Interconnect

Wednesday, June 11, 2014

Pesky Ground Loop Problems Plaguing You?

When a system contains two or more pieces of equipment that are grounded, whether via power cords or other ground connections, a “ground loop” will likely be formed. (See Figure 1, below.)

Although ground loops often involve power line safety ground connections, disabling them is both highly dangerous and illegal.

However, devices called “ground isolators” can be inserted in the signal path to break the loop safely. This approach attacks the problem at its fundamental roots, while tampering with safety ground does not. In simple language, a ground isolator is a device that transfers a signal across an electrically insulated barrier.

This is how it stops the flow of power-line currents that would otherwise generate noise as they flow through signal cables. Because an isolator is not a filter that recognizes and removes noise, it must be inserted in the signal path at the point where the noise coupling actually occurs.

Figure 1: See the ground loop in this home theater system?

On the other hand, a transformer can serve as an extremely effective ground isolator. As shown in Figure 2, it transfers signal voltage from one winding to the other without an electrical connection between them. This electrical isolation blocks the flow of ground noise current in the signal cable.

While the isolation would be total for an ideal transformer, physics imposes limitations on real-world transformers.

Two Basic Types
In practice, noise reduction depends critically on the design of the transformer. Audio transformers fall into two basic types.

The first, known as an output transformer, is by far the cheapest and easiest to build. Because its primary and secondary windings are physically interleaved, considerable capacitance is created which allows noise currents, especially at higher audio frequencies, to flow between windings. This limits its ability to stop ground noise.

Figure 2: A transformer can serve as an extremely effective ground isolator, transferring signal voltage from one winding to the other without an electrical connection between.

The second type, known as an input transformer, is built with internal metal foil shielding between its windings. This “Faraday shield” effectively eliminates capacitive coupling and vastly improves noise rejection. A magnetic shield serves a completely different purpose and, if used, is on the outside of a transformer surrounding both the core and the windings.

Figure 3 shows noise rejection versus frequency for a typical unbalanced interface. With no isolator, by definition, there is 0 dB of rejection in the interface, as shown in the upper plot.

The middle plot in Figure 3 shows results for a typical isolator using an output transformer. Hum at 60 Hz is cut by 70 dB, but buzz artifacts around 3 kHz are reduced by only 35 dB. The lower plot shows results for a typical isolator using an input transformer. Hum is cut by over 100 dB and buzz by over 65 dB.

Figure 3: Noise rejection versus frequency for a typical unbalanced interface.

The overwhelming majority of “black boxes” intended to solve ground loop problems use output transformers. One advantage of these boxes is that they can be installed anywhere along the length of a cable or can be used at patch-bays. Although boxes made with input have some 30 dB better noise rejection, they must be installed thoughtfully.

In Figure 4, we see a commercial black box. Because high-frequency response can be degraded by excessive cable capacitance at their outputs, these types of boxes must be installed near the equipment input they drive, generally through no more than 3 feet of cable.

Some commercial interface devices are “active” (i.e., powered) devices. Although these often have useful features, they invariably use differential amplifier circuits to “isolate” their unbalanced inputs.

Figure 4: A Jensen ISO-MAX transformer, up close and personal.

In a future discussion of balanced interfaces, we’ll find that ordinary diff-amps do this job very poorly. Typical products in this vein often deliver only 15 dB to 30 dB of noise rejection under typical real-world conditions.

Incidentally, to eliminate noise in an unbalanced cable run, it’s not necessary to “balance” the line (using a converter at the driving end) and then “unbalance” it (using another converter at the receiving end).

The noise rejection of such a scheme is no better, and often worse, than that of a single high-performance isolator (i.e., input transformer) installed at the receiving end.

Check performance data for isolators carefully. Many have scanty, vague or conspicuously non-existent specifications, and many use cheap, telephone-grade transformers.

These can cause loss of deep bass, bass distortion, and poor transient response. Data for high-quality isolators is complete, unambiguous, and verifiable. Input-transformer-based isolators have other benefits, too, including:

• Their inputs are truly universal, accepting signals from either unbalanced or balanced sources, while maintaining extremely high noise rejection

• They provide inherent suppression of RF and ultrasonic interference. The subsequent reduction of “spectral contamination” is often described as a marvelous new sonic clarity

• They are passive, requiring no power

• They are inherently robust, reliable, and virtually immune to transient over-voltages.

Explore The Options
In many systems, including the one seen earlier in this article in Figure 1, there is more than one way to break the ground loop.

Observe that the noise voltage between the CATV ground and the AC power safety ground at the subwoofer causes noise current flow in the shield of all the signal cables between the CATV ground and the subwoofer.

Figure 5: Two safe ways of breaking ground loops. The figure above shows a ground isolator inserted in the audio signal path between TV and subwoofer, the figure below depicts a potentially less expensive method of inserting the isolator in the CATV feed.

Common-impedance coupling will induce noise in both audio cables in the path, generally in proportion to their lengths. CATV feeds are notorious for having “ground” at their shield several volts different from utility AC power ground, so this system might exhibit a very loud hum regardless of preamp control settings because of coupling in the 20-foot cable.

Of course, the loop could be broken by defeating the subwoofer safety ground—but don’t do it! Remember, audio cables that connect equipment together will also carry lethal voltages throughout the system or could start a fire if the subwoofer develops a defect.

A safe way to break the ground loop is to install a ground isolator somewhere in the audio signal path from TV to subwoofer. Because longer cables are more likely to couple more noise, the preferred location in this system would be at the receive end of the 20-foot cable (Figure 5, above). Another safe, and potentially less expensive, solution is to break the loop by installing a ground isolator in the CATV feed as shown in Figure 5, below.

CATV isolators must be installed downstream of the lightning ground and should generally be installed where the cable first connects to the audio or video system, such as at a VCR or TV input.

