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Monday, October 26, 2009

Wireworks Debuts MCAT5 Multipin-Based, Multi-Channel CAT5e Network Cabling

Wireworks has introduced new MCAT5, the first-ever multipin-based, multi-channel CAT5e network cabling. MCAT5 simplifies network cabling by eliminating individual cable runs; reducing the wear and tear on equipment by utilizing a sturdy multipin connector instead of the standard RJ45 connector, creating a rugged point-to-point secure connection.

Additional benefits of MCAT5 include ease of use, simplified cable identification and improved network durability.

MCAT5 tails are configured to support six channels of 10/100/1000BASE-T signals and equipment requiring four pairs per RJ45. 

Tails are also available to support 12 channels of 10/100BASE-T utilizing cable sharing technology.

“Wireworks introduced multipin, multi-channel audio cables to replace individual mic cable runs which revolutionized analog audio cabling,” says Gerald Krulewicz, president of Wireworks. “Now, MCAT5 replaces multiple network cable runs in a single durable cabling solution. Wireworks remains at the forefront of innovation providing products that meet the demands of the ever changing audio industry.”

MCAT5 is constructed of heavy-duty CAT5e rated cable with an extra-tough double jacketed shielded cable design.  With a small overall diameter of only 0.57 inch and weight of 14 pounds per 100 feet, MCAT5 is easy to work with and transport. Cable sections are available in lengths up to 250 feet.

Tails are color-coded for easy identification and terminated with either RJ45 or etherCON connectors. 

Rack mounted MCAT5 fanouts can be permanently attached to equipment eliminating wear and tear, transferring the strain from equipment RJ45 connectors to Wireworks heavy-duty, proven road-worthy G-Block Multipin Connector.

Wireworks G-Block rectangular connector design eliminates cross-threading, locking ring damage as well as the possibility of going “out-of-round” – problems typically found in circular connectors.  G-Blocks precision formed brass contacts are resilient to breakage and bending.

Wireworks Website

Wednesday, October 14, 2009

Eliminating Troublesome Hum & Buzz Created By Electric Guitars

You’re recording an electric guitar, or amplifying it through a P.A., and there it is: hum! This annoying sound is a common occurrence.

Acoustic guitars fitted with pickups can have the same problem.

Hum is an unwanted 60 Hz tone—50 Hz outside the U.S.—plus harmonics. If the harmonics are especially strong, the hum becomes an edgy buzz.

Let’s take a look at what’s going on and how to fix it. First we need to review how an electric guitar works.

Inside The Electric Guitar
Built into the guitar, under the strings, is a magnetic pickup: a transducer that converts the strings’ vibration into an electrical signal. The pickup is a bar magnet wrapped with thousands of turns of wire, forming a coil.

When the player plucks the steel strings, they vibrate next to the magnet, producing a similar vibration in the magnet’s magnetic field, which in turn causes a varying current in the coil.

Another type of pickup uses a separate magnet under each string. Some pickups have a screw on each magnet’s polepiece to adjust the distance between the polepiece and string, allowing level control of each string.

A humbucking pickup uses two coils wired in series but with opposite polarity so that they cancel common hum fields. One coil is mounted far from the strings.

The high-impedance signal from the pickup coil goes through a simple circuit (Figure 1) and comes out the unbalanced guitar jack.

Components in the circuit are usually connected by single wires. The sleeve (ground) terminal on the jack is connected to the pickup coil, the strings, and the shield around the circuit.

Figure 1: A typical electric-guitar circuit. (click to enlarge)

From the guitar jack, the signal travels through a guitar cord: an unbalanced shielded cable.

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At the end of the cable, the signal can go to several destinations: a direct box, a guitar amp, a mixer’s high-Z input, or guitar stomp boxes/processors.

Most acoustic-guitar pickups are piezoelectric types installed under the bridge or saddle. Vibrations of the guitar body flex the pickup, which generates an electrical signal. It’s very high impedance, and often is run through a preamp built into the guitar which reduces the impedance.

Whether the guitar is electric or acoustic, any component in the signal chain is susceptible to picking up hum and buzz, especially because the entire circuit is high-impedance unbalanced.

Hum Sources
Alternating current in a building’s power wiring generates electric and magnetic fields that oscillate at 60 Hz and its harmonics. Hum fields also radiate from lighting circuits and fluorescent lights.

The magnetic fields couple inductively to the guitar wiring. When the magnetic lines of force cut the conductors in the guitar and its pickup, the conductors generate a 60 Hz signal, which is amplified by the mixer or guitar amp.

Also, the power wiring and pickup act as two plates of a capacitor. The varying electric fields from the power wiring couple capacitively to the pickup and guitar wiring.

Another hum source is radio-frequency fields from computers, motors, and TV transmitters (vertical sync, blanking and vertical component video).

This RFI can be detected by the guitar or audio equipment.

A major cause of hum is the ground loop. It is the circuit loop that is formed when two pieces of audio gear are connected to each other through a cable shield and also through the AC safety ground.

If the two chassis are at different ground potential, a 60 Hz current can flow on the cable shield connecting them, causing audible hum.

Figure 2 shows a ground loop. Two equipment chassis (guitar amp and mixer) are connected to two separate safety grounds by their AC cords.

Also, the equipment chassis are connected together by the shield of the audio cable coming from the direct box. The shield and safety-ground wires form a ground loop.

Figure 2. A ground loop. (click to enlarge)

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Causes & Cures
Let’s give several examples of causes and cures of hum related to electric guitars.

Magnetic Hum Fields
As we said, AC in a room’s power wiring generates electric and magnetic fields that oscillate at 60 Hz and its harmonics. When the magnetic lines of force cut the conductors in the guitar and its pickup, the conductors generate a 60 Hz signal, which is amplified by the mixer or guitar amp.

Cure: The amount of hum generated depends on the angle between the pickup coil wires and the magnetic hum field. At certain angles, a lot of the hum goes away. So the player can rotate or move around to find a spot with minimum hum.

Electric Hum Fields
The power wiring and pickup act as two plates of a capacitor. The varying electric fields from the power wiring couple capacitively to the pickup and guitar wiring.

Cure: A grounded Faraday shield. In the guitar body are cutouts that house the electronics, wiring and pickups. These cutouts should be lined with conductive foil (such as copper foil) that is soldered together to form one continuous piece. This shield must be connected to the guitar jack’s ground terminal so that the hum fields are bypassed to ground (ideally, to the mixer chassis).

The guitar cord should also be well shielded. Use only high-quality cords with plugs having a metal jacket, which acts as a shield to the wires inside it.

Some guitar amps are painted on the inside with conductive paint that acts as a shield, but this paint coating can crack when the amp cabinet is jostled, breaking the shield connection. It’s best to run a ground strap between all panels of the amp head cabinet.

RFI (Radio Frequency Interference)
A strong TV signal can be rectified and demodulated by some electronic components or a bad solder joint.

Cure: This is a subject in itself (see the references at the end of this article). But for a quick fix, install ferrite beads and .001 microfarad capacitors on mic inputs. Install RFI chokes on guitar cords or mic cables. Check solder joints.

