worship & media director for the church, engaged Entertainment Arts, an Orlando-based design and integration firm, to find and integrate a cost-effective solution.

Digital audio systems for house of worship applications range from straightforward to extremely complex, and from affordable to outrageous in cost. In general, the key factor between these extremes has been the amount of signal routing required. The more channels and destinations desired, the more complex and expensive a system is required.

However, Allen & Heath now offers the M-DANTE card, a plug-and-play audio networking solution for its popular iLive mixing infrastructure. This solution uses the Dante protocol, an Ethernet-based digital audio networking system by Audinate, an Australian company.

Dante provides full 64-channel bi-directional signal routing between the iLive’s MixRack I/O center and an external Ethernet device via a single Cat-5 cable. The Dante card streams a full split of the console’s inputs between the MixRack and the destination device.

For First Baptist Church of Altamonte Springs, that destination was a computer with recording software, enabling discrete 64-by-64 channel recording and playback.

“We love being able to multi-track record our rehearsals and worship services,” reports Williams. “With Dante’s virtual Sound Check, we can actually go back and adjust the monitor mix for the praise team members without holding up the rehearsal.

” It’s also great for training our volunteer tech ministry. By playing back the full multi-track recording, our team members can re-produce the entire original performance and learn to operate the iLive console without the pressure of doing it during rehearsals or services. It’s a phenomenal tool.”

The Dante protocol leverages common Ethernet connectivity over a standard IP computer network to deliver full-bandwidth audio with latency of less than one millisecond, whether using a 100 MB or a full Gigabit connection.

The key to its flexibility is the inherent bi-directional nature of the connection, which means that a full signal split can be sent out from the iLive console to an external device (in this case, a computer with recording software), while also allowing the full signal path to be returned to the originating iLive system for mixdown or playback.

The Dante card comes complete with a license for Audinate’s Virtual Soundcard software, which turn’s a computer’s Ethernet port into an audio interface with plug-and-play simplicity. The Allen & Heath Dante module enables flexible routing and distribution of audio between devices on an existing computer network while eliminating the technical challenge of IP configuration from the process.

The M-DANTE card can be ordered as an option on any new Allen & Heath iLive digital mixing console, or can retrofitted into any Allen & Heath iDR MixRack to significantly expand its functionality.

Allen & Heath
American Music & Sound

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Intercom Goes Real-Time IP With Riedel AVB Solutions

Intercom applications for Riedel’s AVB products feature matrix-to-control panel connections via LAN, audio distribution via LAN, matrix-to-matrix trunking connections via LAN and distribution of digital partylines via LAN

Riedel Communications has premiered a suite of AVB products for the Artist digital matrix intercom platform..

AVB allows for transporting AES3/EBU audio in real-time with guaranteed bandwidth and quality of service (QoS) via IP-based Local Area Networks (LAN).

Connecting intercom panels over an IP-based LAN environment has been the dream of many system planers. But issues inherent to IP like latency, reliability and missing synchronization prevented them from doing so.

Riedel’s AVB product line overcomes these issues and provides a real-time communication solution fulfilling the demands of professional intercom users.

Based on official IEEE next generation Ethernet standards like 802.1Qav, P802.1Qat and P802.1AS, AVB allows risk-free utilization of AVB compliant facility or enterprise LAN infrastructure for intercom applications. This allows for new approaches in system and facility design providing significant savings in infrastructure investments.

Intercom applications for Riedel’s AVB products feature matrix-to-control panel connections via LAN, audio distribution via LAN, matrix-to-matrix trunking connections via LAN and distribution of digital partylines via LAN.

The Riedel suite of AVB products includes the AVB-108 G2 Client Card as well as the Connect AVB and Connect AVBx8 panel interfaces. The AVB-108 G2 card is a regular Artist client card to be used inside the Artist mainframe. It converts eight Artist matrix ports into AVB and vice versa.

The AVB-108 G2 client card communicates either with other AVB-108 G2 client cards in another Artist systems, i.e., for trunking, or with Riedel’s Connect AVB and Connect AVBx8 panel interfaces.

The Connect AVB Panel Interface is a small unit, which converts an AES signal into AVB and vice versa. It is designed to connect one Artist control panel in one or two-channel mode to the intercom matrix via IP-based LANs.

Connect AVBx8 is the big brother of the Connect AVB interface, converting eight AES signals to AVB and vice versa. Built in a compact 9.5-inch/1RU housing the device provides eight CAT5 ports to connect up to eight Artist control panels in one or two-channel mode to the matrix via IP-based LANs.

Connect AVB and Connect AVBx8 are ideally suited for Riedel’s AVB-108 G2 eight channel AVB client card.

Riedel

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Monday, September 19, 2011

Audinate Releases Mac OS X Lion Compatibility For Dante Virtual Soundcard

Can transmit and receive up to 64 channels between networked audio equipment and applications such as Nuendo, Cubase, Logic and Pro Tools 9

Audinate has completed support for both Windows and Mac 64 bit operating systems with the availability of Dante Virtual Soundcard for Mac OSX 10.7 Lion.