 
Bill Whitlock has served as president of Jensen Transformers for more than 25 years and is recognized as one of the foremost technical writers in professional audio.

{extended}
Posted by Keith Clark on 06/11 at 12:05 PM
AVFeatureBlogStudy HallAVInterconnectPowerSignalSystemPermalink

Bose Pro Integrates Configuration For Dante Devices In New Version Of ControlSpace Designer Software

Also allows pre-programmed Dante audio routing to be recalled from any of the Bose user controls, or via a connected third-party control system

Bose Professional Systems is making it easier to connect its ControlSpace processors and PowerMatch amplifiers using Dante digital audio networking, with new version 4.1 ControlSpace Designer software allowing users to discover, route and control Dante channels from within the application. 

In addition to integrating the core functionality found in Dante Controller software, ControlSpace Designer 4.1 also allows pre-programmed Dante audio routing to be recalled from any of the Bose user controls, or via a connected third-party control system, so that everything from basic source selection to the remapping of entire systems can be activated by end-users.

Dante accessory expansion cards are available for the ControlSpace ESP-00 II engineered sound processor, ControlSpace ESP-880/1240/4120 engineered sound processors, and for the line of PowerMatch configurable professional power amplifiers; in total, 12 devices are Dante-capable. The cards all foster 48 kHz/24-bit digital audio with 16 x 16 channels for ControlSpace processors and 8 x 8 channels for PowerMatch amplifiers. 

“With Designer software version 4.1, we continue to expand the capabilities of our processors and amplifiers without sacrificing ease of use,” says Darryl Bryans, Bose DSP product line manager. “By integrating Dante’s automatic device discovery and configuration tools into our ControlSpace Designer software, Bose has taken yet another step toward making it easier for system integrators to create and configure distributed audio systems quickly.”

ControlSpace Designer software version 4.1 will be available September 2, 2014, as a free download at http://pro.Bose.com.

Bose Professional Systems

{extended}
Posted by Keith Clark on 06/11 at 07:10 AM
AVLive SoundChurch SoundNewsProductAmplifierAVDigitalEthernetInterconnectNetworkingProcessorSoftwarePermalink

Tuesday, June 10, 2014

Roland Systems Group Announces XS Series Of Multi-Format AV Matrix Switchers

Designed for fixed installations requiring high-quality integrated video and audio conversion and switching

Roland Systems Group has announced the Roland XS Series, a new line of multi-format matrix switchers designed for fixed installations requiring high-quality integrated video and audio conversion and switching.

The series is adaptable, supporting eight HDMI, RGB/Component/S-video/Composite inputs and up to four HDMI or HDBaseT outputs with scalers to support picture-in-picture, resizing, rotating, and flipping. Audio can be embedded into outputs via eight stereo audio inputs (2 microphone) and/or HDMI audio as well as de-embedded on output. Additional features include iPad control, EDID emulation and HDCP management.

The XS Series is available in three configurations: 8-in x 4-out (XS-84H), 8-in x 3-out (XS-83H), and 8-in x 2-out (XS-82H) and is ideal for many applications that include conference rooms, education, 4K switching to 1080p, performing art centers, churches, convention centers and teleconferencing.

Remotely control the Roland XS Series using the native iPad app, XS Remote, via RS-232C interface, and control over LAN connections. Remote control is valuable for installed applications where front panel controls are not accessible or a programmable touch control interface is required that can access functions of the XS Series.

The XS-84 model is capable of switching four video sources as a group enabling the use of up to two 4K inputs. The multi-format capabilities allows users to switch 4K, HD, SD, XGA and other computer formats up to 1920 x 1200 (WUXGA).

Advanced video processing functions makes multiscreen video productions possible for both signage and live applications. Scale a single image across multiple displays using the SPAN mode and then switch to a different image on each screen in MATRIX mode. Each input channel supports adjustments for scaling, positioning and aspect ratio before output ensuring the best possible picture.

In education environments the XS series can switch between and route HDCP video from computers, smart phones, blu-ray, and even still images from internal memory to displays. Up to four HDBaseT outputs for long distance transmission of audio and video content to displays throughout a campus.

For teleconferencing applications, the XS series features four internal buses each for video and audio. It can be used to create minus-one audio to feed directly into teleconferencing systems. Switch up to eight computer and video devices both analog and digital formats. Audio ducking functions lowers audio levels when microphone audio is detected making it ideal for conference or boardroom environments.

The Roland XS Series of multi-format matrix switchers will be on display at InfoComm 2014 in Las Vegas at booth 10536.

For further details about the new XS Series, go here.

 

image

 

 

Roland Systems Group

{extended}
Posted by Keith Clark on 06/10 at 09:03 AM
AVLive SoundChurch SoundNewsProductAVDigitalEthernetInterconnectNetworkingProcessorSignalSoftwarePermalink

Monday, June 09, 2014

Audient Names New Distributor For South Korea

Kinoton Korea to distribute the full range of Audient products

Kinoton Korea has been named distributor for the full range of Audient products in South Korea, including the iD22 USB interface and ASP880 8-channel mic preamp and ADC.

“We’re very pleased to work with Audient and are very ambitious to promote the high-quality products to the Korean market, and look forward to having great success and feedback from the market very soon,” states Chris Bae, managing director of Kinoton Korea.

Audient sales and marketing director Luke Baldry adds, “Audient’s focus is on quality, which is supported by a proud heritage. Chris Bae and his team have a strong knowledge of the South Korean market, and we have every confidence that they will handle Audient sales, support and service at exactly the right level.”

Bae is the founder of Kinoton Korea and can be contacted via e-mail at .(JavaScript must be enabled to view this email address).

Audient

{extended}
Posted by Keith Clark on 06/09 at 02:09 PM
AVLive SoundNewsAVBusinessConsolesInterconnectManufacturerSound ReinforcementStudioPermalink

What Comes First, System Design Or Functionality?