Ground Loop
Suppose you’re recording a guitar direct, and the guitar is plugged into a guitar amp. The amp and your mixer have 3-prong power cords that connect to the safety ground. The amp is plugged into an AC outlet across the room, and your mixer is plugged into a nearby outlet. When you connect the amp ground to your mixer ground through the mic-cable shield, and monitor the signal, you hear hum.

Chances are that the outlets are fed from different circuit breakers, so the outlets are at different ground voltages. When you plug your amp and mixer into these separated outlets, and connect the equipment together with a mic cable from a direct box, the difference in ground voltages can make a 60-Hz hum current flow between the guitar amp and mixer. That’s a ground loop.

Cure: Flip the ground-lift switch on the direct box to break the loop. Also, it’s a good idea to power the mixer and guitar amp off the same outlet strip. That way, the ground voltage for all the equipment is about the same, so little or no hum current can flow between their chassis. Run a thick extension cord from the mixer’s outlet strip to the guitar amp, and plug the amp into the extension cord.

There still may be a slight voltage difference between components because their power supplies reflect different voltages onto their chassis. A balanced AC power supply can eliminate this problem.

Before you plug in all those power cords, make sure that the sum of the equipment fuse ratings does not exceed the amperage rating for that circuit. In most cases, a single 20-amp breaker will handle a small studio.

Guitar Not Grounded
Suppose you’re recording a guitar with a direct box, and the guitar is NOT plugged into a guitar amp. If the ground is lifted on the direct box, the guitar is not grounded, so you hear a loud buzz.

Or if the shield connection is broken in the guitar cord or mic cable, the guitar is not grounded.

Cure: Flip the ground-lift switch to the grounded position when not using a guitar amp. Check inside the cable connectors to make sure the shield is soldered at both ends. Replace or repair guitar cords that have broken shields.

Player’s Body Not Grounded
When the guitar player touches the strings, does the hum stop? This indicates that the player’s body is acting as one plate of a capacitor.

The capacitance between the body and power wiring adds to the capacitance between the guitar and power wiring, increasing the level of the hum transmitted from the power wiring to the guitar. (Incidentally, the same thing happens if you replace the player’s body with a sheet of aluminum foil).

Cure: Run a wire between a ground point on the guitar and the player’s skin. Figure 3 shows a ground wire (highlighted in yellow) between the guitar-jack ground and the player’s big toe!

Figure 3. A wire between the guitar ground and the player’s body can stop hum. (click to enlarge)

This grounds the player’s body, so that it acts as a partial shield for the guitar, rather than a capacitor.

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A body close to the guitar increases hum, and connecting the body to the guitar ground stops the hum. The body is not a ground for the guitar. Rather, the guitar ground is a ground for the body.

So now we know why some heavy-metal guitarists play without a shirt. They’re removing the dielectric between their skin and the guitar.  (Thanks to Crown’s Chris Vice for that insight).

Caution: Do NOT use this ground wire in a concert situation if the guitar is plugged into a guitar amp. There might be a shock hazard if the player touches a mic. That can happen if the mic, which is grounded to the FOH mixer, is at a different ground potential than the guitar amp onstage.

To reduce the potential between mixer and guitar amp, power the mixer through a thick extension cord plugged into the AC distro outlet that the guitar amp is plugged into.

In other words, do NOT make a permanent connection between the player’s skin and the guitar ground in this situation. You might ask the player to keep their hands on the strings whenever possible.

For a technical discussion of body grounding, please see the sidebar that follows this article.

Shock Hazard
This is not about hum, but is an important related issue. In concerts, electric-guitar players can receive a shock when they touch their guitar strings and a mic simultaneously. This occurs when the guitar amp is plugged into an electrical outlet on stage, and the mixing console (to which the mics are grounded) is plugged into a separate outlet across the room.

As stated before, these two power points may be at widely different ground voltages, so a current can flow between the grounded mic housing and the player touching the grounded guitar strings.

Electric guitar shock is especially dangerous when the guitar amp and the console are on different phases of the AC mains.

The cure is to power all instrument amps and audio gear from the same AC distribution outlets. That is, run a heavy extension cord from a stage outlet back to the mixing console (or vice versa).

Plug all the power-cord ground pins into grounded outlets. That way, you prevent shocks and hum at the same time.
 
Using a neon tester or voltmeter, measure the voltage between the electric-guitar strings and the metal grille of the microphones.

If there is a voltage, flip the polarity switch on the amp. Use foam windscreens for additional protection against shocks.

Quick Tips
When you hear hum or buzz from an electric guitar, try these solutions:

• Turn up the guitar’s volume and treble controls so that the guitar signal overrides hum and noise picked up by the guitar cable and guitar amp.
• Ask the guitarist to move around, or rotate, to find a spot in the room where hum disappears.
• Flip the polarity switch on the guitar amp to the lowest-hum position.
• To remove buzzes between guitar notes, try a noise gate.
• If the hum stops when the player touches the guitar strings, ask the player to keep his or her hands on the strings, or run a wire between the player’s skin and a ground point on the guitar (such as the strings or the jack ground.)
• Set the direct-box ground lift switch to the position where you monitor the least hum.
• Replace or repair guitar cords that have broken shields. Use only high-quality cords with metal-jacket plugs.
• Power the guitar amp off the mixer’s outlet strip.
• Use guitars with humbucking pickups, or install modern humbuckers in older guitars.
• Line cutouts in the guitar body with copper foil wired to the guitar-jack ground.
• If you suspect RFI, install ferrite beads, capacitors and chokes. Also see the references below.
• Replace any defective tubes in the guitar amp. If the power-supply filter capacitors in the guitar amp are corroded, replace them. This replacement should be done by an authorized technician.
• Use a quieter amplifier.
• Don’t use a noisy amp. Instead, record the guitar direct, then process its track with a guitar-amp modeling plug-in or processor.
• Don’t use SCR lighting dimmers because they add noise and hash to the AC power. Instead, use multiway incandescent bulbs to vary the studio lighting levels. If you must use a SCR dimmer, rotate its knob to find a position with the least hum (maybe the “off” position!).
• Run the studio off its own breaker, not shared with noisy loads such as air conditioning, power tools, etc. Don’t ground the neutral at more than one point (have an electrician check this). Use an AC line isolation transformer between the AC power and the studio equipment.

If you follow these suggestions, the only buzz you get should be from the guitar player’s solo! Good luck.

Acknowledgement: Many thanks to these Syn Aud Con members for their helpful discussions: Jim Brown, Rick Kamlet, Bob Hagenbach,  Mike Miles, Pat Brown, Steve Roth, and Peter Patrick.

Bruce Bartlett is a microphone engineer (http://www.bartlettmics.com),  an audio journalist, and a recording enginee (http://www.bartlettrecording.com). He is the author of “Practical Recording Techniques 5th Edition” and “Recording Music On Location”.

Go to NEXT PAGE for a related sidebar article about body grounding by Jim Brown.