The Dante Virtual Soundcard software provides direct connection of a computer to a Dante audio network.

Dante Virtual Soundcard uses the Ethernet port on the computer to communicate with a network of other Dante enabled devices. 

No special hardware is required other than installing Dante Virtual Soundcard on a conventional computer (or laptop).

The Dante Virtual Soundcard can transmit and receive up to 64 channels between networked audio equipment and popular audio applications such as Nuendo, Cubase, Logic and Pro Tools 9. These audio applications use the Dante Virtual Soundcard as they would any standard ASIO or Core Audio sound card.

The Dante Virtual Soundcard has no physical audio inputs or outputs. Instead, audio signals are sent directly to or received from the PC’s Ethernet network interface card.

Mary Cudmore, Audinate director of products, states, “We’ve seen a substantial increase in the use of the Dante Virtual Soundcard and now having compatibility with Lion will open the way for more users to access the power of the application.”

Dante Virtual Soundcard V3.2.0 for MAC OSX 10.7 is now available as a download at the company website.

Audinate

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Friday, September 16, 2011

Adamson Systems Unveils “Project Energia”

Key components include loudspeaker systems with networkable Class D amplifier modules, DSP, cable and power distribution, AVB network hardware with software integration of control, and 3-D simulation and diagnostics

Adamson Systems Engineering has announced the commencement of the much anticipated Project Energia. 

The project was conceived after years of analyzing inefficiencies found in sound reinforcement and with performance venue optimization. 

The key components of Energia include a new series of loudspeaker systems with networkable Class D amplifier modules, DSP, cable and power distribution, AVB network hardware with software integration of control, and three-dimensional (3-D) simulation and diagnostics. 

To ensure reliability and a smooth integration of Adamson technology, there are three phases involved in the release of the first loudspeaker system in Project Energia:

Phase 1 - Mechanical Field Testing
Phase 2 - Class D Amplifier, Power Distribution and Ground Control Field Testing
Phase 3 - Network and Network Hardware Field Testing

In July of this year, Phase 1 began when Adamson unveiled the E15 Line Source system with a series of strategic beta partners. Eighth Day Sound (USA), Wigwam Acoustics (UK), Fluge (Spain) and Big Daddy Productions (Indonesia) have all taken delivery of the E15 Line Source Array. The new system will be found on a variety of fall tours and large format events around the world. 

Additional Energia beta systems are now in place with distribution partners DV2 in France, Adamson Europe GmbH in Germany and in Singapore with long time partner Team 108.

Adamson Systems President and CEO Brock Adamson states, “Phase 1 of Energia is designed to ensure the highest acoustic and mechanical performance of the E15, while we simultaneously begin testing our new three-dimensional Blueprint simulation software. Once we’re finished evaluating transducers, sound chambers and the mechanical elements of the system, we will introduce our new Class D amplifier modules, power distribution and ground control system for Phase 2. 

“We did not want to simply re-package our old systems and offer them as though they were something new,” Adamson continues. “On the contrary, we wanted to provide something exciting that hasn’t been done. Pushing the boundaries of technology is what makes our industry great. Our team is very pleased with the feedback and support coming from our field test partners so far. We couldn’t ask for a better team of industry professionals to help us bring Project Energia to life.”

A sneak peak of the new system can be found on the Adamson web site and Facebook page.

Adamson Systems

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Tuesday, September 13, 2011

Church Sound: Developing An “Easy Button” For A Student/Community Room PA

The whole system ran a little over $1,500, including controllers, and it took me about an hour to set up all the programming

 
Last week I was able to finally deploy the new EZ Mode (our term) for our student/community room PA.

The room is our most-used room every week, supporting everything from Jr. & Sr. High to men’s and women’s ministry to MOPS. There is something happening in that room every day of the week.

The system is relatively simple, but I’ve received more than one call at home on a Tuesday night from someone trying to make a mic work, or get sound out of the iMac.

I began looking for a solution and found it in the Symetrix Jupiter 8 processor and a set of associated wall controls, the ARC2 and the ARC-SWK. I’ll write up a review of the Jupiter in a separate post. For this one, I’ll focus on how I used the wall controls to create an EZ Mode.

It All Starts With Planning
My first step in the process was to sketch out what I wanted to accomplish. Based on the way the room is used, we really needed two major modes of operation, with two subsets each.

Big picture, it looks like this:

—Mix Mode, Delays On
—Mix Mode, Delays Off
—EZ Mode, Delays On
—EZ Mode, Delays Off

The room is a classic “two rooms in one with an air wall” layout; meaning sometimes we need the whole room, other times, just half. The main PA (hung over the stage) is the Electro-Voice LiveX 15s and subs I’ve written about before. We also hung some EAW JF80s (because we had them lying around) as delay loudspeakers to fill in the back half of the room on the other side of the air wall.

Student ministries runs a full band for their events, so they need to be able to mix a full compliment of inputs on the Yamaha MG32 we have in there. Most of the other ministries/events however, require one or two mics, audio for video (either DVD or the iMac) and an iPod input.