This article is provided by Commercial Integrator

 
The neverending plight of the control system programmer is to program a system with nothing more than a set of drawings and a best guess at what the system should do.

Why is this such a common occurrence?

Because of the problematic approach of designing the system first, then worrying about what it should do later!

The first question any programmer asks when approaching a project that is already designed is, “How do you want the system work?” In many cases this becomes one of the most difficult questions to resolve, leading to responses like “use your judgment,” “the client is open to suggestions,” or “provide what is typical.”

The end result of these conversations is a solution that is too complex, too simplified, too flexible, too restrictive, or too far out of budget. What the client really needs and wants to pay for is “just right” but that can’t happen if there isn’t a conversation about functionality needs.

This situation happens way too often but it can be avoided if we change the discussion in the beginning. If we shift from conversations that begin with what equipment should be used to conversations about what the system is intended to do, what need we are trying to satisfy, what challenge the system will resolve, and what top functions the user wants, we are better able to produce solutions that deliver what the client needs.

The answers to these questions help to shape system functionality and define system operation.

Moreover, conversations early in the project that get to what the user actually needs on the most basic level lead to other conversations about features and custom software solution options.

When we define system functionality and user requirements from the start, system design follows naturally to support the operation. The question, what should the system do, is now removed from the discussion and we avoid delays, misfires, and costly errors at the end of the project.

Allowing functionality to shape system design ensures:

—Proper devices are selected with the necessary number and types of inputs and outputs.

—Ample signal paths are established to support system operation and future needs.

—Feature sets of selected equipment provide proper control to support the defined operation.

—User interface size, resolution, and type comfortably supports necessary functionality.

—Control methods and protocol enable effective integration of all equipment and provide the expected user experience.

Again, this demonstrates how a project and client benefit from involving the AV programmer in the early phases of project discussion. Early AV programmer involvement means well-defined functionality executed efficiently.

For more details on the benefits of connecting with the programmer early, ways to learn from others’ experiences (good and bad), and how to define an effective scope of functionality, we have created a community exclusively for Technology Mangers called TechTalk—join us for our inaugural event during InfoComm week on Tuesday, June 17 in Las Vegas.

Steve Greenblatt, CTS, is president of Control Concepts, Inc., a leading independent provider of audiovisual control system solutions.

Go to Commercial Integrator for more content on A/V, installed and commercial systems.

{extended}
Posted by Keith Clark on 06/09 at 12:48 PM
AVFeatureStudy HallAVBusinessInstallationInterconnectNetworkingSoftwareSystemPermalink

Allen & Heath Digital Consoles Help Manage Sound Reinforcement For Carnival Of Brazil

iLive-112s for live performances by internationally renowned artists, GLD-80s for broadcast feeds

Allen & Heath iLive-112 digital mixing systems played an important role in sound reinforcement at the renowned Carnival celebration in Rio de Janeiro, Brazin, featuring performances by renowned international artists including Claudia Leite, Jota Quest, Pericles, and Michel Telo. Mixes were also routed to four Allen & Heath GLD-80 mixing systems, which handled the worldwide broadcast feeds.

Allen & Heath’s distributor Teleponto supplied the consoles, designed the systems, and managed the live performances and global transmission of the carnival. Six teams and a total of 70 staff were required to ensure the successful operation of six straight days of transmission and 72 hours of live broadcasts.

The GLD-80 consoles were used to handle audio feeds sent from several remote locations spanning over 2 kilometerrs. In addition, four AR2412 and eight AR84 audio racks, feeding 7 kilometers of fiber optic cable.

“With more than 2 million people in attendance and simultaneous live feeds to more than a dozen international broadcast and online networks, reliability was of the utmost importance, and as expected Allen & Heath delivered,” states Antonio Pereira Neto, president of Teleponto.

Allen & Heath
Americian Music And Sound

{extended}
Posted by Keith Clark on 06/09 at 07:22 AM
AVLive SoundNewsAVConcertConsolesDigitalInterconnectMixerSignalSound ReinforcementPermalink

Friday, June 06, 2014

System Upgrade At New Jersey Courthouse Headed By Yamaha CIS Components

Serves up to 16 mics at one time for live feed and multi-track recording

The municipal courtroom for the township of Evesham, NJ, which also serves as a council and township meeting room and a township meeting room, was recently outfitted with a new system by ACIR Professional (Mars Landing, NJ) that incorporates Yamaha CIS (Commercial Installation Solutions) components.

Between the court and various meetings, there can be up to 16 microphones on at any given time, and they need to be fed to the live loudspeakers and for multi-track recording purposes.

“In addition to obtaining a better quality sound, the customer still wanted to be able to send audio via multi-track to record,” notes Bobby Harper, ACIR vice president of sales. “Having just demoed the new Yamaha CIS Series products, the design became obvious.”

ACIR Pro chose a Yamaha MTX5-D processor with the EiX8-8 channel extender in order to accommodate the 16 required inputs. The processor also handles the routing of multi-track and a 4-zone (mix minus) to the Yamaha XMV4280-D Dante-enabled 4-channel amplifier.

In addition, Yamaha VXC8W loudspeakers were chosen for their clarity and ease of installation. They’re zoned for maximum gain before feedback, with assistance from a Dugan MY-16 auto mixer card installed in the MTX5-D and the parametric EQ capabilities on each zone output,     

The system’s presets are stored in the Yamaha MTX5-D library, with recall available from an in-wall-mounted Yamaha DCP1V4S-US controller that the clerk has at his/her fingertips.