Sidebar

Technical Discussion About Body Grounding
by Jim Brown, Audio Systems Group
http://audiosystemsgroup.com

The human body is a conductor with relatively high resistivity, and it is a fairly large conductor. This means that when it makes contact with an electrical circuit it can act as an antenna, and it can also act as one “plate” of a capacitor.

The other “plate” of that capacitor might be a noise source like a power line, a noisy electric light, or computer wiring. The noise might be base band (that is, audio frequency), or it might be modulated RF, or it might be both.

The body will react differently to those noise sources depending on what they are—their frequency content, their internal impedance, their orientation with respect to the body, etc. And the body will interact with the circuit of the audio equipment and its wiring.

The various effects of the body in any given circuit will add algebraically—that is, they may be varying degrees of in phase, and they may be in or out of polarity, and they will be at various relative levels with respect to each other, so in any given field condition they will be different.

Some examples of the guitar problem. Let’s say that the body touches the “hot” conductor of a guitar cord plugged into an amplifier. The body can act as both a capacitor, coupling both audio and RF into the equipment, and it can act as an antenna. What’s the difference? 

The word “antenna” implies reception or transmission of an electromagnetic field—that is, the simultaneous existence in space of an electric field and magnetic field at right angles to each other, and in which energy is traded back and forth between electic and magnetic fields.

An antenna has both current flow along it and a potential difference along it that either is caused by the field (reception) or generates the field (transmission). 

So when the body is acting capacitively, it is NOT acting as an antenna, it is not “receiving noise” and coupling it to the equipment. It has become part of the equipment’s wiring and is an element in the equivalent circuit. Now, the body may be acting as a capacitor to one noise source (or in one frequency range) and as an antenna to another, and may be doing so simultaneously! 

Another way that the body can get into the act is by causing current flow on the shield of an unbalanced cable. That current can couple noise in at least two ways. First is the IR (or IZ) drop in the shield, which is added to the signal. Second is via a pin 1 problem.

When the orientation of the guitar is important, there are three mechanisms I can think of that can be at play. First and most obvious is the null that occurs when the circuit that is inductively coupled to a magnetic field is at right angles to that field.

Second is the movement of the body and the guitar so that it is physically closer to the noise source, and thus has a higher capacitance to the noise source.

Third is the directivity of the antenna that it is part of. 

Suggested References

Radio Frequency Susceptibility of Capacitor Microphones (Brown and Josephson). AES Preprint #5720

Noise Susceptibility in Analog and Digital Signal Processing Systems (Muncy). JAES June 1995

Shure Introduces New Professional DJ Headphones

Shure has introduced a new professional headphone model designed specifically for professional DJ applications – the SRH750DJ. They join the SRH840, SRH440, and SRH240 to expand the Shure lineup of professional and home recording equipment, which also includes the X2u XLR-to-USB signal adapter and the PG27USB and PG42USB side address condenser microphones.

Custom 50mm drivers tuned to deliver high-output bass with extended highs offer maximized power handling, which optimize the headphones for use with DJ mixers.

Comfortable, padded ear cups swivel 90 degrees and allow total control of placement on one ear when mixing. Cable and ear pads are easily replaceable.

“Professional DJs have unique requirements when it comes to headphones,” said Scott Sullivan, Shure Senior Director of Global Product Management. “The SRH750DJ is tuned to deliver high-output bass cleanly, with extended highs that enable precise mixing even in noisy club environments.  The SRH750DJ also allows one-ear or two-ear use, and has increased power handling capability for use with DJ mixers.”

Features:
• Custom 50mm drivers tuned to deliver high-output bass with extended highs
• 3,000 mW maximum input power allows for optimized connectivity to DJ mixers
• Adjustable, collapsible headband with 90-degree swivel ear cups for comfort and
easy one-ear placement
• Closed-back, circumaural design rests comfortably over the ears and reduces
background noise in noisy club environments
• Bayonet Clip securely locks cable into ear cup
• Replaceable ear cup pads ensure long product life
• 3m (10ft) coiled detachable cable provides plenty of length and easy storage
• Carrying bag protects headphones when on-the-go or not in use
• Two-year warranty

Pricing is $149.99 retail ($188 MSRP) and availability is November 2009.

Shure Website

Thursday, October 08, 2009

TASCAM Debuts New US-100 Bus-Powered USB 2.0 Audio Interface

TASCAM has announced the US-100, a new USB 2.0 audio interface outfitted with a microphone input as well as instrument level input for direct recording guitar or bass.

Stereo line inputs are provided, and can be switched to phono-level. It is housed in a solid aluminum chassis to withstand any abuse from the road.

The US-100 is a 48k/16-bit stereo audio interface that connects to a Mac or Windows computer using USB 2.0. A free copy of Audacity software is included to start recording right away.

Additional features include a 1/8-inch stereo headphone output and zero-latency monitoring.

The US-100 is available from TASCAM retailers at a street price of under $100.

Features:
• Bus-powered USB 2.0 audio interface
• XLR or 1/4-inch microphone input
• 1/4-inch guitar-level direct input
• RCA line inputs, switchable to RIAA photo inputs with ground lug
• 1/8-inch stereo headphone output
• 48kHz/16-bit audio resolution
• Zero-latency hardware monitoring
• Mac and Windows compatible
• Audacity recording software included
• Solid aluminum case construction

TASCAM Website

Friday, October 02, 2009

Prodigy Engineering Develops First Remote Preamp For API 500 Series

Prodigy Engineering has developed Bella, the first-ever, remote-controllable, microphone preamplifier for the API 500 Series format.

Bella features relay stepped gain from 18 dB to 69 dB in 1 dB increments, switchable high-pass filter (-3 dB @78 Hz), phantom power (48V), polarity inversion (0/180˚), attenuation pad (-22 dB), and mute functionality.

Additionally, Bella can be remote controlled from professional standard digital audio workstations like Apple’s Logic software and Digidesign Pro Tools | HD systems, as well as ICON control surfaces.

Bella will be on display at the Prodigy Engineering booth #642 at the upcoming 127th Audio Engineering Society Convention in New York City at the Javits Center.

Prodigy Engineering Website

Tuesday, September 22, 2009

Extron USB 2.0 Hub For Architectural Mount Applications Now Shipping

Extron Electronics has announce that the new USB HUB4, Four-Port USB 2.0 Hub for architectural mount applications is now shipping.

The USB HUB4 allows the sharing of up to four USB devices on a single host port and is available in two versions: AAP - Architectural Adapter Plate and MAAP - Mini Architectural Adapter Plate.

It streamlines integration by eliminating the need for separate USB Host ports and their cables, and is ideal for any environment where multiple USB devices must share a single USB port.

“A/V integrators often face the challenge of where to mount a USB hub in lecterns, A/V control desks, and other architectural applications,” says Casey Hall, Extron Vice President of Sales and Marketing for North America. “The USB HUB4 provides a simple, elegant solution for USB connectivity in boardrooms, classrooms, and similar environments.”