Since the Jupiter 8 has 8 inputs and 8 outputs, and the inputs can be either mic or line, I set about making up a plan.

The input side looks like this:

—1&2 Stereo In from the MG32
—3&4 Wireless Mics 1&2 (also double patched into the MG32)
—5&6 Audio from Video (will eventually be the audio output of an Extron IN1508, for now, it’s a double-patched ProAV2)
—7&8 A dedicated “EZ” iPod cable.

Outputs 1-4 feed the main speakers (L&R plus L&R Subs), while Output 5 feeds the delays. Output 6 feeds nothing, Output 7 feeds a CD recorder, while Output 8 is used to control the logic output that turns on and off the delay speaker’s amp. More on that later.

Basic Programming
The first thing we did was to get a baseline layout in all the DSP.

Inputs and Outputs were labeled, patched routed and gained.

We set up our crossovers for the main speakers and dialed in system EQ. We set up the delays and got their EQ where we wanted it.

That formed the basis of our programming.

The next step was to build presets that turn inputs and outputs on and off.

Preset 1 is Mix, Delay Off, so we muted inputs 3-8, the delay output and turned off the delay amp. Preset 2 (Mix, Delay On) was created by un-muting the delay output and turning the amp on.

Preset 3 is EZ, Delays Off. To create this preset, inputs 1&2 are muted, meaning the output of the MG32 is completely ignored by the Jupiter.

We unmute inputs 3-8, which enable the two mics, video and the iPod cable. As with Preset 1, the delays are off. Preset 4 is like Preset 3, only with the delays on.

Once that was all set up, tested and found to be working, we hooked up the two controllers.

Controlling

The ARC2 is a menu driven controller. It’s extremely powerful and enables you to control quite a few parameters inside the Jupiter. I could have done everything I needed with this box, but figured the addition of an ARC-SWK would make the system easier to use.

In my setup, the ARC2 does one thing - enable users to switch between the four operating modes.

This is accomplished in the Jupiter software; simply add a controller, create a menu, and load the presets. You can edit the labels to make it easy to navigate.

Once it’s all assigned, you can simulate the controller to visually ensure it’s all working the way it’s supposed to.

The next step is to add in the ARC-SWK controller. The SWK is a 4-button, single encoder remote with an A and B side. This means you can control up to 8 parameters very easily.

Here is how ours is set up:

—Button 1A: Wireless Mic 1
—Button 2A: Wireless Mic 2
—Button 3A: Stereo Audio for Video
—Button 4A: Stereo Audio for EZ iPod Cable

The software makes it easy to control inputs and outputs as mono or stereo channels. At the moment, I don’t have any need to control anything else, though I have the capability to control four more parameters if need be.

When controlling volume, you can specify minimum and maximum values - initial values are stored in the preset. It’s all very easy to do, and took less than 5 minutes to assign everything.

The ARC series of remotes connect daisy-chain style to the Jupiter over Cat5. You can also send audio through certain wall panels, either in or out, depending on the model.

Final Programming
What sold me on the Jupiter is the calendar feature. Once all the presets are built, you can create events (single or repeating) that will automatically switch modes.

So in our case, on Tuesday morning, the system goes into EZ, Delays On at 8:45 AM for the Women’s Bible Study. At 3:00 PM on Wednesday, it switches to Mix, Delay Off for Jr. High. On Thursday at 8:30, it switches back to EZ, Delay On for MOPS.

Eventually, I will add the Jupiter to the network so I can access it from anywhere and create custom events (like next Friday when the Boy Scouts use the room).

After all that, I dove into the logic outputs.

The Jupiter has 4 dual-mode logic outputs. Each logic output can deliver 5 VDC for connecting an LED indicator, or act as a simple contact closure (alternately, you can just use the +5VDC to close an externally powered relay).

The logic outputs are assigned to parameters anywhere in the system.

In our case, to make programming easy, I assigned the control to Output 7, which we weren’t using anyway.

When Out 7 is muted, the delay amp turns on (using a Furman Relay). When it’s unmuted, the amp turns off. The two mute states get saved into presets, and just like that, the amp turns on and off as if by magic.

The Cost
Doing something like this used to require a Crestron or AMX system, and programming could easily run into the thousands of dollars, not to mention the additional costs every time you wanted to make a change or the equipment cost. In this case, the whole system ran a little over $1,500, including controllers.

It took me about an hour to set up all the programming (and another few hours to tune the system).

It’s easy enough to use that any TD will be able to get the system doing whatever they want in no time. I was able to train our entire staff on the EZ Mode of operation in about 10 minutes, and the documentation takes just two pages (and half of each page is a large picture of each controller).

Best of all, I won’t have to take any more calls like this one…

Caller: Mike, the mics aren’t working.

Me: OK, so go to the fader labeled RF-A. Make sure the button above that fader is turned on and lit up.

Caller: [Long pause, obviously frustrated] I…I…don’t even know what a fader is.

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

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