ACIR Professional
Yamaha Commercial Audio 

{extended}
Posted by Keith Clark on 06/06 at 09:32 AM
AVLive SoundNewsAVDigitalInterconnectMicrophoneMonitoringProcessorSound ReinforcementPermalink

CADAC US Launching At InfoComm: Showing Router, MADI & Dante Bridge Components

Focusing attention on its MegaCOMMS system network with the introduction of three key infrastructural network components

CADAC is exhibiting at InfoComm (booth C11532) for the first time as CADAC Audio Ltd US & Canada, the company’s new North American sales and distribution arm under the direction of general manager Paul Morini.

The company will be focusing attention on its MegaCOMMS system network with the introduction of three key infrastructural network components – the CDC MC router, CDC MC MADI network bridge and CDC MC Dante network bridge.

MegaCOMMS extends CADAC audio performance across scalable digital audio networks of up to 3,072 channels of 96 kHz/24-bit audio, comprising multiple CDC I/O stage boxes and consoles. The extensive latency management system automatically manages all internal routing and associated processing latency, providing time-aligned, phase-coherent transmission with sub-0.4 millisecond latency, from analog inputs on stage to analog outputs on stage.

The 2RU CDC MC router is the hub of the MegaCOMMS system network, with a single router supporting 128 channels of bi-directional 96 kHz/24-bit audio. The router currently allows up to four MegaCOMMS consoles to be linked to up to eight MegaCOMMS I/O stage boxes or audio network bridges within the same audio network, all on 150-meter (492-foot) runs from the CDC MC router.

Paul Morini, general manager of CADAC Audio Ltd US & Canada.

In addition, the router can be configured to allow a redundant console to run in parallel with the “live” console. Future upgrades will increase the power and expand the flexibility of this unit to take further advantage of the capacity of the MegaCOMMS network.

CADAC CDC MC MADI and CDC MC Dante network bridges allow the seamless integration of MADI and Dante units within a MegaCOMMS network. Both 1RU rackmount units can operate at 96 kHz or 48 kHz, and can handle up to 64 inputs and outputs.

Both are equipped with independent word clocks and come with dual PSUs as standard. Audio connections are co-axial on the CDC MC Dante network bridge (with RJ45 connectivity to Dante network devices), and co-axial and optical on the CDC MC MADI network bridge. As on the CDC MC router, co-axial connections helpfully glow red for the “ins” and blue for the “outs.”

The CDC eight-16 (single screen plus 16 faders) and the CDC eight-32 (dual screen plus 32 faders) digital production consoles will be shown; both have 128 channels and 48 bus outputs and the latest Version 2.1 software.

Version 2.1 includes several new features. VCA group deployment deploys the VCA group members on the console surface with a press of a button. A touch of the screen allows switching between input-driven and mix-driven Fader Follow. It’s also possible to view and access single input channel contributions to multiple buses, or a single bus contributions from multiple inputs when in Fader-Follow mode.

CADAC brand development manager Richard Ferriday states, “MegaCOMMS networks allow connection of multiple, CDC eight and CDC four, consoles and I/O stage boxes, or third party consoles, sources and audio networks, via the MADI bridge. With all audio samples synchronised before summing, there is absolute phase coherency at all outputs and negligible aggregated latency.

“Together with full 96 kHz/24-bit audio, this offers a major performance improvement in terms of sound quality, along with a network capability of up to 3,072 channels. No other console manufacturer can offer a complete in-house solution on this scale, with one-stop solutions for any large scale multi-roomed installation or large scale, multi-artist live events, such as TV shows, awards ceremonies, festivals or mega tours.”

CADAC Audio Ltd US & Canada will be exhibiting at booth C11532 at InfoComm 2014 in Las Vegas.

 

image

 


CADAC

{extended}
Posted by Keith Clark on 06/06 at 07:23 AM
AVLive SoundChurch SoundNewsProductAVConsolesDigitalEthernetInterconnectMixerNetworkingSound ReinforcementPermalink

Thursday, June 05, 2014

Selecting The “Right” Direct Box (DI) For Bass Guitar

What type of direct (DI) box works best for bass guitar? The answer is easy: it depends. In fact, more than anything else, it depends on the type of bass that the DI is going to be used with.

When it comes to signal flow, there are two types of bass guitars: passive and active. The first electric basses, i.e., the original Fender Precision, were passive, and in fact still are today.

They employed magnetic pickups to generate the signal - as the string moves in and out of the magnetic field, a low-level alternating current is generated.

The signal from the bass travels through the cable to the amplifier, which in turn increases the voltage level so that it is sufficiently powerful to drive another electromagnetic device: a loudspeaker. In essence, the signal is amplified by a series of buffers that work together to increase the voltage and/or current as needed.

For years this worked well, until bands like the Beatles messed everything up! The problem was that the fans at those concerts were so loud that the bass amp was unable to produce enough ‘thump’ to overtake the screaming. The solution: send the bass guitar signal through the PA system.

Eureka! The amazing direct box was born. The first direct boxes were basically hand-made black boxes that had transformers inside.

These passive devices would tap a signal off the bass and split it so that part of the sound would go to the bass amp on stage, and the rest of it would go to the PA system some 50 to 100 feet away.

Origins Of Active
As the PA systems got larger, so did the performance venues (or vice versa). Eventually, things escalated to the point where concerts moved to arenas and stadiums.

And bass players complained because they noticed that when their bass was connected to all of the long cable runs in these larger systems, the sound changed. It was not as beefy, and there was no more thud.

Block diagrams for Radial JDI (passive) and J48 (active) DI boxes.

This shouldn’t have come as a surprise—if you take the signal from a magnetic pickup and ask it to drive hundreds of feet of cable in addition to the bass amp on stage, the level will be weaker. And it will not sound the same. This effect is known today as “loading.”

The solution: buffer the bass signal. In other words, incorporate a small amplifier inside the direct box so that 99 percent of the signal is directed to the bass amp and 1 percent is split off to drive the PA. And thus the active DI box was born! Ye old Fender P-Bass was happy—the thud had returned.