For integration flexibility, the USB HUB4 is equipped with both USB “Mini” Type B and captive screw host inputs. The captive screw input is ideal for installation in lecterns, junction boxes, and other space-constrained applications.

The USB HUB4 also provides 5V, 500mA on each output, delivering power to multiple peripherals such as mass storage devices, keyboards, mice, or other HID - Human Interface Devices.

Both versions of the USB HUB4 are designed to mount easily in Cable Cubby furniture-mountable access enclosures, HSA - Hideaway Surface Access enclosures, AVTrac floor-mounted raceway for A/V connectivity, wallplates, and other products that accept a double space AAP or MAAP.

Extron Website

Visit our Web site at for more information.

Tuesday, September 15, 2009

Focusrite Introduces New OctoPre MkII Multi-Channel Microphone Preamp

Focusrite has announced the imminent arrival of the OctoPre MkII, which features eight channels of Focusrite pre-amplification and a built-in 24-bit/96 kHz ADAT output, providing an affordable input upgrade for a Pro Tools system or any digital audio workstation.

OctoPre MkII combines award-winning Saffire PRO pre-amps with high quality digital conversion and JetPLL jitter elimination technology.

The digital output allows users to make the most of often-neglected ADAT inputs; ideal for expanding the number of mic-pres for interfaces like the Saffire PRO 24. Connect OctoPre MkII to an audio interface’s ADAT input to create a high quality, multi-channel studio recording solution.

OctoPre MkII is equally suited to the live environment as a quality mic-pre expansion for any analog or digital console, or hard disk recorder. With line outputs on every channel, each mic-pre can be routed to a separate channel on an analogue mixer, with the ADAT output left free to send a copy to a digital recorder.

OctoPre MkII has been optimized for drum recording. Designed not to clip, 10dB pads are provided across each channel, and the gain range of the pre-amps has been tailored to handle extreme levels from sound sources like the kick drum.

Equally, Focusrite pre-amps are renowned for the way they handle the dynamic range of vocalists, and these pre- amps are no exception. Finally, OctoPre MkII’s first two channels also feature DI’s, so it can also turn its hand to recording guitar and bass.

OctoPre MkII includes 5-LED input metering on every channel, switchable phantom power and a variety of internal and external clocking solutions.

Cost is $499.99 US MAP, with October, 2009 projected availability.

click to enlarge

Focusrite Website

Wednesday, September 02, 2009

Product Review: Aviom Network Audio Systems

Aviom, known to countless church sound personnel and musicians for its “more of me” personal monitor mixing controls, has put considerable work into how to move audio signals from one place to another. It’s led to the development of a digital snake system and audio networking system, which Kent Gibson discusses in the August, 2009 issue of Church Production magazine.

How does this technology work? What are the advantages (and potential downsides) of transporting signal in the digital domain? And most importantly, what can it bring to your church sound system?

Click here to check out Kent’s insights and explanations regarding these questions and much more, in addition to an overview and perspective of the Aviom approach.

Also, be sure to have a look at the roster of online articles from the August, 2009 issue of Church Production.

Other recent Church Production reviews:
Peavey Sanctuary S-32 Console
DiGiCo SD8 Digital Console
Sony UWP Wireless System
JBL EON 515 Loudspeaker
Audio-Technica AT2020USB Microphone

Tuesday, August 25, 2009

AVnu Alliance Launches To Advance Audio/Video Bridging (AVB) Networking Standard

A group of industry-leading audio/video (A/V), consumer electronics and silicon companies today (August 25, 2009) announced the launch of AVnu Alliance.

AVnu is an industry forum dedicated to enhancing professional-quality audio/video by promoting emerging IEEE 802.1 Audio/Video Bridging (AVB) networking standards for a broad range of markets including professional AV and consumer electronics.

The founding members of the AVnu Alliance include Broadcom Corp., Cisco Systems., Harman International, Intel Corp., Samsung Electronics, and Xilinx

In addition to the founders, Avid Technology, Marvell and Meyer Sound Laboratories have joined the AVnu Alliance as the first promoters. 

AVnu Alliance aims to establish a professional quality A/V experience in networked environments, whether an HD television or music studio, a car, a concert hall, a stadium or a home theater.

Today, out-of-sync audio and video, glitches and delays can occur in many of these settings, unless complex, proprietary solutions are deployed.

For example, in networked whole-home audio systems, there is no standards-based solution to make the speakers play in sync.

To address these issues, the AVnu Alliance is promoting the IEEE standards, currently in development, for 802.1 AVB (Audio Video Bridging) and also the related IEEE 1722 and 1733 (which extends RTP for use with AVB). 

The draft AVB standards are designed to work over widely-used IEEE 802 layer 2 networks. These new standards provide networking features for tightly controlled media stream synchronization, buffering and reservation.

Use of AVB enables higher layer protocols and applications to realize professional-quality A/V even if there are various lower-layer network links in the path between endpoint devices.

AVnu expects to see initial deployment of AVB on Ethernet networks and anticipates other home networking standards will follow.

“The AVB technology developed by the IEEE has reached a level of maturity that permits its use in the creation of innovative new products,” said Rick Kreifeldt, AVnu Alliance chairman and president. “Our mission is to drive these cutting-edge technologies into the professional A/V, automotive, and consumer electronics markets, enhancing the quality of experience across a broad range of products and applications.”

AVnu Alliance is committed to bringing together leading companies to promote and advance these technologies. The organization will support the creation and implementation of compliance test procedures and processes that promote interoperability of AVB-enabled networked products, helping to ensure A/V devices work together to provide a professional level of quality.

These efforts will enhance the network backbone, complementing the ongoing work of existing organizations and standards bodies specifying higher layer A/V protocols and applications in each market space.

“A/V networks are becoming burdened by greater complexity and the ever-increasing demands of streaming content, yet there are few options to ensure reliability in a heterogeneous network based upon open industry standards,” said Jonathan Gaw, Research Manager at leading IT market research and advisory firm IDC.

“Broad, cross-industry efforts are crucial to ensure that the quality-of-experience is addressed early in the product development cycle, and to promote the interoperability of products being deployed in professional A/V, automotive and home networking scenarios that are more demanding than ever before.”

AVnu Alliance invites the participation of companies interested in advancing these efforts. For more information about becoming a member of AVnu Alliance, go to www.AVnu.org 

AVnu Alliance Website

Wednesday, August 19, 2009

Univision “Premios Juventud” Awards Show Features Sennheiser Chrome Wireless

The recent sixth-annual Univision “Premios Juventud” awards show utilized Sennheiser’s wireless technology to recognize the U.S. Hispanic youths’ top recording artists, movie stars, sports figures and pop culture icons.

Over 80 wireless channels were used for the podium mics and most of the monitoring.

Stylized performances from long-time favorites and up-and-coming celebrities punctuated the announcements, with reggaeton star Alexis y Fido performing with a pair of chrome-customized Sennheiser SKM 5200 wireless mics.