Bring The Mayhem
So for the next bunch of years, everything worked just ducky, until one day, some guy decided to put a 9-volt battery inside the bass and buffer the signal.

Now all of the sudden, instead of the bass producing around 1 volt, the battery powered preamp inside the bass was kicking out 5 to 7 volts.

Then the CEO of the Acme Bass Company had a revelation: “We can do even better—let’s put in a second battery!”

A modern 6-string bass could now deliver a whopping 18 volts of mayhem, and bass players rejoiced. They could overload the front end of “ye old SVT” and finally out-blast that pesky lead guitarist and his lowly Marshall!

All good, expect for one problem: that 18-volt output now overloads the direct box, resulting in a distorted, muddy, no-punch sound in the PA system. Or, if you prefer, it just plain sounds bad.

The solution? Dust off the old passive direct box, connect it up and bingo, great tone - the thud is back.

Phantom Solution
Here’s the deal. Early active direct boxes were powered by batteries and in fact, some still are. But the problem with batteries is that they go dead… usually right in the middle of the second set.

So some years ago, DI manufacturers started to use phantom power as a means to supply the needed voltage and current to the active DI box (buffering amplifier).

But phantom power, invented by Dr. Neumann as a means to supply a polarizing voltage to his condenser microphones, was never intended to be a power source for an amplifier. And without current, you do not get headroom.

Think of a bass playing through a miniature guitar amp - turn it up, and it distorts like crazy. DI boxes do exactly the same. Without headroom, high-output bass signals will cause the buffering amplifier in the DI to distort.

But remember, back then, basses were all passive so for the most part, so they worked fine with regular phantom power as the buffers only had to process 1 to 3 volts. The advent of active basses with their huge output levels changed the rules.

Two Groups
So the rule of thumb is that for a high-output bass that already has a built-in buffer, a passive direct box will likely do a great job—the bass will produce the drive. On the other hand, for a low-output passive bass, an active DI will leave the bass sound unaffected while generating the drive for the PA system.

Keep in mind that the sound quality of DI boxes depends on the circuit design and parts that are being used. Better designs focus on eliminating all types of “bad” distortion such as harmonic, phase and inter-modulation distortion. These designs are then categorized into two groups.

Some direct boxes are designed to transfer the signal without artifact or distortion so that the original sound of the bass is delivered as purely and naturally as possible, while others, such as tube DI boxes, tend to be designed to “color” the sound with “good” distortion to create new bass tones and exciting textures. Both are useful, depending on the desired outcome.

Peter Janis is president of Radial Engineering.

 

{extended}
Posted by Keith Clark on 06/05 at 02:47 PM
Live SoundFeatureBlogStudy HallInterconnectSignalSound ReinforcementStagePermalink

Wednesday, June 04, 2014

Church Sound: A Comprehensive Tech Booth Remodel

This article is provided by ChurchTechArts.

 
I’ve spent the last several years looking at church tech booths. I knew, someday, we’d be able to relocate ours from the balcony to the floor. When we did, I wanted to be ready.

The design started about three years ago, according to file creation dates. It’s changed more than a few times over the years, but I’m pretty happy where we’ve landed.

Design Objectives
As you may have gathered from my series on renovations, I’m always starting with a set of design objectives.

In this case, we had several:

—Get front of house to the floor, and keep it in front of the edge of the balcony.

—Bring lighting and ProPresenter to the floor.

—Incorporate the existing camera platform.

—Have a “producer” desk.

—Keep front of house, lighting and ProPresenter in a line at the front of the booth.

—Maintain enough space for new volunteers to observe.

The back wall is existing; we raised the walls of the booth to obscure the monitors, and to provide a little more security. They have since added a short wall extension to the back wall as well.

That’s a pretty good list, and I think we hit it all. Design is all about compromise, and we did have to make some compromises. The booth is probably larger than it needs to be in terms of square footage. At 14 feet deep and 21 feet wide, it’s certainly spacious. The size was dictated by three things.

First, the depth was based on the camera platform we needed to incorporate. The ProPresenter desk sits in front of the camera platform, and we needed enough depth to accommodate the desk, chair and some room behind the chair.

We would have been fine with 6-feet 6-inches or 7-feet in front of the camera platform, but that would have landed us in the middle of a row of seats. If you have to remove a row, you may as well push the front wall out as far as possible.

Second, the width was based on a section of seating. After playing with a good half-dozen designs, it just didn’t make sense to not go full width. At best, we could have saved 1-2 seats on one end, and they would have been terrible seats no one would have ever used. So again, we went big. No need to go home.

Finally, we really needed to keep the front of house position out in front of the overhanging balcony. My original design was actually one row deeper (!) and had FOH on the left side of the booth, almost in the center of the room.

Leadership felt that was one too many rows, and looking at it now, I have to agree. So we moved FOH to the right of the booth, which is a little more off center than I’d like, but due to the curve of the balcony, we’re still in front of it by a few feet.

I wanted room for new volunteers to sit and observe without being in the way.

Fully Wired

Tech booths have a lot of cable in them. For years, we’ve had a pile of cables at the front wall/floor intersection of ours. We’ve cleaned it up quite a bit by adding some conduit and a slotted cable duct (and tearing out old cable that’s no longer used).

But for this one, I wanted it as clean as possible. As I mentioned last time in the conduit post, I located three 12-inch x 12-ich boxes with 36-port panels on them throughout the booth.

We put them as close to the rack locations as possible so everything will be pretty much straight runs. Because we’ll still have a few cables that won’t fit in the boxes (HDMI cables, for example), I’m also running a small 1-inch x 2-inch slotted cable duct around the perimeter.

Because we located the producer desk behind FOH, and it’s on the producer desk that the monitors for LAMA and the (Roland) M-48s live, I had to come up with a solution for that. I didn’t want the engineers to have to keep turning around to check levels or fix an M-48 issue. So I decided to double the monitors.

By using simple HDMI splitters, we’ll have a set of monitors in front of and slightly below the (DiGiCo) SD8, and another set on the producer desk. Wireless keyboards and Magic Trackpads at both locations will enable operation from either location.

We’ll also have the master screen and SD8 remote screen at FOH, along with the overview monitor. Including the built-in touch screen, that makes six screens at FOH. Excessive? Probably. But I’m a glutton for information.

All of our wiring is slated to live in F6 TecFlex with service loops so we can pull the racks and desks out to work on the backside. The desks will be on wheels, making it easy to get back behind for access.

It’s hard to see in this picture, but the brace is just behind the keyboard tray.

No More Smashed Knees
I hate most tech booth counters and desks. They either sag in the middle over time, or have bracing that smashes your knees, or a deep front brace that catches your thighs. I determined to engineer my way out of this.

I’m building the desks out of 4-foot x 4-foot redwood because it’s readily available out here. I’m placing a brace in the center of the desk where most of the weight will be concentrated so it won’t sag.

The tops are two layers of 3/4-inch plywood laminated together with glues and screws. The brace is far enough back that when sitting on an architect’s chair, I’ll be able to sit as high as possible without smashing my knees. I also designed a clever little slide out keyboard tray in the middle.

Sometimes I’m accused of over thinking things. And I’ve probably spent a few hundred hours working on this design over the years. But I believe when it’s done, it will be one of the nicest tech booths around. Even with the ugly pull box in the corner.

Mike Sessler now works with Visioneering, where he helps churches improve their AVL systems, and encourages and trains the technical artists that run them. He has been involved in live production for over 25 years and is the author of the blog Church Tech Arts.

{extended}
Posted by Keith Clark on 06/04 at 02:46 PM
Church SoundFeatureBlogStudy HallConsolesInstallationInterconnectMixerNetworkingPowerProcessorSound ReinforcementPermalink

Tuesday, June 03, 2014

Listen Technologies Introduces iDSP Assistive Listening Products

Smaller than an iPhone, the iDSP receiver has a field-replaceable, non-proprietary rechargeable lithium-ion battery

Listen Technologies has introduced iDSP (Intelligent Digital Signal Processing), a new line of assistive listening products.

In creating iDSP, Listen Technologies focused on the user experience and ease of dispensing, inventory management and battery management at the venue, based upon input from consultants, system integrators, venue owners, and end users.

The result is an improved listening experience, including an integrated neck loop/lanyard; streamlined dispensing, collecting and care; and environmentally friendly advanced battery technology.

The iDSP 72 MHz receiver offers precise clarity with 20 dB less hiss than other RF receivers. The new integrated neck loop improves the experience for people who have hearing aids and cochlear implants with telecoils.

Sleek and small (smaller than an iPhone), the iDSP receiver has a field-replaceable, non-proprietary rechargeable lithium-ion battery, making obsolete the use of alkaline and NiMH batteries. Battery life is 8 hours, with a 2.5-hour charge time.

Two versions of the receiver are available: LR-4200-072 Intelligent DSP RF Receiver (72 MHz) and LR-5200-072 Advanced Intelligent DSP RF Receiver (72 MHz). The LR-5200 Advanced Receiver features the ability for end users to select multiple channels for applications such as language interpretation. 

System components include the charging tray, which can be mounted in several ways; charging case; optional cable management system; earphones (foamless, solving certain sanitary issues); log book; setup/inventory software; and signage.
 
Listen Technologies

{extended}
Posted by Keith Clark on 06/03 at 02:34 PM
AVLive SoundChurch SoundNewsProductAVInstallationInterconnectMonitoringSignalSound ReinforcementWirelessPermalink

Radial Takes On Global Sales & Distribution Of Jensen Iso-Max Line

Upcoming InfoComm show in Las Vegas in June presents opportunity to debut new products and get things started

Radial Engineering has announced that it has taken on the global sales, marketing and distribution of the Jensen Iso-Max range of products.

Iso-Max is a range of isolators that provide ground isolation and noise abatement for audio and video in broadcast, home theater and commercial AV integration. 

Radial has a long history with Jensen. According to company president Peter Janis: “When Radial was founded in 1992, we started life as a distributor. One of our first product lines was Jensen. Back then, we sold raw transformers to sound companies and broadcasters who in turn built custom multi-channel snakes and splitters. As the market for snakes matured, sound companies moved away from custom snakes and home-built isolators to buying off-the-shelf solutions. 

“In 1996, we launched the Radial JDI (Jensen DI) which has become the most popular passive direct box in live concert touring and has been a cornerstone for Radial sales around the globe. Over the years, Radial has become Jensen’s largest customer and as we have grown, Radial and Jensen have become synonymous.”

Janis continues: “Anyone who knows Jensen knows that the company is engineering based. In other words, since their inception over 40 years ago, they have never hired a sales team or had a marketing department. The commitment to building the world’s finest transformers has created a huge following by those in the know.

“A few years ago, Jensen decided to take a similar route to ours by producing a range of plug and play solutions under the Iso-Max range. Jensen recently came to the conclusion that unless there are feet on the street telling dealers and contractors that the product exists, Jensen would miss out on a huge opportunity. This led to discussions which culminated in Radial taking over the sales and marketing side of the business.

“Over the coming months we will be setting up retail and contractor partners, independent reps in the United States and Canada, and formalizing exclusive agreements with distributors around the globe. The upcoming InfoComm show in Las Vegas in June presents a wonderful opportunity to debut the new products and get things started.”

The Jensen Iso-Max range includes ground isolators for baseband video, cable TV, balanced mic and line level signals, consumer devices such as laptops, and most recently, wall plates.