“Whereas most awards shows feature performances that include a little bit of stage scenery, things are a lot more intense at ‘Premios Juventud,’” said broadcast mixer Tom Holmes. “All of the performers are going for full vignettes to help create a mood and a story.

“Alexis y Fido asked us for ‘futuristic looking’ mics to go with their futuristic theme. I asked a few mic manufacturers, and Dawn Birr at Sennheiser was the first to respond and the only one to respond with a range of options. Although we were given the choice of gold, silver, and jewel-encrusted ‘bling’, the futuristic chrome on the Sennheiser SKM 5200 turned out to be exactly the sort of thing they were looking for.”

Like any awards show these days, “Premios Juventud” is awash in RF signals – each of them critical. Both of the podiums used two Schoeps microphones tied to a pair of hidden Sennheiser SK 5212 body-pack transmitters.

“Obviously, dialog is very important to TV people,” laughed Holmes. “We had to go with the best to ensure that there would be no dropouts.”

James Stoffo of Professional Wireless Systems, who handled critical wireless issues for the show, specified a rack of Sennheiser EM 1046 receivers, the company’s top-of-the-line modular receiver and serves as the complement to its high-end 5000 Series transmitters. “Stoffo knows more about RF than anyone I’ve ever met, and he swears by the 1046, both for its reliability and its sound,” said Holmes.

In addition, almost everyone that performed used 300 IEM G2 personal wireless monitoring systems from Sennheiser. “There was a lot going on,” said Holmes, “so we were glad that we didn’t have to worry about our wireless reliability. With Sennheiser, you do your preparation, lock things in, and roll.”

Sennheiser Website

Wednesday, August 12, 2009

System For Southern Baptist Convention Powered By New Yamaha TXn Amplifiers

The Southern Baptist Convention, held in June at the Kentucky Fair and Exposition Center in Louisville, was served by a sound reinforcement system powered by 24 new Yamaha TX6n amplifiers  

The Yamaha amplifiers were purchased from Capital Design Group by the Southern Baptist Convention (SBC) late last year, in preparation for this year’s event. The SBC-owned audio inventory is vast and can support crowds of up to about 15,000 in a convention center/exhibit hall/theater configuration.

“We chose the new Yamaha TX6n amplifiers based on three main criteria,” states Jeff Davidson, SBC Technical Coordinator. “First, was the needed power and sonic quality to properly support our distributed loudspeaker system; second, the ability to maintain a completely digital signal path from the digital mixer through the signal processing and directly to the amplifiers utilizing an EtherSound protocol. This was made possible via Yamaha MY-16-ES64 and MY16-EX Mini-YGDAI cards inserted into the Digital I/O slots on the TX6n amps. And third, the extensive system monitoring and internal digital signal processing capability afforded by the new amplifiers.”

“The Yamaha amp manager software was great for an event like this where the speaker and amplifier zones are hundreds of feet apart,” adds Phil Allison, System Engineer. “Being able to actively monitor individual amplifier channel parameters during the program was a great advantage.”

Front of House Engineer Chris Hinkle notes that ‘the Yamaha TX6n amplifiers did indeed supply an amazing amount of headroom. With the increased overall headroom, the amplifiers produced less distortion throughout the system.  Even though the room had a fairly high ambient noise level (HVAC), we could get the program above the noise to an intelligible level without the distortion we had been accustomed to.’

Davidson also said that Yamaha Commercial Audio staff were available to ensure a successful event and provided exceptional customer support, from sales engineering and network configuration, to telephone and on-site set-up support. “We are so pleased with the new amps, we intend to purchase additional TX6n to replace our current monitor amps before next summer’s event.”

Yamaha Commercial Audio has supported the SBC for over 20 years, and this year provided the mixing, digital signal processing, and amplification selection consisting of DME64N, PM1D, PM5D-RH and PM5D-EX digital consoles, DSP5D Expander, SB-168ES stage box, all on an EtherSound network.

The SBC Louisville event featured choirs, orchestras, and worship teams from First Baptist Church of Woodstock, GA; Second Baptist Church of Springfield, MO; Hunter Street Baptist Church of Hoover, AL; and Highview Baptist Church of Louisville, KY. Featured Christian performers included NewSong, Brian Free & Assurance, music evangelist Luke Garrett, and worship recording artists The Paul Baloche Band.

Audio assistance was provided by Bill Thrasher, Thrasher Design Group; Chris Hinkle, Prestonwood Baptist Church of Dallas; Blair McNair, independent monitor mix engineer; Phil Allison, Waveguide Consulting; Jim Carey, Liberty Baptist Church; Jack Pitts, Capitol Design Group; as well as house of worship product and marketing managers from Yamaha Commercial Audio Systems, Inc. Jeff Davidson of First Baptist Church of Dallas provided technical coordination.

The Southern Baptist Convention sound team, left to right: Bill Thrasher - Thrasher Design Group Inc., Kennesaw, GA; Steve Storie, Southwestern Baptist Theological Seminary, Fort Worth, TX; Jeff Davidson, First Baptist Church of Dallas, TX; Kathy Allison and Phil Allison, Waveguide Consulting Inc., Decaturm GA; Lon Brannies, Yamaha Commercial Audio Systems Inc.; Chris Hinkle, Prestonwood Baptist Church, Plano, TX; Blair McNair, Independent consultant and monitor engineer; Jim Carey, Liberty Baptist Church, Hampton, VA; and Nathan Rathel, First Baptist Church, Woodstock, GA (SBC stage manager).

Yamaha Commercial Audio Website

Sennheiser Wireless Microphones & Personal Monitors On Tour With The Love Willows

The Love Willows, currently on the road in support of their debut album, are utilizing Sennheiser wireless microphone systems as well as Sennheiser wireless personal monitors.

The group is made up of lead vocalist Hope Partlow and guitarist/vocalist/multi-instrumentalist Ryan Wilson, who provide tight, charismatic, easy-going performances.

Their album, “Hey! Hey!”on Decca records, features many more instruments than just vocals and guitar, and the duo bring that support with them on the road using prerecorded tracks.

As a result, the stage setup is fairly simple for FOH/monitoring engineer Jacob Orr. “Both Hope and Ryan sing, and we have a Sennheiser SKM 935 G2 wireless vocal mic paired with a Sennheiser EM 550 G2 receiver for each of them,” said Orr. In addition, both musicians use Sennheiser ew 300 IEM G2 wireless personal monitors.

Orr was grateful to have Sennheiser’s easy-to-use frequency selection while on tour with The Veronicas, who were using at least twelve channels, and The Pretty Reckless, who were using at least four channels.

“With other wireless manufacturers, I feel like I’m playing bingo with wireless channel selection,” he joked. “Sennheiser makes things much more intuitive and reliable. Before we rolled into a city, I went to http://www.sennheiserusa.com and scoped out the frequency landscape so that I had a good starting point. Then I used the scan function to ensure that we wouldn’t conflict with anyone else on the bill.”