Dean Jensen changed the world when he discovered that audio transformers could be significantly improved by widening the bandwidth so that phase anomalies are eliminated and distortion reduced. By incorporating nickel laminations and applying advanced winding techniques, Jensen managed to widen the frequency response to extend between 5 Hz to 100 kHz while eliminating harsh-sounding group delay.

Further refinements included internal Faraday shields and external perm-alloy cans to reduce noise and more recently, improving the wire lead connections for greater durability. Today, Jensen transformers are made by hand in the United States and are supported with a 20-year factory warranty.

Jensen products can be seen at the upcoming InfoComm 2014 show at booth C8725, with Radial Engineering and Primacoustic at booth C11316.

Radial Engineering

{extended}
Posted by Keith Clark on 06/03 at 01:15 PM
AVLive SoundRecordingNewsProductAVInterconnectSignalSound ReinforcementStudioPermalink

Designer Notebook: Inside The Radio Active Designs UV-1G

First, some history. Professional wireless microphone and intercom systems have been operating in the United States for well over half a century, and now they’ve been joined by wireless in-ear monitoring systems.

At the current time, 95 percent of these systems operate in the Ultra High Frequency (UHF) spectrum from 470-698 MHz. This is because the wavelengths associated with radio frequencies in this band are well suited for portable wireless devices such as microphone transmitters and intercom packs due to the small size of their antennas.

In recent years, information technology giants like Google, Dell, Verizon, Sprint and others have acquired a significant interest in these radio bands, which have officially been home to the professional audio and broadcast communities since 1962.

Since the first smartphones were introduced to the consumer market, the requirement for radio bandwidth has tripled, and it grows exponentially as a steady stream of new portable consumer devices are designed and released by these entities.

The result has been overcrowding of the radio spectrum that, until recently, was reserved for the professional audio and broadcasting communities. In fact, pro audio/broadcast has already lost more than one-third of the UHF spectrum that was formerly available.

This year, white space consumer devices are hitting the retail market, resulting in further spectral crunching, and next year, another 100 MHz of radio spectrum above 600 MHz will be auctioned off to the highest bidder.

Meeting The Challenges
If this relentless onslaught of consumer digital radio devices continues to grow unchecked, the days of operating wireless microphone, IEM and intercom systems will swiftly come to an end.

That’s why a small group of wireless audio specialists, including yours truly, formed Radio Active Designs (RAD). Our objective is to design and manufacture spectrally efficient wireless audio products so that all live events, performing arts, and broadcast media may continue to flourish with minimal negative impact from consumer devices.

All five of the owners of RAD have worked for decades in the audio industry as wireless microphone and intercom operators, manufacturers, and event radio frequency (RF) coordinators. We’ve had the opportunity to see—first-hand and in real-time—what’s happening with the radio spectrum.

The Radio Active Designs UV-1G wireless intercom system. (click to enlarge)

We’ve also been involved in FCC discussions regarding the future of these consumer devices and how much radio spectrum the pro audio wireless community will be left with after these devices are introduced to the world.

After studying the amount and type of wireless usage in the U.S., we determined that more than half of the frequencies in use on a typical event are taken up by wireless communications devices. If you add up every wireless microphone, IEM and IFB system at a large event, that number is still less than the number of wireless intercom systems used on that same event.

Therefore, to make the greatest impact in relief from this spectral congestion, we chose our first product to be a wireless intercom system, the new RAD UV-1G.

New Direction
Most of the frequencies used by a wireless intercom are the belt pack to base station frequencies. For example, a stage manager’s wireless intercom system may have only one base station transmitter frequency but require 12 belt pack frequencies.

As a result, our approach with the UV-1G completely removes belt packs from the UHF spectrum, instead using Very High Frequencies (VHF) from the belt pack to base station.

To accomplish this, we’ve designed a unique modulation scheme that we’ve verified is 10 times more spectrally efficient than the current FM (Frequency Modulation) technology on the market today. This proprietary approach, called Enhanced Narrow Band, is a form of Amplitude Modulation (AM), and it makes the transmitter’s occupied bandwidth—a critical figure in wireless frequency coordination—more predictable.

For example, a typical FM wireless intercom system requires 300 kHz of radio band to function properly. By implementing Enhanced Narrow Band, the UV-1G requires less than 30 kHz, and even if it performed in the UHF band, it would still be about 10 times more spectrally efficient.

A RAD-generated comparison contrasting UV-1G and standard system parameters. (click to enlarge)

Thus by moving to the VHF band for the belt pack to base station frequencies, we can fit more than 30 base stations and 200 belt packs into less UHF spectrum than one FM wireless intercom system.

From an RF coordinator’s perspective, this makes the UV-1G system more than 30 times more spectrally efficient than current UHF FM technology. The bottom line is that it frees up valuable UHF spectrum that can be used for wireless mic and IEM systems.

A great deal of time and effort was invested in making Enhanced Narrow Band AM signal sound as good as any FM system currently available. Frankly, in the not-too-distant past, this challenge would have been insurmountable, but due to contemporary engineering techniques, we’ve been able to implement a host of digital processing technologies that help produce sound quality with the warmth and intelligibility of an FM system.

A closer look at a UV-1G belt pack. (click to enlarge)

Further Refinements
Another goal was to improve the design of a standard production belt pack from the user’s perspective. For example, for a production involving 120 belt packs, only one stage manager needs the “stage announce” function. Why dedicate this prime real estate to an “SA” button for essentially 1 percent of the users on the event?

So rather than pre-assign modes of operation for the user interface buttons, UV-1G belt packs include two completely programmable buttons that offer users a choice of multiple independent functions that can be quickly programmed. This can be done locally at the belt pack or via our software program.

What this means for wireless intercom rental companies is that one technician can program an entire event’s worth of packs in a matter of minutes. It’s a far cry from the hours that it takes to manually program current technology for the same number of packs.