The only other instrument on stage is Wilson’s Fender Custom ’72 Reissue Telecaster, which he plays through a Fender 4x10 Deville. Orr used a Sennheiser e 906 to capture its sound with authenticity. “I find that with the ‘usual’ guitar mic, I’m always scooping out the same frequencies,” said Orr. “The e 906 gives a nice representation with minimal processing. I barely touched the EQ.”

Sennheiser USA Website

Saturday, August 08, 2009

Stereo, Surround & More: A Look At The Core Sound TetraMic Microphone

The Core Sound TetraMic is the first portable, coincident, stereo and surround-sound Ambisonic soundfield microphone under $1,000, and it compares very favorably with similar soundfield mics selling for $3,000.

This pencil-sized microphone offers amazing sound and flexibility at an affordable price. Some applications include stereo or surround recordings and broadcasts of musical ensembles, sound effects, news events, sporting events, environmental sounds, and general ambience.

How It Works
The TetraMic is a soundfield system that utilizes a patented four-capsule microphone and decoding software.

On the top of the tiny mic are four sub-cardioid electret-condenser capsules of 12 mm diameter mounted 90 degrees apart.

The capsule grilles define the faces of a tetrahedron. Below the capsules is a slim brass handle with a built-in 6-pin tiny XLR connector.

Other cables and connectors (described later) let you connect the mic to a recorder of your choice.

The signals coming directly from the four capsules are called A-format. While these signals are not usable as they are, they feed into Core Sound’s processing software which creates a four-channel signal called the soundfield B-Format.

Those four channels are known as:
X (front-back)
Y (left-right)
Z (up-down)
W (omnidirectional, a reference for the other three channels).

The X, Y and Z channels are effectively three figure-eight patterns at right angles to each other.

The Core Sound TetraMic

By summing and differencing those B-format channels in varying amounts, the processing software can create a wide variety of mono, stereo, mid-side, or surround polar patterns—even in post-producton after the 4-channel recording has been made.

Surround formats include 5.1, 6.1 and 7.1; loudspeakers in a square, and loudspeakers in an octagon. Almost any loudspeaker arrangement can be user-defined.

Although the capsule diaphragms are separated about 1.3 inches, the processing results in phase-coherent signals, as if the capsules occupied the same point in space.

The four B-format channels capture the 3-D sound field all around the microphone, as picked up at a single point. These signals can sum to mono with no phase cancellations.

The processing also matches the capsule sensitivities, and equalizes the capsule signals to give them a wide, flat frequency response.

So you start with a four-channel recording of the mic-capsule signals. Then using the processing software in post, you can effectively point or steer the “effective” or “virtual” microphone(s) as desired.

The software allows modeling of essentially any number of coincident first-order microphones, each pointing at any arbitrary angle and each having an independent pickup pattern. 

For example, suppose you set the software to create a virtual figure-eight mic. As you move a slider in the software to turn the virtual mic off-axis, you can hear the recorded source become more distant as the null of the figure-eight pattern sweeps toward the source.

Or suppose you create a virtual Blumlein array of two figure-eights angled 90 degrees apart. As the sound source moves from center to 45 degrees to the right, you hear the sound image move the same way to the right monitor speaker.

When the source moves beyond 45 degrees, you hear the sound imaging becoming out-of-phase and diffuse, just as you would with a real Blumlein pair.

It’s as if you had a mono mic, stereo mic, or surround mic that could be rotated, tilted, or zoomed at will—after the recording is made.

The Physical Mic & Cables
A TetraMic system of mic, cables and adapters can be connected in various ways. Figure 1 shows a typical system.

Figure 1. The parts of one TetraMic system (click to enlarge)

The TetraMic is connected to an extension cable, which goes to a breakout cable, which is connected to four PPA phantom-power adapters with XLR plugs.

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The signal at those XLR connectors is low-Z balanced.

The PPA adapters plug into a mixer, audio interface, or mic preamps that supply 48-volt phantom power, or they can plug into an Audio Technica AT8531 power module, and many others.

The front of the ruggedly constructed mic is indicated by the “Core” logo on the machined brass handle.

A Switchcraft TB6M connector in the mic handle mates with a TA6F in the extension cable, and this extension cable plugs into a “four 3-pin mini-XLR-F breakout cable”, which divides into four 3-pin mini-XLR plugs.

The signal chain of this connection option is:
TetraMic > extension cable > breakout cable > PPA phantom power adapters >  mic preamp with phantom power.

Because the mic-capsule signals are medium impedance unbalanced, it’s best to limit cable runs to 20 feet before converting to low-Z balanced with the PPA adapters.

Core Sound offers another connection option:
TetraMic > extension cable > dual 5-pin adapter cable > Core Sound 4Mic power supply/mic pre/A-D converter > 2-track digital recorder with S/PDIF input.

The Core Sound 4Mic ($899) is a handheld, battery operated preamp and A/D converter that provides four discrete outputs or a matrixed 2-channel signal that can be decoded later.

When used with the 4Mic, the TetraMic can record the Ambisonics information to a standard two-channel flash-RAM recorder like the M-Audio MicroTrack.

So for about $2,000, you can carry a versatile, handheld surround recorder in your hand.

The 4Mic provides two output data formats: four channels on two S/PDIF outputs (optical or coaxial), or four channels multiplexed onto a single S/PDIF output.

According to Core Sound, ” We’ve found that a few commercial mic pre/ADCs will do the job. One is our 4Mic. Others are the MOTU Traveler, Prism Sound’s Orpheus, Metric Halo ULN-8, and Apogee’s Ensemble. The MOTU Traveler can be used as an excellent quality stand-alone mic pre-amp, in addition to its FireWire interface for PCs.

Its line level outputs can be connected to a Sound Device’s 744T four-channel recorder’s line inputs to make fine sounding recordings. Apogee’s Ensemble is one of the few mic pre/ADCs that, like our 4Mic, can be battery powered.”

“TetraMic works great with the RED Digital Cinema Camera. Its microphone pre-amp inputs are four mini-XLR jacks. We offer 6-foot cables that plug into the TetraMic PPA (XLR-F) on one end, and the RED camera mic inputs (mini-XLR-M) on the other. For the first time, cinema cameras are now able to record surround sound in-camera, and also track individual sound sources during post-production!”

Included with the TetraMic is an effective shock mount which slips onto the handle. It mounts to any standard 5/8-27 threaded mic stand. A fixed stand mount without shock mounting is also available.

Core Sound offers wind screens to reduce wind noise when recording outdoors with the TetraMic.

Manufacturer’s specifications:
Frequency response (after DSP correction): 30 Hz - 18.5 kHz +/- 2 dB.
Self-noise per capsule: 19 dBA. In my opinion, this may be a little too noisy when recording very quiet sound sources.
Maximum SPL per capsule: 135 dB SPL.
Sensitivity per capsule: 7.0 mV/Pa nominal (-43 dB re 1V/Pa).

Software
Once you have purchased the TetraMic, Core Sound provides a link to download three types of processing software (for free), and the calibration files for your particular TetraMic. I’ll describe the software below.

VVTetraVST and VVMic VST plugins
If your DAW can use VST plugins, these let you monitor the recording in real-time, fully decoded. You can record either A-format (four channels), B-format (also four channels) or files decoded to any specific microphone and playback configuration (from 1 to a very large number of channels).

You insert the plugins into a surround bus in your DAW that is fed by the four A-format tracks that you recorded.

The VVTetraVST plugin corrects the capsule frequency responses with downloaded calibration data (an advantage over many other mics).

Figure 2. VVMic VST screen (click to enlarge)

It also converts from A- to B-format.

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The VVMicVST plugin (Figure 2) decodes the B-format signals so you can set the mic parameters (for example, the number of microphones, the angles at which they point, each mic’s directivity) and match your playback system configuration.

Figure 3 is a screen capture of Reaper DAW software set up for use with the TetraMic.

The four A-format mic signals are on tracks 1, 2, 3, and 4.

Those four tracks feed into a multichannel track 5, which contains the VVMic VST and VVTetraVST plugins.

Figure 3. Reaper DAW software set up for use with the TetraMic (click to enlarge)

Track 5 is soloed so you hear only the decoded signal (the virtual microphones).

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VV Mic for TetraMic
This is a stand-alone application which changes A-format signals to B-format, and decodes the B-format signals into mono, stereo, or surround mic signals.. Here’s one way to use it:
1. Make a 4-channel recording of the TetraMic signals.
2. In your DAW, export each channel’s recording as a mono file.
3. Launch VV Mic. Multi-select those four files and import them for processing.

Here’s another way:
1. Make a 4-channel recording of the TetraMic signals.
2. In your DAW, pan each capsule’s signal to a different surround channel.
3. Export a multichannel file.
4. Launch VV Mic, and import that multichannel file for processing.

Figure 4. VV Microphone software main screen (click to enlarge)

Figure 4 shows the main screen of VVMic.

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The input files can be A-format or B-format. You can output a processed file (in B-format or decoded as a virtual mic) or just listen in real-time.

The Rotate screen lets you rotate, tumble (roll vertically) or zoom the sound images.

Figure 5. VV Microphone software decode screen. The polar pattern graphics change in real-time as you vary the mic parameters (click to enlarge)

In the Decode screen (Figure 5), you can vary the elevation, azimuth, width, directivity and gain of the polar patterns.

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All these actions are instantaneous and work smoothly.

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Measurements
I measured the frequency response of the TetraMic on and off axis. Using Reaper DAW software with VVMic VST and VVTetraVST,

I was able to decode the TetraMic’s signals in real-time to create a single omnidirectional, cardioid, and bidirectional virtual microphone to be measured.

Using TEF software, I first measured the frequency response of a loudspeaker at 1 foot with a laboratory reference mic, with the TEF set to 1 msec receive delay.

Then I measured the frequency response of the same loudspeaker at 1 foot with the TetraMic, with the TEF set to 41 msec receive delay (because the latency of the TetraMic software was 40 msec).

I manually differenced the two response curves to create the plots shown in Figures 6, 7, and 8.

As the plots show, the frequency response of the TetraMic’s virtual microphones is flat within 1.5 dB over most of the audible range.

Data below 200 Hz is omitted because it included sound reflections off the test-room surfaces.

The TetraMic has the flattest and most extended response of any velocity (cardioid or fig-8 type) mic, thanks to the digital EQ in the software.

Figure 6. Omnidirectional frequency response (click to enlarge)

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Figure 7. Cardioid frequency response (click to enlarge)

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Figure 8. Bidirectional frequency response (click to enlarge)

In Use
According to Core Sound, “One stand-alone digital recorder that folks have used with great success is the Sound Device 744. While it only has two microphone pre-amps (two short of TetraMic’s four outputs), the MOTU Traveler’s four mic pre Line outputs can be plugged into the 744’s Line inputs.”

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On its web site, Core Sound offers tips on recording with several different systems that include the TetraMic, MOTU Traveler, a PC, Sound Devices 744, and a 2-channel recorder.

I used the TetraMic to record speech walk-arounds, acoustic guitar in a live room, and an old-time string band with the mic in the center of the group.

The TetraMic fed a PreSonus Firepod audio interface and a computer running Reaper DAW software. I had tried the software with Cakewalk Sonar Producer, but they were not compatible.

Overall, the sound quality of the TetraMic was excellent. Image focus and placement sounded natural. Localization was precise, yet with plenty of ambient detail.

I heard clean, clear sound, plenty of low end and depth, and no audible distortion. Movement of images across the stereo stage was very smooth.

When used outdoors with a windscreen, the mic picked up little or no wind noise.

The string band recording sounded natural but a little lacking in upper-midrange presence compared to a typical close-miked commercial recording. It was hard to hear the guitar strings being strummed.

The response curves do show a small dip in that range. However, it was easy to get a brighter sound with equalization.

While listening to that string-band recording, I set up one virtual cardioid mic in VVTetra VST. As I rotated the cardioid in a circle using the Azimuth setting, I heard each instrument in isolation. Very cool!

According to Core Sound’s president, Len Moskowitz, “Folks are using this function for post-production of panel interviews to highlight specific speakers, and also in movie sound post to follow a moving sound source around. It won’t replace a boom operator but does provide another powerful tool for the sound crew. For music, you can highlight a soloist, like a spot mic.

“You can also do this function more than once (for multiple sound sources), and then mix the resulting decodes together. 

“So do a Blumlein decode for stereo ambience and mix in spot decodes for soloists. You can also move a microphone’s pattern null around to exclude noise sources, even ones that move around.”

I received excellent technical support from Len Moskowitz at Core Sound and from the software’s developers, David McGriffy and Richard Lee.

The TetraMic is a huge advance in recording technology, and a great value. Highly recommended.

Core Sound provides a 30 day trial period. All TetraMics returned within 30 days of shipment qualify for a full refund (minus shipping and handling charges) provided that they are returned in as-new condition. TetraMic is sold with a 1 year limited parts and labor warranty.

You can find out more about the TetraMic at www.core-sound.com, and company owner Len Moskowitz can be reached at .(JavaScript must be enabled to view this email address)

AES and Syn Aud Con member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.
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More recording articles by Bruce Bartlett on PSW
Simulating A Live Drum Solo In The Studio
Remastering Jazz Classics: The Dave Brubeck Quartet, Art Pepper, and Sonny Rollins
Deconstructing Hip-Hop To Hear How The Mix Comes Together
Recording Microphone Techniques To Produce Warm, Spacious Stereo

Friday, August 07, 2009

Recent Wireless Technology Series: Shure UHF-R Series

The Shure UHF-R currently reigns as the most widely rented and specified wireless microphone system.

It’s been on the market for four years and has garnered legions of fans.

Let’s look at some of the features and technical specifications that make the UHF-R system tick.

Components
Receiver options include the single-channel UR4S and the dual-channel UR4D. Most often seen, of course, is the dual-channel variant.

Three transmitters are available, including the UR1 standard bodypack, UR1M micro bodypack and UR2 handheld.