As owners of rental companies ourselves, we’ve experienced numerous problems over the years that have been addressed on the UV-1G based on real world issues that we have experienced with other wireless intercom systems on the market today.

The headset connector on both the belt packs and the base station is field changeable between 5-pin female and 4-pin male to accommodate the various systems based mostly on the industry served. This is a solder-less connection that only requires a mini Philips screwdriver.

Another significant problem at events is the belt pack antenna falling off and getting lost, resulting in the operator losing communications until a repair can be made. Particularly in a life safety environment, we felt that this is an unacceptable risk. UV-1G belt packs implement internal antennas to alleviate the problem of bending, breaking, or completely losing the antenna.

Further, RAD belt packs also include a 1/8-inch stereo audio input so that monitor technicians may connect IEM receivers directly to their belt packs. Naturally, this input may be used for any audio source. And there’s only one band split so that every belt pack works with every base station on the event. No more scrounging for the correct RF band splits.

To minimize rack space and maximize ISO channel operation, UV-1G systems allow for up to six belt packs per base station, and up to six base station links, for a total of 36 ISO channels between packs. That’s three times more than anything we’ve experienced in our wireless careers.

Front and back views of the base station. (click to enlarge)

The UV-1G base station comes factory delivered with two transmitter RF connectors so that it may immediately be connected to a combiner, such as the RAD TX-8, without any modification of hardware.

The base station may also be connected to all standard wired communications systems, including Clear-Com, RTS and 4-wire systems.

Finally, the UV-1G can operate completely out of the portable white spaces device band, maximizing chances for success in even the most hostile RF environments.

James Stoffo is a founding member of Radio Active Designs (RadioActiveRF.com), and prior to that, he founded Professional Wireless Systems. James also continues to work as the RF technician and frequency coordinator on large-scale special events and installations. Contact him at .(JavaScript must be enabled to view this email address).

{extended}
Posted by Keith Clark on 06/03 at 12:32 PM
AVFeatureBlogProductAVInterconnectMicrophoneSound ReinforcementSystemWirelessPermalink

Monday, June 02, 2014

Altinex Announces New CNK210 And CNK210S Interconnect Boxes

Intera-style modules plus power and charging USB capability yields excellent connectivity

Altinex has introduced new CNK210/CNK210S interconnect boxes, and they’ll be on display at the upcoming InfoComm 2014 show in Las Vegas at booth C5116.

Both the CNK210 / CNK210S install into tabletops and provide convenient access to cable ends or fixed connectors, making them a great choice for patching in notebook PCs for presentation purposes, accessing the company network, and similar functions.

Both are also outfitted with the Altinex SP3504SC connector plate, a popular connector choices that has three openings for Intera type modules—two of which are populated: one with a dual AC/charging USB module and the second with digital video (HDMI), VGA (15-pin D-sub) video, plus 3.5 mm audio connectors.

The third opening is populated with two RJ-45 network connectors, with the remaining area open. The package also includes six open snap-in connectors, enabling the unit to be ‘customized’ with connectors of one’s choosing.

CNK210/CNK210S interconnect boxes come fully assembled, consisting of the CNK200/CNK200S cable-nook frame plus the SP3504SC connector plate, enabling integrators to place the entire interconnect system into the table, patch in the associated equipment, and be up and running in very little time.

Grant Cossey, Altinex vice president of sales, states, “The new CNK210 and CNK210S provide today’s AV professional with a one-stop solution to the most common interconnect requirements for today’s projects,” says Cossey. “With dual AC/charging USB ports as well as both digital and VGA video, these new interconnect boxes make it really easy to provide the connectivity required for a high percentage of today’s AV projects.

“With the CNK210, meeting attendees can actually power their tablets or smartphones without searching for a wall outlet. And with the open snap-in connectors, these interconnect units can be ‘fine-tuned’ to each specific environment. Offering exceptional value, I’m confident AV integrators and end users alike will find these new boxes particularly useful.”

The Altinex CNK210/CNK210S interconnect boxes will be available in mid-June 2014, with MSRP pricing at $449.

Altinex

{extended}
Posted by Keith Clark on 06/02 at 02:49 PM
AVLive SoundChurch SoundNewsProductAVEthernetInstallationInterconnectNetworkingPowerSignalPermalink

Friday, May 30, 2014

Ashly Amplifiers And Signal Processors Now Shipping With Dante

Will significantly enhance the power and value of company's networkable amplifiers line and select signal processors

Ashly Audio is now shipping Dante digital media networking solution with all their nXe, nXp, Pema and NE Series networkable amplifiers. NE Series rack-mount system processors (4400, 4800, 8800) are also available for purchase with Dante.

With low latency, robust synchronization, I/O scalability, and simplicity of installation via standard IT technology, Dante will significantly enhance the power and value of Ashly’s networkable amplifiers line and select signal processors.

“We’re now ready to supply our market with this incredible audio networking technology in our nXe and nXp amplifiers” says Anthony Errigo, director of communications at Ashly. “Dante currently brings the best in flexible, yet robust, digital audio transport and we’re poised to build on its success by delivering this connectivity in our most premier amplifiers and processors. Our customers will benefit greatly by this advancement.”

Dante is a robust proven solution which is widely deployed in hotels, transportation centers, shopping centers, public address systems, live sound reinforcement, theaters, concert halls, stadiums, athletic venues, corporate boardrooms, universities, broadcast facilities, recording facilities, houses of worship, government facilities, and courtrooms.

Ashly Audio

{extended}
Posted by Keith Clark on 05/30 at 03:25 PM
AVLive SoundRecordingChurch SoundNewsProductAmplifierAVDigitalInterconnectNetworkingProcessorSignalSound ReinforcementPermalink
Page 50 of 143 pages « First  <  48 49 50 51 52 >  Last »