The bodypack units are connected to lavaliere and headworn microphones via a TA4, and in the case of the smaller UR1M, a 3-pin Lemo can be ordered as an option.

The handheld transmitter can be outfitted with a wide variety of Shure mic capsules, ranging from the SM58 to the SM86 to the high-end KSM9.

This is indeed arguably one of the stronger aspects of this series – the direct compatibility with Shure’s industry-standard range of capsules.

The handheld transmitter is available in standard black finish as well as the more “sexy” satin nickel. In the past few years, the old rule that “microphones must be black” (otherwise they might reflect light into TV cameras and/or be noticed by the audience) has abated, so this additional finish option fits right in with the current preference of options. 

Technical
The UHF-R system employs Shure’s proprietary and patented Audio Reference Compander system.

Single-channel UR4S and the dual-channel UR4D (click to enlarge)

Audio companding is a “necessary evil” in any analog wireless system, because otherwise there’s not enough dynamic range available in the link to provide an appropriate full-range musical signal.

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The Shure approach is very well regarded by engineers and artists alike for its natural sound quality. Dynamic range is specified as >105 dB, A-weighted, which is quite good - better than CD, in fact. Most high-quality wireless systems today have a similar specification for dynamic range.

Overall frequency response of the system is listed as 40 Hz -18 kHz +1, -3 dB. That’s a good specification and compares well to other high-quality analog systems.

In contrast, digital and digital hybrid systems extend a bit lower and a bit higher, with overall flatter response.

Nonetheless, the audio range of the UHF-R is more than enough to satisfy the demands of touring and installed sound markets, as evidenced by the high level of acceptance.

The UR2 handheld transmitter can be outfitted with a variety of Shure mic capsules(click to enlarge)

Transmitters in the UHF-R line allow for two different RF power settings, which I mentioned in an earlier post is something we’re seeing more.

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For the U.S. market, the UR1 bodypack transmitter can be switched between 10 mW and 100 mW.  The UR1M microbodypack and the UR2 handheld can be switched between 10 mW and 50 mW. 

For Europe, the handhelds and bodypacks all offer selectivity between 10 mW and 50 mW, to satisfy different regulations. The basic idea behind switchable power is that the user can choose between “low battery consumption” and “long range”.

The different in range between 10 mW and 100 mW (a 10 dB difference in power) is about 50 to 60 percent, which can be significant depending on the application.

And, of course, with all questions of range there are a vast number of external factors involved as well. I plan to get into this topic in more detail in a separate post – stay tuned.

The frequency ranges offered in the UHF-R package cover the entire available spectrum.

The UR1M micro bodypack (click to enlarge)

The G1 range covers the very low end of the available UHF spectrum, from 470 MHz - 530 MHz, followed by the H4 range from 518 MHz - 578 MHz.

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Note that these two ranges overlap a bit - a nice feature in some cases, but users should be aware that it is possible to have a transmitter at the top end of G1 step on one at the bottom of H4.

The J5 covers 578 MHz - 638 MHz and the L3 range covers 638 MHz - 698 MHz (the top of the current legal UHF range for wireless microphones in the U.S.). The system is also available in bands for Europe, Japan and Korea, among others.

Note that each of the U.S. ranges covers a fairly wide 60 MHz and provides nearly 2,400 frequencies in 25 kHz steps. The frequency ranges for Europe tune as wide as 75 MHz.

Networking
The UHF-R system was among the first to include complete networking capability, which greatly enhances the ability to quickly coordinate all frequencies and implement them throughout the receivers and transmitters.

Other companies have followed suit: Sennheiser with the Net1 system, and more recently, the 2000 Series; AKG with the DMS700, and Audio Technica.

Most, if not all, professional wireless systems will have these features within a few more years, but Shure was very wise to take such a comprehensive approach early.

Pricing
The UR4D receiver typically sells for about $2,800, while the standard beltpack transmitters are about $800 and handheld transmitters range from about $900 to about $1,800, depending on the capsule chosen.

These prices compare directly to the Sennheiser 2000 Series (see previous post) and thus are in the “upper middle” range. The Sennheiser 3000 and 5000 Series systems are considerably more expensive, while the Lectrosonics 400 Series and AKG 700 are somewhat less expensive.

Conclusion
It is obvious that the UHF-R system from Shure is well thought-out, with an excellent set of features and with a high level of quality that satisfies professional customers in many markets.

Shure did its homework when designing this system and it has paid off, and most certainly will continue to do so.

Find out more about the Shure UHF-R Series here.

Signing off for now…
Mike Wireless

Mike Wireless is the nom de plume of a long-time RF geek devoted to better entertainment wireless system practices the world over.

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More posts by Mike Wireless:
The Myths of Wireless System Transmitter Power
Latest Wireless Series #4: Inside the Sennheiser 2000 Series
The RF Spectrum Before & After The “Big Day”
Latest Wireless Series #3: Inside The Lectrosonics D4
Latest Wireless Series #2: Inside The MIPRO ACT Digital
Latest Wireless Series #1: Inside The AKG DMS 700
Is The UHF Spectrum Going To Ease Up After June 12?
Change The Only Constant In Marketplace For Wireless System Spectrum

Thursday, August 06, 2009

Peavey Expands Networking Options For MediaMatrix Products With Audinate

Audinate has announced that Peavey has expanded its networking options for the MediaMatrix product portfolio by including Audinate’s networking solution. 

The first MediaMatrix products to have Audinate’s advanced networking technology are the NION nX product family and the CAB 4n audio bridge. 

The NION system is a configurable DSP core for the commercial, engineered systems marketplace.  The CAB 4n is a break-out box designed specifically for NION-based systems. 

In addition, support for legacy MediaMatrix makes the CAB 4n an efficient option for a wide variety of systems.

Audinate has created the world’s first truly compliant IP over Ethernet networking solution for the professional AV industry.  Having solved the problem of transporting high quality real-time media over standard multi-use TCP/IP computer networks, Audinate’s networking solution, Dante, provides a high performance media transport that is tightly synchronized, with sub-millisecond latency.

Audinate’s patent-pending networking solution provides a no-hassle, self-configuring, plug-and-play digital audio network that uses standard internet protocols. Dante is a scalable solution that works on both 100Mbits and 1Gigabit Ethernet.

“Audinate brings a level of performance to the marketplace above the existing CobraNet standard,” said Mark Pinske, General Manager of MediaMatrix, Crest Audio and Architectural Acoustics. “Contractors should take advantage of Dante’s higher bandwidth, lower latency and trouble free configuration that is now available to them.”

Audinate’s solution simplifies network installation due to its innovative automatic device discovery and system configuration capability. With this capability, specialized skills are no longer needed to set-up and manage an audio and video media network.

“Peavey’s MediaMatrix solution is recognized as one of the market leaders of large scale networked audio systems in stadiums, airports, casinos, hotels, theaters, education and government facilities,” said Lee Ellison, CEO of Audinate. “We are thrilled that Peavey continues to broaden the adoption of Audinate’s solution across their product offering.”

Audinate Website
Peavey MediaMatrix Website